From jason at jasonjgw.net  Thu Apr  1 00:00:52 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:00:52 +1100
Subject: [Freeswitch-users] ZRTP on the FreeSWITCH PBX - questions on
 the log report
In-Reply-To: <COL108-W1015593859EC5279008F8CD91D0@phx.gbl>
References: <COL108-W1015593859EC5279008F8CD91D0@phx.gbl>
Message-ID: <20100401070052.GA13609@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> I recently deployed ZRTP on the FreeSWITCH PBX on a Linux machine and I have
> two X-Lite softphones (on Windows) properly connected and secure with zfone
> and I am wondering why the FreeSWITCH log is returning the following errors.

These have been discussed previously on the list, and as I remember, they are
normal errors resulting from the fact that the first several packets aren't
encrypted.
> Also, the zfone on the machine receiving calls won't stay secure for a
> longtime after the two softphones are connected. I need help to understand
> this.

Can you be more specific? How do you know it doesn't stay secure? Someone who
knows Zfone will have to answer this.

I regularly use ZRTP successfully with calls between FreeSWITCH systems, and
they are always set up and maintained as secure ZRTP-encrypted sessions.

I do however get the erros that you quote as a normal part of the process.



From jason at jasonjgw.net  Thu Apr  1 00:09:15 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:09:15 +1100
Subject: [Freeswitch-users] ZRTP protocol measurement
In-Reply-To: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>
References: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>
Message-ID: <20100401070915.GB13609@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> I would like to know how to measure the performence of the zrtp using
> FreeSWITCH that connect two X-Lite softphones which use the zfone for
> encrypting voice packets on both end phones.

What do you want to know? There are almost as many performance measurements as
there are people doing the measuring. I would suggest measuring in whatever
way you normally would, comparing ZRTP sessions with sessions that do not
involve ZRTP, and seeing if there are performance differences that affect your
usage scenario.
> 
> Also, can someone contrast and compare ZRTP to SRTP focusing on these two
> protocol behaviors.

Have a look at http://www.zfone.com/ for a description of ZRTP. From an
operational perspective, the main difference is that in configuring SRTP, you
need to use TLS to secure the SIP signaling; otherwise, the cryptographic keys
are transmitted in the clear, which completely eliminates the security.
Setting up TLS securely requires a public-key infrastructure whereby each side
verifies the identity of the other.

In ZRTP, the negotiation takes place entirely in the RTP stream; there are
several protection mechanisms provided to prevent third-parties from
masquerading as one of the end-points (namely, key finger-prints, displayed to
the user as words that can be verified in the conversation, and
the use of mathematically related keys in subsequent sessions between the same
parties, but without diminishing security). No public-key infrastructure is
needed, hence no X.509 certificates or TLS are required.



From jason at jasonjgw.net  Thu Apr  1 00:13:48 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:13:48 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <20100329040146.GA19949@jdc.jasonjgw.net>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
	<4BAF7871.3020909@aktzero.com>
	<20100329040146.GA19949@jdc.jasonjgw.net>
Message-ID: <20100401071348.GA13720@jdc.jasonjgw.net>

Jason White <jason at jasonjgw.net> wrote:
 
> http://pastebin.freeswitch.org/12565
> with SILK at 24000h specified in my configuration as the only codec, and after
> having verified that mod_silk was loaded. All of the SILK codecs (8000, 12000,
> etc., including 24000) were logged as having been registered.
> 
> I didn't notice anything interesting in the logs that I've posted other than
> the 48 khz codec being offered, presumably due to the fact that mod_portaudio
> was configured for 48 khz. Note that this always works perfectly well with a
> variety of codecs; FreeSWITCH resamples the input, as necessary, prior to
> encoding it.

Following a suggestion by Frank Carmickle, I confirmed that the profile status
shows SILK as the first codec in both inbound and outbound lists. It still
isn't showing up in the SIP request, though, so the problem remains exactly as
described earlier in this thread.



From lists at infosecurity.ch  Thu Apr  1 02:32:20 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 01 Apr 2010 11:32:20 +0200
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
	text-to-speech of SaS
Message-ID: <4BB46824.6060905@infosecurity.ch>

Hi all,

i would like to call from the product my company is going to release 
(www.privatewave.com, mobile voice encryption with ZRTP for Nokia, 
iPhone, Blackberry, Android) a FS extension and, without the ZRTP 
enrollment, having the extension to be answered negotiating ZRTP and 
playing with Text to Speech the SaS .

Is this possible with current FS/ZRTP integration?

Fabio



From a.afzali2003 at gmail.com  Thu Apr  1 02:33:10 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 1 Apr 2010 14:03:10 +0430
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
Message-ID: <g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>

same issue for me ( also with error in updating openzap directory), I just
used :

make clean
make

-- afshin

On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hello,
>
> Trying to update the svn source.
>
> svn up
>
> make current
>
> I got the following Errors.
>
>
> cd libs/openzap && autoconf
> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>       If this token and others are legitimate, please use m4_pattern_allow.
>       See the Autoconf documentation.
> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
> make: *** [libs/openzap/Makefile] Error 1
>
>
> How to fix the make error.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  1 02:47:45 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 20:47:45 +1100
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <4BB46824.6060905@infosecurity.ch>
References: <4BB46824.6060905@infosecurity.ch>
Message-ID: <20100401094745.GA15156@jdc.jasonjgw.net>

Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:
> Hi all,
> 
> i would like to call from the product my company is going to release 
> (www.privatewave.com, mobile voice encryption with ZRTP for Nokia, 
> iPhone, Blackberry, Android) a FS extension and, without the ZRTP 
> enrollment, having the extension to be answered negotiating ZRTP and 
> playing with Text to Speech the SaS .
> 
> Is this possible with current FS/ZRTP integration?

Possibly, but you'll have to deal with licencing issues first associated with
the ZRTP library. It's under the GNU AGPLv3 licence, as I recall.

ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
it. If there's no ZRTP support on the other end, it just continues as an
unencrypted call.



From lists at infosecurity.ch  Thu Apr  1 02:58:48 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 01 Apr 2010 11:58:48 +0200
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <20100401094745.GA15156@jdc.jasonjgw.net>
References: <4BB46824.6060905@infosecurity.ch>
	<20100401094745.GA15156@jdc.jasonjgw.net>
Message-ID: <4BB46E58.8060308@infosecurity.ch>

On 01/04/10 11.47, Jason White wrote:
> ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
> it. If there's no ZRTP support on the other end, it just continues as an
> unencrypted call.
>    
So if a call is ZRTP enabled FS automatically negotiate it.
Is there already some prototype and/or support to extract the negotiated 
SaS and do a text to speech?

I would like to arrange a "test echo service" that's ZRTP enabled.

Fabio



From jason at jasonjgw.net  Thu Apr  1 03:20:36 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 21:20:36 +1100
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <4BB46E58.8060308@infosecurity.ch>
References: <4BB46824.6060905@infosecurity.ch>
	<20100401094745.GA15156@jdc.jasonjgw.net>
	<4BB46E58.8060308@infosecurity.ch>
Message-ID: <20100401102036.GA15452@jdc.jasonjgw.net>

Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:
> On 01/04/10 11.47, Jason White wrote:
> > ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
> > it. If there's no ZRTP support on the other end, it just continues as an
> > unencrypted call.
> >    
> So if a call is ZRTP enabled FS automatically negotiate it.

That's what ZRTP is designed to do.
> Is there already some prototype and/or support to extract the negotiated 
> SaS and do a text to speech?

The SAS is available in a variable - I can't remember the details now, and the
text to speech should be easy at that point.




From nagalenoj at gmail.com  Thu Apr  1 03:37:02 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 1 Apr 2010 16:07:02 +0530
Subject: [Freeswitch-users] Grouping channels in span
Message-ID: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>

Dear friends,
    I'm using Sangoma A102 and I've configured it with wanpipe. I've
question with regard to grouping the spans. There are 2 spans in the card
and I would want to group the spans separately, so when I want to make call
I could specify as either openzap/1/ or openzap/2/...
    But, Now I'm unable to make calls like this. It is asking me to give the
dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..

    I don't want this way., I would want to group the spans differently.
Kindly help me.

-- 
Regards,
Nagalenoj H.
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From moises.silva at gmail.com  Thu Apr  1 06:04:05 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 1 Apr 2010 09:04:05 -0400
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
Message-ID: <i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>

There can be only 1 boost span currently. All b-channels must be added to
that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
belong to which group. You can see how to configure spans, channels and
groups here:

http://wiki.sangoma.com/wanpipe-pri-advanced-options

<http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing from
FreeSWITCH use the syntax requested:

openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
openzap/1/1234 at g2 to dial using group 2 and so on.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
> question with regard to grouping the spans. There are 2 spans in the card
> and I would want to group the spans separately, so when I want to make call
> I could specify as either openzap/1/ or openzap/2/...
>     But, Now I'm unable to make calls like this. It is asking me to give
> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>
>     I don't want this way., I would want to group the spans differently.
> Kindly help me.
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cucku.cucku at yahoo.com.vn  Thu Apr  1 07:17:48 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 1 Apr 2010 22:17:48 +0800 (SGT)
Subject: [Freeswitch-users]  :  need help on call to gateway
In-Reply-To: <c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
Message-ID: <384781.44441.qm@web76212.mail.sg1.yahoo.com>

Hi Francios 

Thank you for you recommend

from your recommendation, i add the host name : sip.yeah.com into the 
/etc/hosts

but i am still getting error on DNS 
 Failed with status DNS Error [503]

it seems that FS needs resolve the sip.yeah.com from the DNS server.
the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??

i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name

[root at localhost external]# nslookup sip.yeah.com
Server:         8.8.8.8
Address:        8.8.8.8#53

** server can't find sip.yeah.com: NXDOMAIN

[root at localhost external]# ping sip.yeah.com
PING sip.yeah.com (118.69.239.250) 56(84) bytes of data.
64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms

Thank you
Ha`


________________________________
T?: Fran?ois Legal <devel at thom.fr.eu.org>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 20:51:17, Th? T?, 31 th?ng 3 2010 
Ch? ??: Re: [Freeswitch-users] V?:  V?:  V?:  need help on call to gateway


Then either provide DNS to that system so that it can resolve the address, either create a static binding in /etc/hosts file on that host for sip.yeah.com
 
On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false <cucku.cucku at yahoo.com.vn> wrote:
Hi David
>
>when change the 
>the Register message send out with wrong format : from sip:071234 at x.x.x.x
>the right format : sip:071234 at sip.yeah.com
>
>is there any way to fix it
>
>Thank you
>
>
>
>
>
________________________________
 T?: David Ponzone 
>??n: freeswitch-users at lists.freeswitch.org
>G?i ng?y: 15:50:00, Th? T?, 31 th?ng 3 2010 
>Ch? ??: Re: [Freeswitch-users] V?:  V?:  need help on call to gateway
>
>I guess you need to put the IP of the proxy also in:
>
>   
>David Ponzone  Direction Technique
>email: david.ponzone at ipeva.fr
>tel:      01 74 03 18 97
>gsm:   06 66 98 76 34
>
>
>Service Client IPeva
>tel:      0811 46 26 26
>www.ipeva.fr  -   www.ipeva-studio.com
>
>
>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>Le 31/03/2010 ? 10:22, false a ?crit :
>
>Hi David
>>
>>the domain sip.yeah.com cannot resolve in the FS host
>>the thing here is FW support  for Registration to remote SIP server
>>so the domain no need to resolve
>>how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,
>>
>>Thank you
>>
>>
>>
>>
________________________________
 T?: David Ponzone <david.ponzone at gmail.com>
>>??n: freeswitch-users at lists.freeswitch.org
>>G?i ng?y: 15:12:50, Th? T?, 31 th?ng 3 2010 
>>Ch? ??: Re: [Freeswitch-users] V?: need help on call to gateway
>>
>>This looks like a DNS error.
>>
>>Are you sure you can resolve sip.yeah.com from the FS host ?
>>
>>
>>David Ponzone  Direction Technique
>>email: david.ponzone at ipeva.fr
>>tel:      01 74 03 18 97
>>gsm:   06 66 98 76 34
>>
>>
>>Service Client IPeva
>>tel:      0811 46 26 26
>>www.ipeva.fr  -   www.ipeva-studio.com
>>
>>
>>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>Le 31/03/2010 ? 09:51, false a ?crit :
>>
>>Hi David and Brian
>>>
>>>sorry to bother you again
>>>the domain name : sip.yeah.com is not real, How to fix the issue
>>>
>>>FreeSwitch register to sip server with user 071234 successull
>>>FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
>>>the extension make call out using the sip server is fail
>>>and the error :
>>>sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>>>nta: for  "sip.yeah.com" query "sip.yeah.com" A (cached)
>>>nua(0x9473108): call state changed: init -> calling, sent offer
>>>soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
>>>nua(0x9473108): event i_state INVITE sent
>>>nua(0x9473108): event r_invite 503 DNS Error
>>>nua(0x9473108): call state changed: calling -> init
>>>nua(0x9473108): event i_state 503 DNS Error
>>>nua(0x9473108): event i_terminated 503 DNS Error
>>>
>>>
>>>i create 3 file:
>>>    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
>>>    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
>>>    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call
>>>
>>>the content of 071234.xml  :
>>>
>>> 
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>> 
>>>
>>>
>>>the content of 00_inbound_did_fpt.xml:
>>>
>>>  
>>>    
>>>      
>>>      
>>>    
>>>  
>>>
>>>
>>>the content of  01_fpt.net.xml:
>>>
>>>  
>>>    
>>>      
>>>    
>>>  
>>>
>>>
>>>
>>>
________________________________
 T?: David Ponzone <david.ponzone at gmail.com>
>>>??n: freeswitch-users at lists.freeswitch.org
>>>G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010 
>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>
>>>False,
>>>
>>>I think Brian is sleeping so I give you a quick answer.
>>>For OUTBOUND:
>>>Following my previous mail, you have to check if your user 1000 has the default context  configured.
>>>If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
>>>(You may also remove from this path the remaining .xml files from the default install)
>>>In 01_whatever.xml, put:
>>>    
>>>      
>>>        
>>>      
>>>    
>>>
>>>So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
>>>For INBOUND:
>>>In conf/dialplan/public/, add a file named whatever.xml with the following content:
>>>   
>>>      
>>>        
>>>      
>>>    
>>>
>>>Here, YOUR_DID is the DID number your provider allocated to you.
>>>That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.
>>>Of course, after doing all this, reloadxml.
>>>It will perhaps not fit in your  config exactly, but you should get the idea to get started.
>>>
>>>
>>>David Ponzone  Direction Technique
>>>email: david.ponzone at ipeva.fr
>>>tel:      01 74 03 18 97
>>>gsm:   06 66 98 76 34
>>>
>>>
>>>Service Client IPeva
>>>tel:      0811 46 26  26
>>>www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>
>>>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>>
>>>
>>>Le 30/03/2010 ? 09:01, false a ?crit :
>>>
>>>Hi Brian
>>>>
>>>>could you guide me more to config on freeswitch
>>>>
>>>>i want xilte with user 1000 can make call out to sip server 
>>>>and the prefix when make call out to sip server : 0
>>>>and receive incomming call from SIP server
>>>>
>>>>what file name i should create 
>>>>
>>>>Thank you
>>>>
>>>>
>>>>--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:
>>>>
>>>>
>>>>>T?: Brian West <brian at freeswitch.org>
>>>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>>>??n: freeswitch-users at lists.freeswitch.org
>>>>>Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>>>>
>>>>>
>>>>>its looking in context default... not public.
>>>>>
>>>>>/b
>>>>>
>>>>>
>>>>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>>>>
>>>>>Hi all
>>>>>>
>>>>>>i use freeswitch to register to sip server
>>>>>>the domain is not real: sip.yeah.com
>>>>>>the outbound proxy  : 118.69.145.5
>>>>>>
>>>>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>>>>>
>>>>>>    sip.yeah.com">
>>>>>>        
>>>>>>            
>>>>>>        sip.yeah.com"/>
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>    
>>>>>>
>>>>>>
>>>>>>so i use xlite register to freeswitch with user 1000
>>>>>>i start freeswitch , freeswitch start ok,
>>>>>>freeswitch register to sip server successfull with username 071234
>>>>>>xlite register to freeswitch successfull
>>>>>>
>>>>>>so i edit the public.xml
>>>>>>for incomming call from sip server, i will forward to extension 1000
>>>>>>    sip.yeah.com">
>>>>>>      
>>>>>>        
>>>>>>      
>>>>>>    
>>>>>>
>>>>>>but when i use  xlite make call out from freeswitch to sip server and get no routing
>>>>>> 
>>>>> 
>>>>________________________________
 >>>>T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo! Mail m?i h?m nay!_______________________________________________
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>________________________________
 Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
>>>T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?._______________________________________________
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>________________________________
 >>Yahoo! Mail nay NHANH H?N - Th? ngay!_______________________________________________
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>________________________________
 >Yahoo! Mail nay nhanh v? nhi?u kh?ng gian tho?ng h?n.H?y tr?i nghi?m ngay h?m nay! 


      Xem h?nh c? ?m m?i v?i ?ng B?t v? c? T?m?! T?i phi?n b?n Yahoo! Messenger m?i nh?t b?ng ti?ng Vi?t t?i ??y.
http://vn.messenger.yahoo.com
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From brian at freeswitch.org  Thu Apr  1 07:37:15 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Apr 2010 09:37:15 -0500
Subject: [Freeswitch-users] :  need help on call to gateway
In-Reply-To: <384781.44441.qm@web76212.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
Message-ID: <EE7D0215-5EE3-4D11-9F4E-C2500C129A37@freeswitch.org>

Why not get a provider that actually does this stuff correctly?  My blood boils ever time I hear of this stupid setup.

/b

On Apr 1, 2010, at 9:17 AM, false wrote:

> Hi Francios 
> 
> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]
> 
> it seems that FS needs resolve the sip.yeah.com from the DNS server.
> the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??
> 
> i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name
> 
> [root at localhost external]# nslookup sip.yeah.com
> Server:         8.8.8.8
> Address:        8.8.8.8#53
> 
> ** server can't find sip.yeah.com: NXDOMAIN
> 
> [root at localhost external]# ping sip.yeah.com
> PING sip.yeah.com (118.69.239.250) 56(84) bytes of data.
> 64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms
> 
> Thank you
> Ha`

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From 12ukwn at gmail.com  Thu Apr  1 07:50:05 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Thu, 1 Apr 2010 16:50:05 +0200
Subject: [Freeswitch-users] :  need help on call to gateway
In-Reply-To: <384781.44441.qm@web76212.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
Message-ID: <20100401165005.43ad63d1@anubis.defcon1>

Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT),
false <cucku.cucku at yahoo.com.vn> a ?crit :

> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the 
> /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]

This is because /etc/host.conf isn't configured to first pick a name
into /etc/hosts before a DNS resolution occurs (man host.conf)

But it is usually better to create a specific DNS zone, such as:

; Zone for *.fusionpbx.set
$TTL	1D
$ORIGIN	fusionpbx.set.
;
@		   IN   SOA   fusionpbx.set.   hostmaster.fusionpbx.set. ( 
			2010030401	; serial
				8H	; refresh
				4H	; retry
				4W	; expire
				1D);
; DNS names
		   IN   NS   ns1.fusionpbx.set. 
		   IN   NS   ns2.fusionpbx.set.
; MX
		   IN   MX   10   mail.fusionpbx.set.
; Virtual Machines & Hosts
		   IN    A   192.168.1.25
ns1		   IN    A   192.168.1.1
ns2		   IN    A   192.168.1.2
mail		   IN    A   192.168.1.50
*.fusionpbx.set.   IN    A   192.168.1.25

This way, you can address any subdomain (but may be that's not what you're
looking for?)


-- 
"Take that, you hostile sons-of-bitches!"
		-- James Coburn, in the finale of _The_President's_Analyst_


From msc at freeswitch.org  Thu Apr  1 09:53:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Apr 2010 09:53:23 -0700
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
Message-ID: <m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>

Not sure if you have a broken source tree or what. Try doing:
svn up && ./bootstrap.sh && ./configure && make install

That's a bit drastic I know but I've used it before to clear out goofy build
errors.
-MC

On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> same issue for me ( also with error in updating openzap directory), I just
> used :
>
> make clean
> make
>
> -- afshin
>
> On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hello,
>>
>> Trying to update the svn source.
>>
>> svn up
>>
>> make current
>>
>> I got the following Errors.
>>
>>
>> cd libs/openzap && autoconf
>> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>>       If this token and others are legitimate, please use
>> m4_pattern_allow.
>>       See the Autoconf documentation.
>> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
>> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
>> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
>> make: *** [libs/openzap/Makefile] Error 1
>>
>>
>> How to fix the make error.
>>
>> Thanks
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Thu Apr  1 10:12:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 1 Apr 2010 13:12:29 -0400
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
Message-ID: <x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>

Hi Michael,

Yes I follow the same steps to solve my problem.

BTW I have another question, When the will be the migration to Git complete?

SVN or GIT which one we need to choose?

Thanks
Lloyd


On Thu, Apr 1, 2010 at 12:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Not sure if you have a broken source tree or what. Try doing:
> svn up && ./bootstrap.sh && ./configure && make install
>
> That's a bit drastic I know but I've used it before to clear out goofy
> build errors.
> -MC
>
>
> On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> same issue for me ( also with error in updating openzap directory), I just
>> used :
>>
>> make clean
>> make
>>
>> -- afshin
>>
>> On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> Hello,
>>>
>>> Trying to update the svn source.
>>>
>>> svn up
>>>
>>> make current
>>>
>>> I got the following Errors.
>>>
>>>
>>> cd libs/openzap && autoconf
>>> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>>>       If this token and others are legitimate, please use
>>> m4_pattern_allow.
>>>       See the Autoconf documentation.
>>> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
>>> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
>>> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
>>> make: *** [libs/openzap/Makefile] Error 1
>>>
>>>
>>> How to fix the make error.
>>>
>>> Thanks
>>> Lloyd
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  1 10:38:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Apr 2010 10:38:46 -0700
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
	<x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
Message-ID: <v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>

On Thu, Apr 1, 2010 at 10:12 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi Michael,
>
> Yes I follow the same steps to solve my problem.
>
> BTW I have another question, When the will be the migration to Git
> complete?
>
> SVN or GIT which one we need to choose?
>
> Thanks
> Lloyd
>
> You really only need git if you plan on having commit access. There is an
SVN mirror, so you don't really have to change anything if all you ever do
is download FS to update your system.
-MC
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From fs-list at communicatefreely.net  Thu Apr  1 18:52:29 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 01 Apr 2010 21:52:29 -0400
Subject: [Freeswitch-users] mod_local_stream path based on rate
In-Reply-To: <191c3a031003290916o3b524d95p51eb637255afec20@mail.gmail.com>
References: <4BB0CF2F.7070700@communicatefreely.net>
	<191c3a031003290916o3b524d95p51eb637255afec20@mail.gmail.com>
Message-ID: <4BB54DDD.5010806@communicatefreely.net>

That did it!

Thanks, I'll try to get this in the Wiki this week for anyone else that is interested.

-Tim

Anthony Minessale wrote:
> the default config works this way
> it runs a stream on each rate called moh/8000 moh/16000 moh/32000 and
> moh/48000
> if you try to run stream "moh" it will pick the right one.
> 
> 
> On Mon, Mar 29, 2010 at 11:02 AM, Tim St. Pierre
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> wrote:
> 
>     Hello,
> 
>     I'm using local_stream:// as our music on hold source.  I have both
>     8000 KHz and 16000 KHz files
>     encoded so that I can play them at the native rate.  We have some
>     endpoints that can only do PCMU
>     and others that can do higher rates.
> 
>     I want to set music on hold for a channel and have it automatically
>     pick the appropriate stream
>     based on the channel's native rate, just like the voice prompts do.
>      I tried creating a stream for
>     each rate, like rock-8000 and rock-16000 and then set the channel's
>     music on hold variable to
>     local_stream://rock-${read_rate} but the read rate hasn't been
>     negotiated when this is set in the
>     dialplan.
> 
>     Any suggestions?
> 
>     -Tim
> 
>     _______________________________________________
>     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/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From nagalenoj at gmail.com  Thu Apr  1 21:49:11 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 2 Apr 2010 10:19:11 +0530
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
	<i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
Message-ID: <l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>

Moises,
      In all cases, I should use openzap/1/a/123 and there is no chance for
configuring like openzap/1 and openzap/2. Am I right?

      The difficulty I face here is, In dialplan I'm unable to route the
calls based on spans, like,
   <condition field="chan_name" expression="^OpenZAP/1.*/(.*)">
       <action application="bridge" data="openzap/3/a/$1"/>
   </condition>
   <condition field="chan_name" expression="^OpenZAP/2.*/(.*)">
       <action application="playback" data="under_maintenance.wav"/>
   </condition>

       Is there any other way to right the expression to match the spans?

On Thu, Apr 1, 2010 at 6:34 PM, Moises Silva <moises.silva at gmail.com> wrote:

> There can be only 1 boost span currently. All b-channels must be added to
> that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
> belong to which group. You can see how to configure spans, channels and
> groups here:
>
> http://wiki.sangoma.com/wanpipe-pri-advanced-options
>
> <http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing
> from FreeSWITCH use the syntax requested:
>
> openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
> openzap/1/1234 at g2 to dial using group 2 and so on.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
>> question with regard to grouping the spans. There are 2 spans in the card
>> and I would want to group the spans separately, so when I want to make call
>> I could specify as either openzap/1/ or openzap/2/...
>>     But, Now I'm unable to make calls like this. It is asking me to give
>> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>>
>>     I don't want this way., I would want to group the spans differently.
>> Kindly help me.
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
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From infos at madovsky.org  Thu Apr  1 10:17:43 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 1 Apr 2010 13:17:43 -0400
Subject: [Freeswitch-users] svn up then make current - Errors
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com><g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
Message-ID: <85CCB9D6A5514599801186E0D6558000@MOBILEE1705>

same issue for me.
needed to comment out openzap in modules.conf (don't need it)


  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, April 01, 2010 12:53 PM
  Subject: Re: [Freeswitch-users] svn up then make current - Errors


  Not sure if you have a broken source tree or what. Try doing:
  svn up && ./bootstrap.sh && ./configure && make install

  That's a bit drastic I know but I've used it before to clear out goofy build errors.
  -MC


  On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com> wrote:

    same issue for me ( also with error in updating openzap directory), I just used :

    make clean
    make

    -- afshin


    On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:

      Hello,


      Trying to update the svn source.


      svn up


      make current


      I got the following Errors.




      cd libs/openzap && autoconf
      configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
            If this token and others are legitimate, please use m4_pattern_allow.
            See the Autoconf documentation.
      configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
      configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
      configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
      make: *** [libs/openzap/Makefile] Error 1




      How to fix the make error.


      Thanks
      Lloyd

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://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
    http://www.freeswitch.org






------------------------------------------------------------------------------


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From grsingh750 at gmail.com  Thu Apr  1 13:52:08 2010
From: grsingh750 at gmail.com (guru singh)
Date: Fri, 2 Apr 2010 02:22:08 +0530
Subject: [Freeswitch-users] PSTN Integration and real deployments
Message-ID: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>

Hi,
After long nights and lots of coffee =) , I think I've largely understood
FreeSwitch. I've been playing with it and have managed most fancy things it
can do. But I've done this on my LAN using SIP softphones. Here's my problem
now, I know nothing about PSTN integration and real deployments. Here are my
questions, mostly based on what I read on wikipedia.

PSTN integration:

I have an ADSL internet connection, with a split-box? installed by my ISP
which splits the incoming line to two, one for the phone provided and one
for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
also be able to outbound calls to PSTN/VoIP phones via an SIP client
registered with my FS server through an external gateway or the PSTN line.

0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
seamlessly converts SIP-PSTN traffic both ways or does it require
configuration? What are the ATA's that work best with FS?

1) I should register with a VoIP/SIP/DID? provider for making outbound
calls? Will I be provided with an incoming number reachable by normal PSTN
numbers? If yes, where will the number reside, as in will PSTN numbers
calling me be charged extra?

Real Deployments:

Supposing I'm to do a real deployment for a client. What are the options
that I have for hardware?

0) Get IP phones that talk SIP? Is this the most expensive option?

1) Suppose the client has a traditional plain intercom installment(think
hotels etc). with phones connecting via RJ11 connectors. Is it possible to
have something like an ATA with lots of ports working as a hub/switch, So
that all phones can be plugged into ATA and managed via FS?

Thanks

PS: If the above hardly makes sense, pardon me, you can understand my
confusion =). FS has really got me hooked and I'm itching to do more with
it.
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From nandy1925 at gmail.com  Thu Apr  1 22:33:52 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 2 Apr 2010 13:33:52 +0800
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
Message-ID: <q2s7d0bfd8c1004012233u6750bd89n2cab7cbc36dca0f2@mail.gmail.com>

hi guru,

re pstn integration:
0) get an ATA. connect the FXS port to an analog phone and the FXO port to
your phone line. set the FXO port to dial (send INVITE) to Freeswitch after
1 or 2 rings. you can create an Auto-Attendant on FS to handle this call.
you can control the route of outgoing calls (PSTN or VoIP) via dialplan.

i hv tried Grandstream HT-503 but the FXO port has problems. i'd like to get
Audiocodes. grab their User Manual to give you an idea.

1) you hv to register w/ a VoIP provider if u route your calls via Internet.
the incoming number is usually offered as an option. re charges. it depends
on the plan you get.

re real deployments:

0) you can mix the client phones - IP hardphones or via FXS gateways (4/8/24
ports) to connect analog phones
1) yes. that's multi-port FXS gateways. there are multiport FXO gateways
where you can place FS to handle PSTN calls for the legacy PABX

PSTN <==> FS <==> PABX

i hope it clears up a bit from your cloud confusion :-)

-nandy

On Fri, Apr 2, 2010 at 4:52 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
> After long nights and lots of coffee =) , I think I've largely understood
> FreeSwitch. I've been playing with it and have managed most fancy things it
> can do. But I've done this on my LAN using SIP softphones. Here's my problem
> now, I know nothing about PSTN integration and real deployments. Here are my
> questions, mostly based on what I read on wikipedia.
>
> PSTN integration:
>
> I have an ADSL internet connection, with a split-box? installed by my ISP
> which splits the incoming line to two, one for the phone provided and one
> for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
> also be able to outbound calls to PSTN/VoIP phones via an SIP client
> registered with my FS server through an external gateway or the PSTN line.
>
> 0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
>
> 1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal PSTN
> numbers? If yes, where will the number reside, as in will PSTN numbers
> calling me be charged extra?
>
> Real Deployments:
>
> Supposing I'm to do a real deployment for a client. What are the options
> that I have for hardware?
>
> 0) Get IP phones that talk SIP? Is this the most expensive option?
>
> 1) Suppose the client has a traditional plain intercom installment(think
> hotels etc). with phones connecting via RJ11 connectors. Is it possible to
> have something like an ATA with lots of ports working as a hub/switch, So
> that all phones can be plugged into ATA and managed via FS?
>
> Thanks
>
> PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more with
> it.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  1 22:42:14 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 16:42:14 +1100
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
	<x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
	<v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>
Message-ID: <20100402054214.GA15808@jdc.jasonjgw.net>

Michael Collins <msc at freeswitch.org> wrote:
> > You really only need git if you plan on having commit access. There is an
> SVN mirror, so you don't really have to change anything if all you ever do
> is download FS to update your system.

Correct. If, however, you modify packaging or make other changes before
building, it should be easier with Git as you can establish your own private
branches, merge changes and access the entire history of the repository.

I've cloned the git repository but I haven't tried building FreeSWITCH from it
yet. I modify the Debian package files before building to enable certain
modules, set the version number appropriately, and so on.



From casteven at gmail.com  Thu Apr  1 23:24:15 2010
From: casteven at gmail.com (Campbell Steven)
Date: Fri, 02 Apr 2010 19:24:15 +1300
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
Message-ID: <1270189455.2845.4055.camel@macmini>

Hi Guru,

If you have a bunch of analog extensions from say an old PBX
installation that you want to reuse (I *think* this is what you are
getting at?) then you can use a channelbank like these:

http://www.patton.com/products/pe_products.asp?category=406

Which essentially are an up to 32 port ATA, like Nandy says, an ATA's
purpose is to allow you to use an analogue handset on a VoIP system.
This would allow you to have each individual analogue extension
registering to it's own SIP account.

Campbell


On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:

> Hi,
> 
> After long nights and lots of coffee =) , I think I've largely
> understood FreeSwitch. I've been playing with it and have managed most
> fancy things it can do. But I've done this on my LAN using SIP
> softphones. Here's my problem now, I know nothing about PSTN
> integration and real deployments. Here are my questions, mostly based
> on what I read on wikipedia.
> 
> 
> PSTN integration:
> 
> 
> I have an ADSL internet connection, with a split-box? installed by my
> ISP which splits the incoming line to two, one for the phone provided
> and one for the adsl modem. I want to handle incoming PSTN calls via
> FreeSwitch and also be able to outbound calls to PSTN/VoIP phones via
> an SIP client registered with my FS server through an external gateway
> or the PSTN line.
> 
> 
> 0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
> 
> 
> 1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal
> PSTN numbers? If yes, where will the number reside, as in will PSTN
> numbers calling me be charged extra?
> 
> 
> Real Deployments:
> 
> 
> Supposing I'm to do a real deployment for a client. What are the
> options that I have for hardware?
> 
> 
> 0) Get IP phones that talk SIP? Is this the most expensive option?
> 
> 
> 1) Suppose the client has a traditional plain intercom
> installment(think hotels etc). with phones connecting via RJ11
> connectors. Is it possible to have something like an ATA with lots of
> ports working as a hub/switch, So that all phones can be plugged into
> ATA and managed via FS?
> 
> 
> Thanks
> 
> 
> PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more
> with it.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From nandy1925 at gmail.com  Fri Apr  2 00:44:42 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 2 Apr 2010 15:44:42 +0800
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <1270189455.2845.4055.camel@macmini>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
	<1270189455.2845.4055.camel@macmini>
Message-ID: <m2l7d0bfd8c1004020044q35e01de7jca433d843437a598@mail.gmail.com>

IP Channel Bank is new to me. i thought, all the while, that channel banks
uses expensive  E1/T1 interface.  now, it widens my options. tks for this
info.


On Fri, Apr 2, 2010 at 2:24 PM, Campbell Steven <casteven at gmail.com> wrote:

>  Hi Guru,
>
> If you have a bunch of analog extensions from say an old PBX installation
> that you want to reuse (I *think* this is what you are getting at?) then you
> can use a channelbank like these:
>
> http://www.patton.com/products/pe_products.asp?category=406
>
> Which essentially are an up to 32 port ATA, like Nandy says, an ATA's
> purpose is to allow you to use an analogue handset on a VoIP system. This
> would allow you to have each individual analogue extension registering to
> it's own SIP account.
>
> Campbell
>
>
>
> On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:
>
> Hi,
>
>  After long nights and lots of coffee =) , I think I've largely understood
> FreeSwitch. I've been playing with it and have managed most fancy things it
> can do. But I've done this on my LAN using SIP softphones. Here's my problem
> now, I know nothing about PSTN integration and real deployments. Here are my
> questions, mostly based on what I read on wikipedia.
>
>
>
>  PSTN integration:
>
>
>
>  I have an ADSL internet connection, with a split-box? installed by my ISP
> which splits the incoming line to two, one for the phone provided and one
> for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
> also be able to outbound calls to PSTN/VoIP phones via an SIP client
> registered with my FS server through an external gateway or the PSTN line.
>
>
>
>  0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
>
>
>
>  1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal PSTN
> numbers? If yes, where will the number reside, as in will PSTN numbers
> calling me be charged extra?
>
>
>
>  Real Deployments:
>
>
>
>  Supposing I'm to do a real deployment for a client. What are the options
> that I have for hardware?
>
>
>
>  0) Get IP phones that talk SIP? Is this the most expensive option?
>
>
>
>  1) Suppose the client has a traditional plain intercom installment(think
> hotels etc). with phones connecting via RJ11 connectors. Is it possible to
> have something like an ATA with lots of ports working as a hub/switch, So
> that all phones can be plugged into ATA and managed via FS?
>
>
>
>  Thanks
>
>
>
>  PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more with
> it.
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-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 vfclists at googlemail.com  Fri Apr  2 02:12:25 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 10:12:25 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on Freeswitch
Message-ID: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>

I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.

What configuration changes do I need to allow freeswitch-codec-passthru-g729.

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From jason at jasonjgw.net  Fri Apr  2 02:23:10 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 20:23:10 +1100
Subject: [Freeswitch-users] How to configure g729 pass though on
 Freeswitch
In-Reply-To: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
Message-ID: <20100402092310.GA18680@jdc.jasonjgw.net>

Frank Church <vfclists at googlemail.com> wrote:
> I am just trialling Freeswitch with Linksys adapters, whose default
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> When I change that setting to 'yes' the calls don't go through. I am
> using the latest Windows SVN.

FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.

My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.



From gavin.henry at gmail.com  Fri Apr  2 02:43:51 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 10:43:51 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT phones on
	FreeSWITCH Version 1.0.trunk (17177)
Message-ID: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>

Hi,

We've just upgraded (switched back to building from svn in case this
was a git migration issue) and all our phones behind various networks
on NAT won't registered anymore, whereas on version dated 2010-02-22
20:06 (don't have the revision) all these phones were fine and handled
the the FS NAT features. No NDLB features on enabled or rport. They
weren't on the 2010-02-22 20:06 version either.


freeswitch at internal> recv 524 bytes from udp/[external_ip]:43078 at
09:45:02.413013:
   ------------------------------------------------------------------------
   REGISTER sip:pbx1.xxxx.co.uk SIP/2.0
   Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790
   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=706763496
   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>
   Call-ID: 1553007081 at internal_ip
   CSeq: 24 REGISTER
   Contact: <sip:609 at internal_ip:5062>
   Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
   Max-Forwards: 70
   User-Agent: Yealink SIP-T28P 2.43.0.50
   Expires: 3600
   Content-Length: 0

   ------------------------------------------------------------------------
send 641 bytes to udp/[external_ip]:5062 at 09:45:02.413206:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
internal_ip:5062;branch=z9hG4bK892744790;received=external_ip
   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=706763496
   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=gtDayy31FDN9K
   Call-ID: 1553007081 at internal_ip
   CSeq: 24 REGISTER
   User-Agent: Our test PBX 1.0
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="pbx1.xxxx.co.uk",
nonce="cf9aaa61-89bf-4d9d-adbb-51a068b23440", algorithm=MD5,
qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------


What's changed?

Thanks,

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jaybinks at gmail.com  Fri Apr  2 02:52:49 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 2 Apr 2010 19:52:49 +1000
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
Message-ID: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>

I know its not much help..
but I think the first thing you need to do here is identify the version you
were on before ..

and see if you can easily narrow it down to a SVN revision that broke it.

J



On Fri, Apr 2, 2010 at 7:43 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> We've just upgraded (switched back to building from svn in case this
> was a git migration issue) and all our phones behind various networks
> on NAT won't registered anymore, whereas on version dated 2010-02-22
> 20:06 (don't have the revision) all these phones were fine and handled
> the the FS NAT features. No NDLB features on enabled or rport. They
> weren't on the 2010-02-22 20:06 version either.
>
>
> freeswitch at internal> recv 524 bytes from udp/[external_ip]:43078 at
> 09:45:02.413013:
>   ------------------------------------------------------------------------
>   REGISTER sip:pbx1.xxxx.co.uk SIP/2.0
>   Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790
>   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=706763496
>   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>>
>   Call-ID: 1553007081 at internal_ip
>   CSeq: 24 REGISTER
>   Contact: <sip:609 at internal_ip:5062>
>   Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
> REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
>   Max-Forwards: 70
>   User-Agent: Yealink SIP-T28P 2.43.0.50
>   Expires: 3600
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> send 641 bytes to udp/[external_ip]:5062 at 09:45:02.413206:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP
> internal_ip:5062;branch=z9hG4bK892744790;received=external_ip
>   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=706763496
>   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=gtDayy31FDN9K
>   Call-ID: 1553007081 at internal_ip
>   CSeq: 24 REGISTER
>   User-Agent: Our test PBX 1.0
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="pbx1.xxxx.co.uk",
> nonce="cf9aaa61-89bf-4d9d-adbb-51a068b23440", algorithm=MD5,
> qop="auth"
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
>
>
> What's changed?
>
> Thanks,
>
> Gavin.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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
>



-- 
Sincerely

Jay
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From gavin.henry at gmail.com  Fri Apr  2 03:10:44 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:10:44 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.
> J

Yeah, I wish there was more than just an svn revision. Some kind of
other tag within the code or configs....

Thanks.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From gavin.henry at gmail.com  Fri Apr  2 03:18:42 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:18:42 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <w2m13ca621c1004020318hb5c7d65dl433429082a3dbdc2@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.
> J

The only phone that seems to get through is:

Agent:      	snom300/7.3.14

Where as all the others did too:

Yealink T28
Cisco 7940G
Polycom IP330
Aastra 31i
Aastra 51i
Aastra 53i
Aastra 55i
Aastra 57i


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jason at jasonjgw.net  Fri Apr  2 03:22:33 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 21:22:33 +1100
Subject: [Freeswitch-users] Registrations stop working for all NAT
 phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
Message-ID: <20100402102233.GA19211@jdc.jasonjgw.net>

Gavin Henry <gavin.henry at gmail.com> wrote:
 
> Yeah, I wish there was more than just an svn revision. Some kind of
> other tag within the code or configs....

When the migration to Git is settled, you'll be able to run git bisect, which
is designed for the purpose you're discussing. It isn't supposed to be a
substitute for debugging the code, but it's reputed to be used to track down
bugs which are hard to identify otherwise, in the Linux kernel in particular.




From gavin.henry at gmail.com  Fri Apr  2 03:56:12 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:56:12 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <20100402102233.GA19211@jdc.jasonjgw.net>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
	<20100402102233.GA19211@jdc.jasonjgw.net>
Message-ID: <v2p13ca621c1004020356l67be38dwd8fb36545e65df1@mail.gmail.com>

On 2 April 2010 11:22, Jason White <jason at jasonjgw.net> wrote:
> Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> Yeah, I wish there was more than just an svn revision. Some kind of
>> other tag within the code or configs....
>
> When the migration to Git is settled, you'll be able to run git bisect, which
> is designed for the purpose you're discussing. It isn't supposed to be a
> substitute for debugging the code, but it's reputed to be used to track down
> bugs which are hard to identify otherwise, in the Linux kernel in particular.

OK, sounds good!


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From vfclists at googlemail.com  Fri Apr  2 05:03:19 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 13:03:19 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <20100402092310.GA18680@jdc.jasonjgw.net>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
Message-ID: <g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>

On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
> Frank Church <vfclists at googlemail.com> wrote:
>> I am just trialling Freeswitch with Linksys adapters, whose default
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>> When I change that setting to 'yes' the calls don't go through. I am
>> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
> compelling reason, for example a carrier that only supports G.729.
>
>

The carrier insists on G729, although they can accept G711. I think
their call volume does not make it easy on them and their customers as
well.

I did some googling and came up with freeswitch-codec-passthru-g729. I
have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
http://wiki.freeswitch.org/wiki/Bypass_Media.
In my module.conf.xml there is also <load module="mod_g729"/>.

Does that mean that my installlation is configured for pass thru if I
make the right adjustments?

I have looked at
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
which speak of modifying the dialplan.

This is a basic freeswitch setup using the defaults. I just added the
extensions to conf/directory/default and changed the provider in
vars.xml and I want to be able to do the same in
conf/dialplan/default.xml.

In conf/dialplan/default.xml the extension is matched by the
destination. Is there an option for not falling through to other
extensions if they also match?

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



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From cucku.cucku at yahoo.com.vn  Fri Apr  2 05:14:10 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Fri, 2 Apr 2010 20:14:10 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?b?VuG7gTogIDogIG5lZWQgaGVscCBvbiBjYWxs?=
 =?utf-8?q?_to_gateway?=
In-Reply-To: <20100401165005.43ad63d1@anubis.defcon1>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
	<20100401165005.43ad63d1@anubis.defcon1>
Message-ID: <925478.50757.qm@web76215.mail.sg1.yahoo.com>

Hi Jean

yeah, your way solves my issue

Thank you





________________________________
T?: Jean-Yves F. Barbier <12ukwn at gmail.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 21:50:05, Th? N?m, 1 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] :  need help on call to gateway

Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT),
false <cucku.cucku at yahoo.com.vn> a ?crit :

> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the 
> /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]

This is because /etc/host.conf isn't configured to first pick a name
into /etc/hosts before a DNS resolution occurs (man host.conf)

But it is usually better to create a specific DNS zone, such as:

; Zone for *.fusionpbx.set
$TTL    1D
$ORIGIN    fusionpbx.set.
;
@           IN   SOA   fusionpbx.set.   hostmaster.fusionpbx.set. ( 
            2010030401    ; serial
                8H    ; refresh
                4H    ; retry
                4W    ; expire
                1D);
; DNS names
           IN   NS   ns1.fusionpbx.set. 
           IN   NS   ns2.fusionpbx.set.
; MX
           IN   MX   10   mail.fusionpbx.set.
; Virtual Machines & Hosts
           IN    A   192.168.1.25
ns1           IN    A   192.168.1.1
ns2           IN    A   192.168.1.2
mail           IN    A   192.168.1.50
*.fusionpbx.set.   IN    A   192.168.1.25

This way, you can address any subdomain (but may be that's not what you're
looking for?)


-- 
"Take that, you hostile sons-of-bitches!"
        -- James Coburn, in the finale of _The_President's_Analyst_

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



      Yahoo! Mail nay NHANH H?N - Th? ngay! http://vn.mail.yahoo.com
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From david.ponzone at gmail.com  Fri Apr  2 05:30:51 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 14:30:51 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
Message-ID: <EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>

Frank,

mod_g729 needs to be loaded, and then G729 needs to be negotiated on  
both legs.
I really recommend you enable G729 on the Linksys, enable SIP trace on  
FS console:
sofia profile external siptrace on
sofia profile internal siptrace on

then make a (failing) call, capture the log on FS console and paste  
that to:
http://pastebin:freeswitch at pastebin.freeswitch.org/

Then, send us back the link to your paste.

You can also join us on #freeswitch (irc.freenode.net), for some live  
help.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 02/04/2010 ? 14:03, Frank Church a ?crit :

> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>> Frank Church <vfclists at googlemail.com> wrote:
>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>> When I change that setting to 'yes' the calls don't go through. I am
>>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you really
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>
> The carrier insists on G729, although they can accept G711. I think
> their call volume does not make it easy on them and their customers as
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
> http://wiki.freeswitch.org/wiki/Bypass_Media.
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
> make the right adjustments?
>
> I have looked at
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
> and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
> extensions to conf/directory/default and changed the provider in
> vars.xml and I want to be able to do the same in
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
> destination. Is there an option for not falling through to other
> extensions if they also match?
>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 david.ponzone at gmail.com  Fri Apr  2 05:46:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 14:46:21 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
Message-ID: <06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>

Frank,

sorry, I completely forgot one important detail:
in the default conf, G729 is not allowed on any SIP profiles, so you  
have to modify vars.xml.
You will find the following lines:
   <X-PRE-PROCESS cmd="set"  
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>

For now, I recommend you replace them by:
   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>

Then in FS console, do:
sofia profile external restart reloadxml
sofia profile internal restart reloadxml

It should then work far better.

What we did there is to make G729 an accepted codec in inbound INVITEs  
and a proposed codec for outbound INVITEs.
Look in external.xml or internal.xml, and look at the variables  
inbound-codec-prefs and outbound-codec-prefs.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 02/04/2010 ? 14:30, David Ponzone a ?crit :

> Frank,
>
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on  
> both legs.
> I really recommend you enable G729 on the Linksys, enable SIP trace  
> on FS console:
> sofia profile external siptrace on
> sofia profile internal siptrace on
>
> then make a (failing) call, capture the log on FS console and paste  
> that to:
> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> Then, send us back the link to your paste.
>
> You can also join us on #freeswitch (irc.freenode.net), for some  
> live help.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>> Frank Church <vfclists at googlemail.com> wrote:
>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>> When I change that setting to 'yes' the calls don't go through. I  
>>>> am
>>>> using the latest Windows SVN.
>>>
>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>>> bypass media
>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>>> you really
>>> need it.
>>>
>>> My recommendation would be to use a codec other than G.729 unless  
>>> you have a
>>> compelling reason, for example a carrier that only supports G.729.
>>>
>>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>> their call volume does not make it easy on them and their customers  
>> as
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>> make the right adjustments?
>>
>> I have looked at
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>> and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>> extensions to conf/directory/default and changed the provider in
>> vars.xml and I want to be able to do the same in
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>> destination. Is there an option for not falling through to other
>> extensions if they also match?
>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> -- 
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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 lloyd.aloysius at gmail.com  Fri Apr  2 07:35:36 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 2 Apr 2010 10:35:36 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <191c3a031003231529w5e940113y899bcbfeff10d0eb@mail.gmail.com>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
	<F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
	<A643303450404497B14F79175BEC0EF4@MOBILEE1705>
	<005801cacad5$8a009820$9e01c860$@fr.eu.org>
	<191c3a031003231529w5e940113y899bcbfeff10d0eb@mail.gmail.com>
Message-ID: <u2s8a19bf2e1004020735kcb6be322o86556cdbb1a3f12b@mail.gmail.com>

After update the most recent version ,  voice mail to email stop working.

Here is the FreeSWITCH Environment

CentOS 5.4

FreeSWITCH
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17152)

Exim

---

Initially there is a sendmail problem then I start to using the Exim . Now
Exim also stop working.

What is the reliable method to deliver the Voice mail to Email.

Thanks in advance.

Lloyd




On Tue, Mar 23, 2010 at 6:29 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> lets open a jira about it as a reminder.
> we will figure out what to do there and come up with something.
>
>
>
> On Tue, Mar 23, 2010 at 5:09 PM, <devel at thom.fr.eu.org> wrote:
>
>>  So I did continue to track that problem.
>>
>>
>>
>> I found that when running sendmail from freeswitch, it seems that the
>> setgid bit on sendmail is not honored (I get messages like sendmail[13534]:
>> NOQUEUE: SYSERR(freeswitch): can not
>>
>>  chdir(/var/spool/mqueue-client/): Permission denied or NOQUEUE:
>> SYSERR(UID101): can not wri
>>
>> te to queue directory /var/spool/mqueue-client/ (RunAsGid=102,
>> required=107): Pe
>>
>> rmission denied when I gave freeswitch the permission on
>> /var/spool/mqueue-client).
>>
>> In any case that ends up to sendmail segfault, maybe due to a stack
>> problem.
>>
>>
>>
>> I could solve this by installing msmtp (it can live aside sendmail without
>> breaking my messaging setup, at least on debian)
>>
>>
>>
>> Fran?ois
>>
>>
>>
>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Madovsky
>> *Envoy? :* samedi 20 mars 2010 18:13
>> *? :* freeswitch-users at lists.freeswitch.org
>> *Objet :* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> ok irght.
>>
>> So for now I will continue to use my patch to make it work.
>>
>> I will give to you the status of any system upgrade that resolves the
>> problem
>>
>>  ----- Original Message -----
>>
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>>
>> *Sent:* Saturday, March 20, 2010 12:58 PM
>>
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> I am willing to change anything we have to as long as it works on all
>> platforms and we can give proper warnings to others.
>> Maybe we will do something like another param to say if your mailer app
>> needs a pipe or can accept the filename so we can do both
>> and leave the defaults to the current way.
>>
>> Maybe we can also have another option to make a simple relay code
>> internally that can deliver the file to a mail relay if you set that option.
>>
>>
>>  On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Ok I undertand.
>>
>> buy Why to continue to use pipe and STDIN since to use "filename" path
>> works ?
>>
>>  ----- Original Message -----
>>
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>>
>> *Sent:* Saturday, March 20, 2010 12:12 PM
>>
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> I think the problem is that fs sets the stack size for the core and new
>> threads to 240k so when the child is born from system() the new process
>> inherits the small stack size and sendmail needs more than 240k
>>
>> If you are root, it has the priveledge to raise the stack size to the
>> default size of 8m and sets it back down when the command completes.
>>
>> If you are not root you lack these privs and the stack size remains small.
>>
>> This has been a long time problem how to make a solution that is cross
>> platform and will not break working configurations.
>>
>> On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:
>>
>> I tried also with sudo and freeswitch user in sudoers with the same
>> result, segmentation fault.
>>
>> Anyway my patch works perfectly now, even with sendmail (the only
>> difference without patch is the "From" is taken from the voicemail.tpl and
>> not the user variable)
>>
>>
>> >
>> > ----- Original Message -----
>> > From: Fran?ois Legal
>> > To: freeswitch-users at lists.freeswitch.org
>>
>>
>> > Sent: Friday, March 19, 2010 6:51 AM
>> > Subject: Re: [Freeswitch-users] voicemail options [tempora...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> ...
>>
>>   ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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:+19193869900
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 12ukwn at gmail.com  Fri Apr  2 08:19:58 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 17:19:58 +0200
Subject: [Freeswitch-users] default_gateway
Message-ID: <20100402171958.2baecaaf@anubis.defcon1>

Hi list,

I'm making tests for a residential conf that use an ATA to bridge with
PSTN, can I define 'default_gateway' to a SIP extension (something like:
2222.xml), or am I oblige to use transfer (or something else)?

-- 
It's the RINSE CYCLE!!  They've ALL IGNORED the RINSE CYCLE!!


From brian at freeswitch.org  Fri Apr  2 08:26:10 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Apr 2010 10:26:10 -0500
Subject: [Freeswitch-users] default_gateway
In-Reply-To: <20100402171958.2baecaaf@anubis.defcon1>
References: <20100402171958.2baecaaf@anubis.defcon1>
Message-ID: <8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>

Thats just a gateway name based on a gateway you have setup on your system.

/b

On Apr 2, 2010, at 10:19 AM, Jean-Yves F. Barbier wrote:

> I'm making tests for a residential conf that use an ATA to bridge with
> PSTN, can I define 'default_gateway' to a SIP extension (something like:
> 2222.xml), or am I oblige to use transfer (or something else)?



From 12ukwn at gmail.com  Fri Apr  2 08:48:08 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 17:48:08 +0200
Subject: [Freeswitch-users] default_gateway
In-Reply-To: <8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>
References: <20100402171958.2baecaaf@anubis.defcon1>
	<8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>
Message-ID: <20100402174808.4305f53b@anubis.defcon1>

Le Fri, 2 Apr 2010 10:26:10 -0500,
Brian West <brian at freeswitch.org> a ?crit :


Could you point me to a doc that would help me to configure my ATA as a
GW? (it is a ht488, so it has FXO & FXS on the same IP)

> Thats just a gateway name based on a gateway you have setup on your
> system.

-- 
Obviously the only rational solution to your problem is suicide.


From 12ukwn at gmail.com  Fri Apr  2 11:03:39 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 20:03:39 +0200
Subject: [Freeswitch-users] difference
Message-ID: <20100402200339.26cde9a8@anubis.defcon1>

Hi,

What is exactly the difference(s) between extensions and endpoints?

-- 
Why do we have two eyes?  To watch 3-D movies with.


From clint at 42lines.net  Fri Apr  2 09:38:55 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 10:38:55 -0600
Subject: [Freeswitch-users] mod_conference lag
Message-ID: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>

Hi,

I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
has no lag, and the same is true for my coworker, but when we both dial a
mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
the machine is nil.  Any ideas?

Thanks,
-Clint

-- 
Clint Popetz
http://42lines.net
Scalable Web Application Development
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From vfclists at googlemail.com  Fri Apr  2 11:52:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 19:52:36 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
Message-ID: <s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>

On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
> Frank,
> sorry, I completely forgot one important detail:
> in the default conf, G729 is not allowed on any SIP profiles, so you have to
> modify vars.xml.
> You will find the following lines:
> ??<X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
> For now, I recommend you replace them by:
> ??<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
> Then in FS console, do:
> sofia profile external restart reloadxml
> sofia profile internal restart reloadxml
> It should then work far better.
> What we did there is to make G729 an accepted codec in inbound INVITEs and a
> proposed codec for outbound INVITEs.
> Look in external.xml or internal.xml, and look at the variables
> inbound-codec-prefs and outbound-codec-prefs.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>
> Frank,
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
> legs.
> I really recommend you enable G729 on the Linksys, enable SIP trace on FS
> console:
> sofia profile external siptrace on
> sofia profile internal siptrace on
> then make a (failing) call, capture the log on FS console and paste that to:
> http://pastebin:freeswitch at pastebin.freeswitch.org/

This the pastebin link

http://pastebin.freeswitch.org/12616

There is also one below it with the successful call.

I have set the default provider in vars.xml rather than under the
sip_profiles/external. Is there a section in the default dialplan that
handles the default context?

Can the options below be used in the default for the dialplan and sip_profiles?

In the dial plan
<action application="set" data="proxy_media=true"/>

In the sip profile
<param name="inbound-proxy-media" value="true"/>

 The Linksys accounts are in the default contexts.

> Then, send us back the link to your paste.
> You can also join us on #freeswitch (irc.freenode.net), for some live help.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
>
> The carrier insists on G729, although they can accept G711. I think
> their call volume does not make it easy on them and their customers as
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
> http://wiki.freeswitch.org/wiki/Bypass_Media.
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
> make the right adjustments?
>
> I have looked at
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
> and
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
> extensions to conf/directory/default and changed the provider in
> vars.xml and I want to be able to do the same in
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
> destination. Is there an option for not falling through to other
> extensions if they also match?
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From gavin.henry at gmail.com  Fri Apr  2 12:03:29 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 20:03:29 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.

I've worked out what version I was on before upgrading, so just
testing now. I remember,
as when compiling I raised this JIRA ticket:

http://jira.freeswitch.org/browse/MODLANG-157

so it was revision 16718


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From msc at freeswitch.org  Fri Apr  2 12:16:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:16:01 -0700
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
	<i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
	<l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>
Message-ID: <m2u87f2f3b91004021216tb8834e00h7260e3cd352443d5@mail.gmail.com>

I'm not sure why you are matching on "chan_name" but even so, you should be
able to do it with the groups:
<condition field="chan_name" expression="OpenZAP/1/(\d+)\@g1">
... stuff for group 1
</condition>
<condition field="chan_name" expression="OpenZAP/1/(\d+)\@g2">
... stuff for group 2
</condition>

-MC

On Thu, Apr 1, 2010 at 9:49 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Moises,
>       In all cases, I should use openzap/1/a/123 and there is no chance for
> configuring like openzap/1 and openzap/2. Am I right?
>
>       The difficulty I face here is, In dialplan I'm unable to route the
> calls based on spans, like,
>    <condition field="chan_name" expression="^OpenZAP/1.*/(.*)">
>        <action application="bridge" data="openzap/3/a/$1"/>
>    </condition>
>    <condition field="chan_name" expression="^OpenZAP/2.*/(.*)">
>        <action application="playback" data="under_maintenance.wav"/>
>    </condition>
>
>        Is there any other way to right the expression to match the spans?
>
> On Thu, Apr 1, 2010 at 6:34 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> There can be only 1 boost span currently. All b-channels must be added to
>> that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
>> belong to which group. You can see how to configure spans, channels and
>> groups here:
>>
>> http://wiki.sangoma.com/wanpipe-pri-advanced-options
>>
>> <http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing
>> from FreeSWITCH use the syntax requested:
>>
>> openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
>> openzap/1/1234 at g2 to dial using group 2 and so on.
>>
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>
>> On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> Dear friends,
>>>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
>>> question with regard to grouping the spans. There are 2 spans in the card
>>> and I would want to group the spans separately, so when I want to make call
>>> I could specify as either openzap/1/ or openzap/2/...
>>>     But, Now I'm unable to make calls like this. It is asking me to give
>>> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>>>
>>>     I don't want this way., I would want to group the spans differently.
>>> Kindly help me.
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  2 12:23:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:23:18 -0700
Subject: [Freeswitch-users] difference
In-Reply-To: <20100402200339.26cde9a8@anubis.defcon1>
References: <20100402200339.26cde9a8@anubis.defcon1>
Message-ID: <q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>

On Fri, Apr 2, 2010 at 11:03 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Hi,
>
> What is exactly the difference(s) between extensions and endpoints?
>
Ah, a philosophical question. :)

In FreeSWITCH an "extension" is something in the dialplan, whereas an
endpoint usually means a physical endpoint, like a phone. Sometimes we throw
the words around loosely, so don't let that confuse you.

Just curious - what lead up to this question?
-MC
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From msc at freeswitch.org  Fri Apr  2 12:25:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:25:55 -0700
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
Message-ID: <k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>

On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:

> Hi,
>
> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
> has no lag, and the same is true for my coworker, but when we both dial a
> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
> the machine is nil.  Any ideas?
>

Do you mean that when you speak, it takes 10-12 seconds before the audio is
heard by someone else in the conference? Also, can you try the same exercise
with a soft phone like x-lite? I'm curious to know if this happens only on
Skype calls or on any calls made to a conference.
-MC
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From david.ponzone at gmail.com  Fri Apr  2 12:36:14 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 21:36:14 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
Message-ID: <1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>

Frank,

re-read my second mail.
You have to enable G729 in FS prefs (vars.xml).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/04/2010 ? 20:52, Frank Church a ?crit :

> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>> Frank,
>> sorry, I completely forgot one important detail:
>> in the default conf, G729 is not allowed on any SIP profiles, so  
>> you have to
>> modify vars.xml.
>> You will find the following lines:
>>   <X-PRE-PROCESS cmd="set"
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>> For now, I recommend you replace them by:
>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>> Then in FS console, do:
>> sofia profile external restart reloadxml
>> sofia profile internal restart reloadxml
>> It should then work far better.
>> What we did there is to make G729 an accepted codec in inbound  
>> INVITEs and a
>> proposed codec for outbound INVITEs.
>> Look in external.xml or internal.xml, and look at the variables
>> inbound-codec-prefs and outbound-codec-prefs.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>>
>> Frank,
>> mod_g729 needs to be loaded, and then G729 needs to be negotiated  
>> on both
>> legs.
>> I really recommend you enable G729 on the Linksys, enable SIP trace  
>> on FS
>> console:
>> sofia profile external siptrace on
>> sofia profile internal siptrace on
>> then make a (failing) call, capture the log on FS console and paste  
>> that to:
>> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> This the pastebin link
>
> http://pastebin.freeswitch.org/12616
>
> There is also one below it with the successful call.
>
> I have set the default provider in vars.xml rather than under the
> sip_profiles/external. Is there a section in the default dialplan that
> handles the default context?
>
> Can the options below be used in the default for the dialplan and  
> sip_profiles?
>
> In the dial plan
> <action application="set" data="proxy_media=true"/>
>
> In the sip profile
> <param name="inbound-proxy-media" value="true"/>
>
> The Linksys accounts are in the default contexts.
>
>> Then, send us back the link to your paste.
>> You can also join us on #freeswitch (irc.freenode.net), for some  
>> live help.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>> their call volume does not make it easy on them and their customers  
>> as
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>> make the right adjustments?
>>
>> I have looked at
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>> and
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>> extensions to conf/directory/default and changed the provider in
>> vars.xml and I want to be able to do the same in
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>> destination. Is there an option for not falling through to other
>> extensions if they also match?
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 12ukwn at gmail.com  Fri Apr  2 12:43:07 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 21:43:07 +0200
Subject: [Freeswitch-users] difference
In-Reply-To: <q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
Message-ID: <20100402214307.744fb47a@anubis.defcon1>

Le Fri, 2 Apr 2010 12:23:18 -0700,
Michael Collins <msc at freeswitch.org> a ?crit :

Well, I'm greping the wiki's docs trying to understand how I could
configure FS to reach my ATA as a GW, or just have an output route to it
(but didn't succeed yet), and found 'endpoint'; so at first glance, I though
it was the term for what I was looking after.

If I understand right, an '9000' is an extension, but 1001 is an endpoint.

JY

> On Fri, Apr 2, 2010 at 11:03 AM, Jean-Yves F. Barbier
> <12ukwn at gmail.com>wrote:
> 
> > Hi,
> >
> > What is exactly the difference(s) between extensions and endpoints?
> >
> Ah, a philosophical question. :)
> 
> In FreeSWITCH an "extension" is something in the dialplan, whereas an
> endpoint usually means a physical endpoint, like a phone. Sometimes we
> throw the words around loosely, so don't let that confuse you.
> 
> Just curious - what lead up to this question?
> -MC




-- 
... Logically incoherent, semantically incomprehensible, and legally ...
impeccable!


From brian at freeswitch.org  Fri Apr  2 12:49:14 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Apr 2010 14:49:14 -0500
Subject: [Freeswitch-users] difference
In-Reply-To: <20100402214307.744fb47a@anubis.defcon1>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
	<20100402214307.744fb47a@anubis.defcon1>
Message-ID: <804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>


On Apr 2, 2010, at 2:43 PM, Jean-Yves F. Barbier wrote:

> Le Fri, 2 Apr 2010 12:23:18 -0700,
> Michael Collins <msc at freeswitch.org> a ?crit :
> 
> 
> If I understand right, an '9000' is an extension, but 1001 is an endpoint.
> 

1001 is an extension pointed at an registered sip user 1001 which happens to be same number.

/b


> JY

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From lcm at marshap.com  Fri Apr  2 12:51:43 2010
From: lcm at marshap.com (Larry Marshall)
Date: Fri, 2 Apr 2010 12:51:43 -0700
Subject: [Freeswitch-users] Error in 'svn up'
Message-ID: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars

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From 12ukwn at gmail.com  Fri Apr  2 13:01:02 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 22:01:02 +0200
Subject: [Freeswitch-users] difference
In-Reply-To: <804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
	<20100402214307.744fb47a@anubis.defcon1>
	<804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>
Message-ID: <20100402220102.729a8251@anubis.defcon1>

Le Fri, 2 Apr 2010 14:49:14 -0500,
Brian West <brian at freeswitch.org> a ?crit :

> 
> On Apr 2, 2010, at 2:43 PM, Jean-Yves F. Barbier wrote:
> 
> > Le Fri, 2 Apr 2010 12:23:18 -0700,
> > Michael Collins <msc at freeswitch.org> a ?crit :
> > 
> > 
> > If I understand right, an '9000' is an extension, but 1001 is an
> > endpoint.
> > 
> 
> 1001 is an extension pointed at an registered sip user 1001 which happens
> to be same number.

Raaaahhhhhh, I think I'll call all of'em extensions

-- 
Debian Hint #6: There is no hint #6. Submit a hint today !


From clint at 42lines.net  Fri Apr  2 13:04:55 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 14:04:55 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
Message-ID: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>

On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>
>> Hi,
>>
>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
>> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
>> has no lag, and the same is true for my coworker, but when we both dial a
>> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
>> the machine is nil.  Any ideas?
>>
>
> Do you mean that when you speak, it takes 10-12 seconds before the audio is
> heard by someone else in the conference?
>

Correct.


> Also, can you try the same exercise with a soft phone like x-lite? I'm
> curious to know if this happens only on Skype calls or on any calls made to
> a conference.
>

Sip is a whole other can of worms that's not working either.  When
connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
fine, but it won't listen to DTMF numbers to change the menu.  When running
xopier on linux, I can't hear anything and DTMF doesn't get through.  Both
those softphones work fine with asterisk, including dtmf.  I was punting on
getting sip working correctly in freeswitch until I determined whether
skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.

(It's entirely possible/probably I just have sip misconfigured...I'm just
using the default configuration in that regard.)

-Clint


-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
>
>


-- 
Clint Popetz
http://42lines.net
Scalable Web Application Development
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From anthony.minessale at gmail.com  Fri Apr  2 13:17:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Apr 2010 14:17:11 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
Message-ID: <m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>

Try calling the public conference at
sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
that is a properly setup box and can accept any registrations.

The sip and the conference all work fine so it's for sure a problem on your
end.




On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint at 42lines.net> wrote:

>
>
> On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>>
>>> Hi,
>>>
>>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
>>> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
>>> has no lag, and the same is true for my coworker, but when we both dial a
>>> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
>>> the machine is nil.  Any ideas?
>>>
>>
>> Do you mean that when you speak, it takes 10-12 seconds before the audio
>> is heard by someone else in the conference?
>>
>
> Correct.
>
>
>> Also, can you try the same exercise with a soft phone like x-lite? I'm
>> curious to know if this happens only on Skype calls or on any calls made to
>> a conference.
>>
>
> Sip is a whole other can of worms that's not working either.  When
> connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
> fine, but it won't listen to DTMF numbers to change the menu.  When running
> xopier on linux, I can't hear anything and DTMF doesn't get through.  Both
> those softphones work fine with asterisk, including dtmf.  I was punting on
> getting sip working correctly in freeswitch until I determined whether
> skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
>
> (It's entirely possible/probably I just have sip misconfigured...I'm just
> using the default configuration in that regard.)
>
> -Clint
>
>
>  -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
>>
>>
>
>
> --
> Clint Popetz
> http://42lines.net
> Scalable Web Application Development
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Fri Apr  2 13:27:54 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 2 Apr 2010 21:27:54 +0100
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com> 
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com> 
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
Message-ID: <m2hc32ee1cf1004021327h9bf17f06nc380bc56e736c0c0@mail.gmail.com>

> Sip is a whole other can of worms that's not working either.  When
> connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
> fine, but it won't listen to DTMF numbers to change the menu.
>

By coincidence, for the last couple of hours I've also been trying to work
out why I can't get DTMF on FreeSwitch on EC2 from Bria. Can't even think
how to debug it - the Bria diagnostic logs show the dtmf being prepared and
sent, but dtmf's don't seem to show on the server if  sofia loglevel all 9
is turned on.

So if anyone has any ideas on how to work it out, there's two of us that
could do with some help :-)

In my case I had set things up on a local dev server (with DTMF working, so
it's not a pure-Bria problem), then have transferred the same setup to the
EC2 server. I'm calling out from behind a NAT, so it could be something
related to that maybe. (The differences between the local and remote servers
are all related to being on internal vs external networks.

Cheers,
Fraser
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From gavin.henry at gmail.com  Fri Apr  2 13:42:49 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 21:42:49 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
Message-ID: <p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>

On 2 April 2010 20:03, Gavin Henry <gavin.henry at gmail.com> wrote:
> On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
>> I know its not much help..
>> but I think the first thing you need to do here is identify the version you
>> were on before ..
>> and see if you can easily narrow it down to a SVN revision that broke it.
>
> I've worked out what version I was on before upgrading, so just
> testing now. I remember,
> as when compiling I raised this JIRA ticket:
>
> http://jira.freeswitch.org/browse/MODLANG-157
>
> so it was revision 16718

OK, for some reason it works now with:

<param name="NDLB-force-rport" value="true"/>

when I'm positive it wasn't enabled before.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From clint at 42lines.net  Fri Apr  2 13:49:42 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 14:49:42 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
	<m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>
Message-ID: <j2xce06a7161004021349i6f6dc55qf55f6957b9896ea@mail.gmail.com>

On Fri, Apr 2, 2010 at 2:17 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> Try calling the public conference at sip:888 at conference.freeswitch.org
> that is a properly setup box and can accept any registrations.
>

Yup, that works, as does calling my existing asterisk pbx.

>
> The sip and the conference all work fine so it's for sure a problem on your end.

Well, this is with a fresh install using:

cd /usr/src?; wget http://www.freeswitch.org/eg/Makefile?; make
make all
cd freeswitch.trunk
make install
make cd-sounds-install
make cd-moh-install

the only thing I'v changed is to add <load module="mod_skypopen"/> and
	<action application="set"
data="skype_add_outband_dtmf_also_when_bridged=true"/>

and to install skype and make it listen to skypopen.   The rest is the
default config.

-Clint


>
>
>
> On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint at 42lines.net> wrote:
>>
>>
>> On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>>>>
>>>> Hi,
>>>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. ?CPU usage on the machine is nil. ?Any ideas?
>>>
>>> Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
>>
>> Correct.
>>
>>>
>>> Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
>>
>> Sip is a whole other can of worms that's not working either. ?When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. ?When running xopier on linux, I can't hear anything and DTMF doesn't get through. ?Both those softphones work fine with asterisk, including dtmf. ?I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
>> (It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)
>> -Clint
>>
>>> -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
>>>
>>
>>
>>
>> --
>> Clint Popetz
>> http://42lines.net
>> Scalable Web Application Development
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Clint Popetz
http://42lines.net
Scalable Web Application Development


From gavin.henry at gmail.com  Fri Apr  2 13:53:41 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 21:53:41 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
Message-ID: <y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>

> OK, for some reason it works now with:
>
> <param name="NDLB-force-rport" value="true"/>
>
> when I'm positive it wasn't enabled before.

Shouldn't FS NAT features detect that the phones are registering from
an internal IP and do the usual fs_nat=yes stuff in the contact header
without forcing rport above and send back responses to the port the
REGISTER came from?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From vfclists at googlemail.com  Fri Apr  2 13:53:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 21:53:36 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
	<1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
Message-ID: <q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>

On 2 April 2010 20:36, David Ponzone <david.ponzone at gmail.com> wrote:
> Frank,
> re-read my second mail.

http://pastebin.freeswitch.org/12617

> You have to enable G729 in FS prefs (vars.xml).
Was that obvious from the sip trace? I missed it the first time.

> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 20:52, Frank Church a ?crit :
>
> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>
> Frank,
>
> sorry, I completely forgot one important detail:
>
> in the default conf, G729 is not allowed on any SIP profiles, so you have to
>
> modify vars.xml.
>
> You will find the following lines:
>
> ??<X-PRE-PROCESS cmd="set"
>
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>
> For now, I recommend you replace them by:
>
> ??<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>
> Then in FS console, do:
>
> sofia profile external restart reloadxml
>
> sofia profile internal restart reloadxml
>
> It should then work far better.
>
> What we did there is to make G729 an accepted codec in inbound INVITEs and a
>
> proposed codec for outbound INVITEs.
>
> Look in external.xml or internal.xml, and look at the variables
>
> inbound-codec-prefs and outbound-codec-prefs.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>
> Frank,
>
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
>
> legs.
>
> I really recommend you enable G729 on the Linksys, enable SIP trace on FS
>
> console:
>
> sofia profile external siptrace on
>
> sofia profile internal siptrace on
>
> then make a (failing) call, capture the log on FS console and paste that to:
>
> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> This the pastebin link
>
> http://pastebin.freeswitch.org/12616
>
> There is also one below it with the successful call.
>
> I have set the default provider in vars.xml rather than under the
> sip_profiles/external. Is there a section in the default dialplan that
> handles the default context?
>
> Can the options below be used in the default for the dialplan and
> sip_profiles?
>
> In the dial plan
> <action application="set" data="proxy_media=true"/>
>
> In the sip profile
> <param name="inbound-proxy-media" value="true"/>
>
> The Linksys accounts are in the default contexts.
>
> Then, send us back the link to your paste.
>
> You can also join us on #freeswitch (irc.freenode.net), for some live help.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
>
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
>
> The carrier insists on G729, although they can accept G711. I think
>
> their call volume does not make it easy on them and their customers as
>
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
>
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>
> http://wiki.freeswitch.org/wiki/Bypass_Media.
>
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
>
> make the right adjustments?
>
> I have looked at
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>
> and
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
>
> extensions to conf/directory/default and changed the provider in
>
> vars.xml and I want to be able to do the same in
>
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
>
> destination. Is there an option for not falling through to other
>
> extensions if they also match?
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> --
>
> Frank Church
>
> =======================
>
> http://devblog.brahmancreations.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
>
>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From lon at kickasspixels.com  Fri Apr  2 14:24:51 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 14:24:51 -0700
Subject: [Freeswitch-users] CLI status?
Message-ID: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>

Hi gang!

I'm trying to determine the best way to periodically check the status
of production Freeswitch deployments.

While Freeswitch has been incredibly stable in production, I would
like to ping it periodically to check that its not hung in someway.

Has anyone considered or done this yet?

My thought it so issue a command via fs_cli periodically and compare
the out. But am trying to determine what a good command would be.

thinking of using show channels or show calls, in addition to status.

Any thoughts? Is this a wrong-headed approach?

Thanks for any feedback.

Lon


From vetali100 at gmail.com  Fri Apr  2 14:37:49 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 3 Apr 2010 00:37:49 +0300
Subject: [Freeswitch-users]  Calling in context public instead of default,
	mod_xml_curl
Message-ID: <k2ma858e7691004021437s5d64917cqb80c456b2bb9e119@mail.gmail.com>

Hi,

I am using mod_xml_curl to generate user directory data.
It produces data like this:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${domain}">
      <params>
        <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
          <users>
            <user id="20001">
              <params>
                <param name="a1-hash"
value="d8c32787fd9f00b5ad4246e93f757360"/>
                <param name="vm-a1-hash"
value="a53b5e8b5edf34208274cbaa00412097"/>
              </params>
              <variables>
                *<variable name="user_context" value="**default**"/>*
                <variable name="accountcode" value="20001"/>
              </variables>
            </user>
          </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Both internal and external sofia profiles have context = "public".
But it should be overridden here in the user settings to  DEFAULT.


Wwhen I call to any number using this subscriber 20001 it calls in *context
PUBLIC instead of DEFAULT.*
2010-04-02 17:27:32.384472 [INFO] mod_dialplan_xml.c:418 Processing
20001->001...[removed]... in context *public*

I need it to call in context default.

When I change sofia profile settings to context="default", it works fine,
but shouldn't it work like this as well?

Please any hints where else should I change the context?

Thank you,
vIT
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From sos at sokhapkin.dyndns.org  Fri Apr  2 14:40:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 2 Apr 2010 17:40:22 -0400
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <201004021740.22716.sos@sokhapkin.dyndns.org>

To me monit works the best. http://mmonit.com/monit/

On Friday 02 April 2010, Lon Baker wrote:
> Hi gang!
> 
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
> 
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
> 
> Has anyone considered or done this yet?
> 
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
> 
> thinking of using show channels or show calls, in addition to status.
> 
> Any thoughts? Is this a wrong-headed approach?
> 
> Thanks for any feedback.
> 
> Lon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From troy at tlainvestments.com  Fri Apr  2 14:46:21 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 2 Apr 2010 14:46:21 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <B2428CEC-A0E4-4224-9968-74D86DC444C6@tlainvestments.com>

One way I've contemplated is to use munin/munin node on the FS box and nagios on a central server.  That would provide notification if the service were to go offline.  I have the components in place, but haven't gotten around to writing a FS nagios plugin (which could use fs_cli -x status, for example).

-Troy


On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:

> Hi gang!
> 
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
> 
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
> 
> Has anyone considered or done this yet?
> 
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
> 
> thinking of using show channels or show calls, in addition to status.
> 
> Any thoughts? Is this a wrong-headed approach?
> 
> Thanks for any feedback.
> 
> Lon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From spencer at 5ninesolutions.com  Fri Apr  2 14:54:48 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 2 Apr 2010 14:54:48 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>

We run monit on each local machine sending out an options ping every  
minute.  If it doesn't get a response then monit restarts the  
freeswitch process.  The nice thing about this is that you can  
actually verify that Freeswitch is responding to SIP requests.


On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:

> Hi gang!
>
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
>
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
>
> Has anyone considered or done this yet?
>
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
>
> thinking of using show channels or show calls, in addition to status.
>
> Any thoughts? Is this a wrong-headed approach?
>
> Thanks for any feedback.
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From troy at tlainvestments.com  Fri Apr  2 14:58:02 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 2 Apr 2010 14:58:02 -0700
Subject: [Freeswitch-users] make fails
Message-ID: <5EB95B30-AF81-43D6-8583-6FEE7A967C92@tlainvestments.com>

Help.  I've compiled FS successfully on this box many times, but now, it's failing.  It must be something I've done as it even fails when I check out an svn revision that compiled in the past on this box.  I'm hopeful someone here can point me in the right direction.  Here's the output:

make
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
Making all in .
gcc -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/home/chronos_client/freeswitch.svn/libs/apr/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -I/home/chronos_client/freeswitch.svn/libs/stfu -I/home/chronos_client/freeswitch.svn/libs/sqlite -I/home/chronos_client/freeswitch.svn/libs/pcre -I/home/chronos_client/freeswitch.svn/libs/speex/include -Ilibs/speex/include -I/home/chronos_client/freeswitch.svn/libs/srtp/include -I/home/chronos_client/freeswitch.svn/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -DSWITCH_HAVE_ODBC -I/usr/include -I/home/chronos_client/freeswitch.svn/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o  -pthread -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a libs/libedit/src/.libs/libedit.a -L/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -lrt -ldl -lcrypt -lpthread -lssl -lcrypto -lncurses -Wl,--rpath -Wl,/usr/local/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `XML_ErrorString'
./.libs/libfreeswitch.so: undefined reference to `XML_SetUserData'
./.libs/libfreeswitch.so: undefined reference to `XML_ParserFree'
./.libs/libfreeswitch.so: undefined reference to `XML_SetElementHandler'
./.libs/libfreeswitch.so: undefined reference to `XML_SetCharacterDataHandler'
./.libs/libfreeswitch.so: undefined reference to `XML_GetErrorCode'
./.libs/libfreeswitch.so: undefined reference to `XML_ParserCreate'
./.libs/libfreeswitch.so: undefined reference to `XML_Parse'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1

Thanks!
Troy



From lon at kickasspixels.com  Fri Apr  2 17:23:22 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 17:23:22 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com> 
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
Message-ID: <v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>

Would you care to share an example of the config for monit? I will
document the practice on the wiki.

On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> We run monit on each local machine sending out an options ping every
> minute. ?If it doesn't get a response then monit restarts the
> freeswitch process. ?The nice thing about this is that you can
> actually verify that Freeswitch is responding to SIP requests.
>
>
> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>
>> Hi gang!
>>
>> I'm trying to determine the best way to periodically check the status
>> of production Freeswitch deployments.
>>
>> While Freeswitch has been incredibly stable in production, I would
>> like to ping it periodically to check that its not hung in someway.
>>
>> Has anyone considered or done this yet?
>>
>> My thought it so issue a command via fs_cli periodically and compare
>> the out. But am trying to determine what a good command would be.
>>
>> thinking of using show channels or show calls, in addition to status.
>>
>> Any thoughts? Is this a wrong-headed approach?
>>
>> Thanks for any feedback.
>>
>> Lon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sos at sokhapkin.dyndns.org  Fri Apr  2 17:55:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 2 Apr 2010 20:55:11 -0400
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
Message-ID: <201004022055.12057.sos@sokhapkin.dyndns.org>

check process freeswitch
  with pidfile /opt/freeswitch/run/freeswitch.pid
  start program = "/etc/init.d/freeswitch restart"
  stop program =  "/etc/init.d/freeswitch stop"
  if totalmem > 1000.0 MB for 5 cycles then alert
  if totalmem > 1500.0 MB for 5 cycles then alert
  if totalmem > 2000.0 MB for 5 cycles then restart
  if cpu > 60% for 5 cycles then alert
  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2 
cycles
  then restart
  if 5 restarts within 5 cycles then timeout


On Friday 02 April 2010, Lon Baker wrote:
> Would you care to share an example of the config for monit? I will
> document the practice on the wiki.
> 
> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
> 
> <spencer at 5ninesolutions.com> wrote:
> > We run monit on each local machine sending out an options ping every
> > minute. ?If it doesn't get a response then monit restarts the
> > freeswitch process. ?The nice thing about this is that you can
> > actually verify that Freeswitch is responding to SIP requests.
> >
> > On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
> >> Hi gang!
> >>
> >> I'm trying to determine the best way to periodically check the status
> >> of production Freeswitch deployments.
> >>
> >> While Freeswitch has been incredibly stable in production, I would
> >> like to ping it periodically to check that its not hung in someway.
> >>
> >> Has anyone considered or done this yet?
> >>
> >> My thought it so issue a command via fs_cli periodically and compare
> >> the out. But am trying to determine what a good command would be.
> >>
> >> thinking of using show channels or show calls, in addition to status.
> >>
> >> Any thoughts? Is this a wrong-headed approach?
> >>
> >> Thanks for any feedback.
> >>
> >> Lon
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr  2 21:18:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Apr 2010 22:18:33 -0600
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
Message-ID: <w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>

Nothing is wrong with it,  there is no revision that broke it or there would
be 100,000 people saying it was broken.
Change you perspective to looking for the problem on your end and you will
have more luck finding your problem.

Backup your config by moving it out of the way, re-install and try the
defaults.
We use FS with nat like this 12 hours a day.


On Fri, Apr 2, 2010 at 2:53 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> > OK, for some reason it works now with:
> >
> > <param name="NDLB-force-rport" value="true"/>
> >
> > when I'm positive it wasn't enabled before.
>
> Shouldn't FS NAT features detect that the phones are registering from
> an internal IP and do the usual fs_nat=yes stuff in the contact header
> without forcing rport above and send back responses to the port the
> REGISTER came from?
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lon at kickasspixels.com  Fri Apr  2 23:16:59 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 23:16:59 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <201004022055.12057.sos@sokhapkin.dyndns.org>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
Message-ID: <48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>

Thank you!

On Apr 2, 2010, at 5:55 PM, Sergey Okhapkin wrote:

> check process freeswitch
>  with pidfile /opt/freeswitch/run/freeswitch.pid
>  start program = "/etc/init.d/freeswitch restart"
>  stop program =  "/etc/init.d/freeswitch stop"
>  if totalmem > 1000.0 MB for 5 cycles then alert
>  if totalmem > 1500.0 MB for 5 cycles then alert
>  if totalmem > 2000.0 MB for 5 cycles then restart
>  if cpu > 60% for 5 cycles then alert
>  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2 
> cycles
>  then restart
>  if 5 restarts within 5 cycles then timeout
> 
> 
> On Friday 02 April 2010, Lon Baker wrote:
>> Would you care to share an example of the config for monit? I will
>> document the practice on the wiki.
>> 
>> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
>> 
>> <spencer at 5ninesolutions.com> wrote:
>>> We run monit on each local machine sending out an options ping every
>>> minute. ?If it doesn't get a response then monit restarts the
>>> freeswitch process. ?The nice thing about this is that you can
>>> actually verify that Freeswitch is responding to SIP requests.
>>> 
>>> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>>>> Hi gang!
>>>> 
>>>> I'm trying to determine the best way to periodically check the status
>>>> of production Freeswitch deployments.
>>>> 
>>>> While Freeswitch has been incredibly stable in production, I would
>>>> like to ping it periodically to check that its not hung in someway.
>>>> 
>>>> Has anyone considered or done this yet?
>>>> 
>>>> My thought it so issue a command via fs_cli periodically and compare
>>>> the out. But am trying to determine what a good command would be.
>>>> 
>>>> thinking of using show channels or show calls, in addition to status.
>>>> 
>>>> Any thoughts? Is this a wrong-headed approach?
>>>> 
>>>> Thanks for any feedback.
>>>> 
>>>> Lon
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 



From david.ponzone at gmail.com  Fri Apr  2 23:41:18 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 3 Apr 2010 08:41:18 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
	<1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
	<q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>
Message-ID: <B4FE4E64-39EF-43F0-9C1B-6157807AFEB7@gmail.com>

Yes, in the lines:

2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001 at 192.168.1.133 
  [CS_NEW] [INCOMPATIBLE_DESTINATION]
send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
     
------------------------------------------------------------------------
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
    To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
    Call-ID: 1c27844d-e299e5e9 at 192.168.4.154

That is the answer from FS to the phone, just after receiving the  
INVITE that contains only G729 and NSE (??) in the SDP.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/04/2010 ? 22:53, Frank Church a ?crit :

> On 2 April 2010 20:36, David Ponzone <david.ponzone at gmail.com> wrote:
>> Frank,
>> re-read my second mail.
>
> http://pastebin.freeswitch.org/12617
>
>> You have to enable G729 in FS prefs (vars.xml).
> Was that obvious from the sip trace? I missed it the first time.
>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 20:52, Frank Church a ?crit :
>>
>> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>>
>> Frank,
>>
>> sorry, I completely forgot one important detail:
>>
>> in the default conf, G729 is not allowed on any SIP profiles, so  
>> you have to
>>
>> modify vars.xml.
>>
>> You will find the following lines:
>>
>>   <X-PRE-PROCESS cmd="set"
>>
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>>
>> For now, I recommend you replace them by:
>>
>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>>
>> Then in FS console, do:
>>
>> sofia profile external restart reloadxml
>>
>> sofia profile internal restart reloadxml
>>
>> It should then work far better.
>>
>> What we did there is to make G729 an accepted codec in inbound  
>> INVITEs and a
>>
>> proposed codec for outbound INVITEs.
>>
>> Look in external.xml or internal.xml, and look at the variables
>>
>> inbound-codec-prefs and outbound-codec-prefs.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>>
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>>
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>>
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>>
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>>
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>>
>> Frank,
>>
>> mod_g729 needs to be loaded, and then G729 needs to be negotiated  
>> on both
>>
>> legs.
>>
>> I really recommend you enable G729 on the Linksys, enable SIP trace  
>> on FS
>>
>> console:
>>
>> sofia profile external siptrace on
>>
>> sofia profile internal siptrace on
>>
>> then make a (failing) call, capture the log on FS console and paste  
>> that to:
>>
>> http://pastebin:freeswitch at pastebin.freeswitch.org/
>>
>> This the pastebin link
>>
>> http://pastebin.freeswitch.org/12616
>>
>> There is also one below it with the successful call.
>>
>> I have set the default provider in vars.xml rather than under the
>> sip_profiles/external. Is there a section in the default dialplan  
>> that
>> handles the default context?
>>
>> Can the options below be used in the default for the dialplan and
>> sip_profiles?
>>
>> In the dial plan
>> <action application="set" data="proxy_media=true"/>
>>
>> In the sip profile
>> <param name="inbound-proxy-media" value="true"/>
>>
>> The Linksys accounts are in the default contexts.
>>
>> Then, send us back the link to your paste.
>>
>> You can also join us on #freeswitch (irc.freenode.net), for some  
>> live help.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>>
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>>
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>>
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>>
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>>
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>>
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>>
>> their call volume does not make it easy on them and their customers  
>> as
>>
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>>
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>>
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>>
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>>
>> make the right adjustments?
>>
>> I have looked at
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>>
>> and
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>>
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>>
>> extensions to conf/directory/default and changed the provider in
>>
>> vars.xml and I want to be able to do the same in
>>
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>>
>> destination. Is there an option for not falling through to other
>>
>> extensions if they also match?
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>> Frank Church
>>
>> =======================
>>
>> http://devblog.brahmancreations.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
>>
>>
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 david.ponzone at gmail.com  Fri Apr  2 23:45:32 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 3 Apr 2010 08:45:32 +0200
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
	<48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>
Message-ID: <31C0F8F8-AE63-4D13-9BA7-49DCE237A170@gmail.com>

I would recommend to, obviously, ping all profiles of FS.
I recently had an issue where one profile only stopped working.
It happened only once, in a previous svn, but who knows...

Another way to monitor is to send a test call with sipp through a  
registered user to a gateway.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 03/04/2010 ? 08:16, Lon Baker a ?crit :

> Thank you!
>
> On Apr 2, 2010, at 5:55 PM, Sergey Okhapkin wrote:
>
>> check process freeswitch
>> with pidfile /opt/freeswitch/run/freeswitch.pid
>> start program = "/etc/init.d/freeswitch restart"
>> stop program =  "/etc/init.d/freeswitch stop"
>> if totalmem > 1000.0 MB for 5 cycles then alert
>> if totalmem > 1500.0 MB for 5 cycles then alert
>> if totalmem > 2000.0 MB for 5 cycles then restart
>> if cpu > 60% for 5 cycles then alert
>> if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip  
>> within 2
>> cycles
>> then restart
>> if 5 restarts within 5 cycles then timeout
>>
>>
>> On Friday 02 April 2010, Lon Baker wrote:
>>> Would you care to share an example of the config for monit? I will
>>> document the practice on the wiki.
>>>
>>> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
>>>
>>> <spencer at 5ninesolutions.com> wrote:
>>>> We run monit on each local machine sending out an options ping  
>>>> every
>>>> minute. ?If it doesn't get a response then monit restarts the
>>>> freeswitch process. ?The nice thing about this is that you can
>>>> actually verify that Freeswitch is responding to SIP requests.
>>>>
>>>> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>>>>> Hi gang!
>>>>>
>>>>> I'm trying to determine the best way to periodically check the  
>>>>> status
>>>>> of production Freeswitch deployments.
>>>>>
>>>>> While Freeswitch has been incredibly stable in production, I would
>>>>> like to ping it periodically to check that its not hung in  
>>>>> someway.
>>>>>
>>>>> Has anyone considered or done this yet?
>>>>>
>>>>> My thought it so issue a command via fs_cli periodically and  
>>>>> compare
>>>>> the out. But am trying to determine what a good command would be.
>>>>>
>>>>> thinking of using show channels or show calls, in addition to  
>>>>> status.
>>>>>
>>>>> Any thoughts? Is this a wrong-headed approach?
>>>>>
>>>>> Thanks for any feedback.
>>>>>
>>>>> Lon
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From abid_freeswitch at live.com  Sat Apr  3 00:02:55 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 3 Apr 2010 12:02:55 +0500
Subject: [Freeswitch-users] Radius/Authentication/Authorization
Message-ID: <COL109-W11819E0B21AF7600E30CD4F41B0@phx.gbl>



Dear Tihomir,
Any update and help on the below please.
Abid
From: abid_freeswitch at live.com
To: tculjaga at gmail.com
CC: neal at wanlink.com
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization
Date: Wed, 31 Mar 2010 19:35:32 +0500








Dear Tihomir,
Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps.
Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows.
2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str**
Please advise. Thanks.
Regards-----------Abid Saleem
--Forwarded Message Attachment--
From: tculjaga at gmail.com
CC: neal at wanlink.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 30 Mar 2010 21:42:03 +0200
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization

hello,

here is an example



<configuration name="rad_auth.conf" description="radius authentification module">
  <settings>
     <!-- backward compatibility to allow radiusclient config file instead of an embedded config -->

     <!--  <param name="radius_config" value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->
  </settings>

  <client>
    <param name="authserver" value="10.1.1.10:1812:gateway"/>

    <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>
    <param name="seqfile" value="/var/run/radius.seq"/>
    <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>

    <param name="default_realm" value=""/>
    <param name="radius_timeout" value="3"/>
    <param name="radius_retries" value="2"/>
    <param name="radius_deadtime" value="0"/>

    <param name="bindaddr" value="*"/>
  </client>

  <vsas>
    <!-- 
         name:       just a description
         value:      direct input or variable 
         pec:        vendor ID (0 for default, 9 for cisco...)

         expr:       1 for channel variable, 2 for direct input
         direction:  in for radius-request, out for radius-response
    -->

    <!-- mappings for radius request message; input attributes -->

    <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/>
    <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/>

    <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/>
    <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/>

    <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/>
    <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/>

    <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>

    <!-- mappings for radius-response message; output values from returning outributes -->

    <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/>
    <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/>

    <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/>
    <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/>

    <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/>
    <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/>

    <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>
    <!-- expr param is to be ignored here-->

  </vsas>
</configuration>




in the dialplan you need to trigger auth as:

      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



there are two behaviours:

1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number
2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance







  <extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

<!--      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>  -->

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


  <extension name="unitest_rad-ANI-balance">
    <condition field="destination_number" expression="^602$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


will update the wiki by the end of the week.


you have enough information for now.

Tihomir.



On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch at live.com> wrote:






Hi Neal and other Contributors to FS,
I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration.

If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation.

Regards----------Abid SaleemSr. Product Manager
 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now. 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From jayesh.voip at gmail.com  Sat Apr  3 00:43:38 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Sat, 3 Apr 2010 13:13:38 +0530
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
Message-ID: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>

I expected at east one reply saying that the question is stupid, and the
solution is simple !!
Any folks who can help me understand only how to achieve this in FS which is
acheivable in asterisk as follows:
domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
the call goes to context "mydomain" in dialplan)
domain = yourdomain.com, yourdomain (if any call has domain as "
yourdomain.com", the call goes to context "yourdomain" in dialplan)

These calls can come from anywhere, in my case it comes from an Opensips
instance !!

Thanks for any replies :)

--- Jayesh

On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi All,
> I am quite very new to freeswitch and I am kind of playing with it to
> understand it better.
> I am primarily using Opensips as registrar and SIP Proxy and intend to use
> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
> Opensips is a multi-domain setup and I wish to have all the configuration of
> media-capabilties segregated domain-wise in the FS too.
>
> For eg: When a call for user at domain1.com needs to go to voicemail, I
> redirect that call to FS IP address keeping the URI intact. I add the
> mailbox number as a header as X-Mailbox and have FS extract it and go to
> appropriate mailbox. Similarly when a call for user at domain2.com needs to
> go to voicemail I do the same thing.
> The requirement is I want to maintain the dialplans for each domains
> separately. Thus if call from Opensips comes to FS with domain as domain1,
> the call should go to dialplan context domain1 and similarly if call from
> Opensips comes to FS with domain2 the call handling should be mentioned in
> the domain2 context.
>
> The problem is; I am not able to send the calls to respective contexts
> according to their domains when they come from Opensips. I've read the
> examples on multi-domain setup and have tried taking some help from that
> example, but whenever the call comes from Opensips to FS, it tries to go
> into the context that is defined in the SIP Profile. If i don't mention
> anything in the SIP Profile, it tries to search for default context.
> I have tried the following:
> 1) Created file domain1.xml and domain2.xml in the directory folder.
> 2) mentioned parameters in domain1.xml as follows:
> <include>
> <domain name = "domain1.com"
> <params>
> <param name="user_context" value "domain1.com"/>
> </params>
> </domain>
> </include>
> 3) Similarly done for file domain2.xml.
>
> But I am just not able to get the calls to the required context according
> to the domain value in the r-uri. In asterisk something like this in
> sip.conf worked fine for me:
> domain=domain1.com, domain1.com
> domain=domain2.com, domain2.com
> Can someone please help me understanding where I am going wrong or have I
> mis-understood something?
>
> Thanks in advance !!
>
> --- Jayesh
>
>
>
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From hungngm at bkav.com.vn  Sat Apr  3 01:05:36 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Sat, 03 Apr 2010 15:05:36 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>



 

Hi Seven Du.

Thanks to yours suggetion.

I have an ideal, it is: when the call between caller and agent is set, the
caller_id is determined. So, i want to edit code to sent the agent information
(the call_id and call_id_number) which will be displayed againt in the agent's
softphone (as Xlite..) when the call is happening.

I read some documents but i still can't determine: It's maybe yes or maybe to do
this and where to do this.

Can you give me some comments.

Best Regard.

 

Seven Du [dujinfang at gmail.com]

 

??As discussed in the list, it's not a freeswitch problem but a reality of
life.

Think about customer A and B calls in one after another, then if
FreeSWITCH call agent X with caller id A and Y with caller id B, and
angent Y answers before X, then

1) if bridge Y with A with the FIFO rule, then the caller id is wrong
2) if bridge Y with B, the caller id is right but it breaks the rule
of FIFO - A should be served before B!! And what even worse is that
if X never answer A then A never can be served which is really
unfair!!

Of course you don't want 1), and you don't need mod_fifo if you want
behavior 2), you just need some dialplan trick or some simple Lua
script I think. Also FreeSWITCH is designed to be easily extended with
almost any languages so feel free to implement anything.

2010/3/31 Nguy??n M???nh H??ng 

:
> Hi Mike and Seven Du.
> Thanks to yours help.
> I known the mechanism of mod_fifo.
>>>http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.ht
ml.
> What a pity, It can't solve this problem. I can't use freeswitch for my call
> center.
> Hope new version can solve 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

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From red.rain.seven at gmail.com  Sat Apr  3 02:30:52 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 3 Apr 2010 02:30:52 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <201004022055.12057.sos@sokhapkin.dyndns.org>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
Message-ID: <m2m59ad9ca11004030230h6af1ccfet3b22ca4b914dc953@mail.gmail.com>

Sergey:

Thank you for sharing , I am just recently looking to use monit to monitor
our freeswitch and other production servers.

Henry

Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
Contact Me [image:
Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image:
Facebook] <http://www.facebook.com/profile.php?id=595148342>[image:
Twitter]<http://twitter.com/unicsolution>


On Fri, Apr 2, 2010 at 5:55 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> check process freeswitch
>  with pidfile /opt/freeswitch/run/freeswitch.pid
>  start program = "/etc/init.d/freeswitch restart"
>  stop program =  "/etc/init.d/freeswitch stop"
>  if totalmem > 1000.0 MB for 5 cycles then alert
>  if totalmem > 1500.0 MB for 5 cycles then alert
>  if totalmem > 2000.0 MB for 5 cycles then restart
>  if cpu > 60% for 5 cycles then alert
>  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2
> cycles
>  then restart
>  if 5 restarts within 5 cycles then timeout
>
>
> On Friday 02 April 2010, Lon Baker wrote:
> > Would you care to share an example of the config for monit? I will
> > document the practice on the wiki.
> >
> > On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
> >
> > <spencer at 5ninesolutions.com> wrote:
> > > We run monit on each local machine sending out an options ping every
> > > minute. ?If it doesn't get a response then monit restarts the
> > > freeswitch process. ?The nice thing about this is that you can
> > > actually verify that Freeswitch is responding to SIP requests.
> > >
> > > On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
> > >> Hi gang!
> > >>
> > >> I'm trying to determine the best way to periodically check the status
> > >> of production Freeswitch deployments.
> > >>
> > >> While Freeswitch has been incredibly stable in production, I would
> > >> like to ping it periodically to check that its not hung in someway.
> > >>
> > >> Has anyone considered or done this yet?
> > >>
> > >> My thought it so issue a command via fs_cli periodically and compare
> > >> the out. But am trying to determine what a good command would be.
> > >>
> > >> thinking of using show channels or show calls, in addition to status.
> > >>
> > >> Any thoughts? Is this a wrong-headed approach?
> > >>
> > >> Thanks for any feedback.
> > >>
> > >> Lon
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From lloydie.t at googlemail.com  Sat Apr  3 04:10:30 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Sat, 3 Apr 2010 12:10:30 +0100
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
Message-ID: <n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>

On your DNS server does mydomain.com and yourdomain point to your FS server
ip address. I suspect you may not get much response with showing the errors
for the fs cli at least. Someone more qualified maybe able to help with this
info

Lloydie T

On 3 April 2010 08:43, Jayesh Nambiar <jayesh.voip at gmail.com> wrote:

> I expected at east one reply saying that the question is stupid, and the
> solution is simple !!
> Any folks who can help me understand only how to achieve this in FS which
> is acheivable in asterisk as follows:
> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
> the call goes to context "mydomain" in dialplan)
> domain = yourdomain.com, yourdomain (if any call has domain as "
> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>
> These calls can come from anywhere, in my case it comes from an Opensips
> instance !!
>
> Thanks for any replies :)
>
> --- Jayesh
>
>
> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi All,
>> I am quite very new to freeswitch and I am kind of playing with it to
>> understand it better.
>> I am primarily using Opensips as registrar and SIP Proxy and intend to use
>> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
>> Opensips is a multi-domain setup and I wish to have all the configuration of
>> media-capabilties segregated domain-wise in the FS too.
>>
>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>> redirect that call to FS IP address keeping the URI intact. I add the
>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>> go to voicemail I do the same thing.
>> The requirement is I want to maintain the dialplans for each domains
>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>> the call should go to dialplan context domain1 and similarly if call from
>> Opensips comes to FS with domain2 the call handling should be mentioned in
>> the domain2 context.
>>
>> The problem is; I am not able to send the calls to respective contexts
>> according to their domains when they come from Opensips. I've read the
>> examples on multi-domain setup and have tried taking some help from that
>> example, but whenever the call comes from Opensips to FS, it tries to go
>> into the context that is defined in the SIP Profile. If i don't mention
>> anything in the SIP Profile, it tries to search for default context.
>> I have tried the following:
>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>> 2) mentioned parameters in domain1.xml as follows:
>> <include>
>> <domain name = "domain1.com"
>> <params>
>> <param name="user_context" value "domain1.com"/>
>> </params>
>> </domain>
>> </include>
>> 3) Similarly done for file domain2.xml.
>>
>> But I am just not able to get the calls to the required context according
>> to the domain value in the r-uri. In asterisk something like this in
>> sip.conf worked fine for me:
>> domain=domain1.com, domain1.com
>> domain=domain2.com, domain2.com
>>  Can someone please help me understanding where I am going wrong or have I
>> mis-understood something?
>>
>> Thanks in advance !!
>>
>> --- Jayesh
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloydie.t at googlemail.com  Sat Apr  3 04:16:44 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Sat, 3 Apr 2010 12:16:44 +0100
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
	<n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>
Message-ID: <t2m8b61bd671004030416k17298698h48daf41f5e957fa1@mail.gmail.com>

with showing = without showing

On 3 April 2010 12:10, lloyd thomas <lloydie.t at googlemail.com> wrote:

> On your DNS server does mydomain.com and yourdomain point to your FS
> server ip address. I suspect you may not get much response with showing the
> errors for the fs cli at least. Someone more qualified maybe able to help
> with this info
>
> Lloydie T
>
> On 3 April 2010 08:43, Jayesh Nambiar <jayesh.voip at gmail.com> wrote:
>
>> I expected at east one reply saying that the question is stupid, and the
>> solution is simple !!
>> Any folks who can help me understand only how to achieve this in FS which
>> is acheivable in asterisk as follows:
>> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
>> the call goes to context "mydomain" in dialplan)
>> domain = yourdomain.com, yourdomain (if any call has domain as "
>> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>>
>> These calls can come from anywhere, in my case it comes from an Opensips
>> instance !!
>>
>> Thanks for any replies :)
>>
>> --- Jayesh
>>
>>
>> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>>
>>> Hi All,
>>> I am quite very new to freeswitch and I am kind of playing with it to
>>> understand it better.
>>> I am primarily using Opensips as registrar and SIP Proxy and intend to
>>> use FS as media server handling voicemails, IVR, Announcements, MeetMe etc.
>>> My Opensips is a multi-domain setup and I wish to have all the configuration
>>> of media-capabilties segregated domain-wise in the FS too.
>>>
>>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>>> redirect that call to FS IP address keeping the URI intact. I add the
>>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>>> go to voicemail I do the same thing.
>>> The requirement is I want to maintain the dialplans for each domains
>>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>>> the call should go to dialplan context domain1 and similarly if call from
>>> Opensips comes to FS with domain2 the call handling should be mentioned in
>>> the domain2 context.
>>>
>>> The problem is; I am not able to send the calls to respective contexts
>>> according to their domains when they come from Opensips. I've read the
>>> examples on multi-domain setup and have tried taking some help from that
>>> example, but whenever the call comes from Opensips to FS, it tries to go
>>> into the context that is defined in the SIP Profile. If i don't mention
>>> anything in the SIP Profile, it tries to search for default context.
>>> I have tried the following:
>>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>>> 2) mentioned parameters in domain1.xml as follows:
>>> <include>
>>> <domain name = "domain1.com"
>>> <params>
>>> <param name="user_context" value "domain1.com"/>
>>> </params>
>>> </domain>
>>> </include>
>>> 3) Similarly done for file domain2.xml.
>>>
>>> But I am just not able to get the calls to the required context according
>>> to the domain value in the r-uri. In asterisk something like this in
>>> sip.conf worked fine for me:
>>> domain=domain1.com, domain1.com
>>> domain=domain2.com, domain2.com
>>>  Can someone please help me understanding where I am going wrong or have
>>> I mis-understood something?
>>>
>>> Thanks in advance !!
>>>
>>> --- Jayesh
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Sat Apr  3 08:11:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:11:50 -0600
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
Message-ID: <i2v191c3a031004030811q2ed9458dsdf05e10c51dd31f7@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension

The first one transfers the call to the desired exten/dialplan/context and
the 2nd one executes the specified extension in a similar manner and returns
to the same point in the dp.

You make one inbound context and use routing logic from there to decide
which context to transfer to.

Next time, please have a little more patience, I don't like it when people
reply to themselves on the list asking why nobody answered when their
question is only unanswered for 2 days especially during a holiday weekend.



On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> I expected at east one reply saying that the question is stupid, and the
> solution is simple !!
> Any folks who can help me understand only how to achieve this in FS which
> is acheivable in asterisk as follows:
> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
> the call goes to context "mydomain" in dialplan)
> domain = yourdomain.com, yourdomain (if any call has domain as "
> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>
> These calls can come from anywhere, in my case it comes from an Opensips
> instance !!
>
> Thanks for any replies :)
>
> --- Jayesh
>
>
> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi All,
>> I am quite very new to freeswitch and I am kind of playing with it to
>> understand it better.
>> I am primarily using Opensips as registrar and SIP Proxy and intend to use
>> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
>> Opensips is a multi-domain setup and I wish to have all the configuration of
>> media-capabilties segregated domain-wise in the FS too.
>>
>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>> redirect that call to FS IP address keeping the URI intact. I add the
>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>> go to voicemail I do the same thing.
>> The requirement is I want to maintain the dialplans for each domains
>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>> the call should go to dialplan context domain1 and similarly if call from
>> Opensips comes to FS with domain2 the call handling should be mentioned in
>> the domain2 context.
>>
>> The problem is; I am not able to send the calls to respective contexts
>> according to their domains when they come from Opensips. I've read the
>> examples on multi-domain setup and have tried taking some help from that
>> example, but whenever the call comes from Opensips to FS, it tries to go
>> into the context that is defined in the SIP Profile. If i don't mention
>> anything in the SIP Profile, it tries to search for default context.
>> I have tried the following:
>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>> 2) mentioned parameters in domain1.xml as follows:
>> <include>
>> <domain name = "domain1.com"
>> <params>
>> <param name="user_context" value "domain1.com"/>
>> </params>
>> </domain>
>> </include>
>> 3) Similarly done for file domain2.xml.
>>
>> But I am just not able to get the calls to the required context according
>> to the domain value in the r-uri. In asterisk something like this in
>> sip.conf worked fine for me:
>> domain=domain1.com, domain1.com
>> domain=domain2.com, domain2.com
>>  Can someone please help me understanding where I am going wrong or have I
>> mis-understood something?
>>
>> Thanks in advance !!
>>
>> --- Jayesh
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Sat Apr  3 08:15:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:15:19 -0600
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>
References: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>
Message-ID: <x2y191c3a031004030815s34dbf8e0s430c28efcac5e1cc@mail.gmail.com>

We already do it.
X-Lite does not support it.
If you try it with a phone like snom or polycom you will see it works just
like that.


2010/4/3 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi Seven Du.
>
> Thanks to yours suggetion.
>
> I have an ideal, it is: when the call between caller and agent is set, the
> caller_id is determined. So, i want to edit code to sent the agent
> information (the call_id and  call_id_number) which will be displayed
> againt in the agent's softphone (as Xlite..) when the call is happening.
>
> I read some documents but i still can't determine: It's maybe yes or maybe
> to do this and where to do this.
>
> Can you give me some comments.
>
> Best Regard.
>
> Seven Du <dujinfang at gmail.com> [dujinfang at gmail.com]
>
>
> ?As discussed in the list, it's not a freeswitch problem but a reality of
> life.
>
>
> Think about customer A and B calls in one after another, then if
> FreeSWITCH call agent X with caller id A and Y with caller id B, and
> angent Y answers before X, then
>
> 1) if bridge Y with A with the FIFO rule, then the caller id is wrong
> 2) if bridge Y with B, the caller id is right but it breaks the rule
> of FIFO - A should be served before B!! And what even worse is that
> if X never answer A then A never can be served which is really
> unfair!!
>
> Of course you don't want 1), and you don't need mod_fifo if you want
> behavior 2), you just need some dialplan trick or some simple Lua
> script I think. Also FreeSWITCH is designed to be easily extended with
> almost any languages so feel free to implement anything.
>
> 2010/3/31 Nguy?n M?nh H?ng :
>
> > Hi Mike and Seven Du.
> > Thanks to yours help.
> > I known the mechanism of mod_fifo.
> >>>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
> .
> > What a pity, It can't solve this problem. I can't use freeswitch for my
> call
> > center.
> > Hope new version can solve 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Sat Apr  3 08:20:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:20:28 -0600
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
Message-ID: <r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>

yes, it was merged from external to internal.
delete it and update again.


On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

>  The current version of FS I?m using is 17135. I tried to ?make current?
> and it errored out in the svn up portion:
>
>
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked out from
> a different repository
>
>
>
> Should I just delete the libs/openzap directory?
>
>
>
> openzap is commented out in modules.conf.xml.
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Sat Apr  3 08:49:21 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 3 Apr 2010 16:49:21 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
Message-ID: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>

I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office
external-IP, server iptables disabled, and changed the FreeSwitch ACL
domains to "allow", so it's all wide open for now.

In the office I'm trying to connect to the server from Bria/X-lite. I've
entered a stun server (stun.freeswitch.org) and I can now call to the
server, but not from the server. I read this page:
   http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc
which suggested adding this variable to the user config:

http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction

With that on I can now call to and from the server. However with or without
that although I can hear audio from the server, audio to the server isn't
arriving (doesn't appear in recordings), and dtmf doesn't get received
either.

When I hang up from the client, I see in the CLI that it gets that
instruction, so it hasn't started the call and lost all contact with the
softphone, it's receiving some instructions, but not the audio and dtmf.

The problem is that both the server and client are each behind NAT, so
either could be having the problem (on EC2 the auto-NAT doesn't work, so
I've specified the external rtp and sip ip's.. I've also turned on
aggressive-NAT in case that helps. Also I'm connecting to the server by a
sub-domain (A-name) rather than IP.)

I've got almost the same setup working fine on the internal network (same
dialplan and directory, and all the config is the same if it can be), so
it's got to be something to do with the NAT's.

Any suggestions on what the problem might be, or how to find it?

Cheers,
Fraser
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From larclap at yahoo.com  Sat Apr  3 08:56:22 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 08:56:22 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
Message-ID: <008801cad346$3511a240$9f34e6c0$@com>

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.



On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From riedinger at sns.eu  Sat Apr  3 08:14:59 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 03 Apr 2010 17:14:59 +0200
Subject: [Freeswitch-users] Conference Dial In Isn't Working (mod_conference)
Message-ID: <4BB75B73.7040501@sns.eu>

If I dial in to a conference from my Cisco IP Phone 7940 all is working 
as it should.

However, if I dial in via a voip carrier, the calls are disconnect with 
disconnect cause "facility rejected" by the voip carrier. It seems that 
Freeswitch tries to change some parameters of the call setup, which is 
refused. You find below the console log output. Do you have an idea, 
what I have to change to get it working?

I know that it is working in prinicple, because I can setup outgoing 
conferences via the voip carrier.

Thank you very much in advance
     Jan



=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.04.03 16:56:10 
=~=~=~=~=~=~=~=~=~=~=~=
2010-04-03 16:56:07.593334 [NOTICE] switch_channel.c:669 New 
Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
[27446b50-dc2a-4088-83f2-7b7cc8f014e8]
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4164 Remote SDP:
v=0

o=- 1270306567 1270306567 IN IP4 XXX.XX.XX.X6

s=-

c=IN IP4 XXX.XX.XX.X6

t=0 0

m=audio 19188 RTP/AVP 18 4 8 0 101

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[GSM:3:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[GSM:3:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:2354 Set Codec 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3524 Set 2833 
dtmf send/recv payload to 101
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4310 
(sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_INIT
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State INIT
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:83 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:117 
(sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT -> CS_ROUTING
2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State INIT going to sleep
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_ROUTING
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State ROUTING
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:140 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:77 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard ROUTING
2010-04-03 16:56:07.593334 [INFO] mod_dialplan_xml.c:418 
Processing 49XXXXXXX6->49331YYYYYY in context public
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->unloop] continue=false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->outside_call] continue=true
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Absolute 
Condition [outside_call]
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
set(outside_call=true)
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->call_debug] continue=true
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
...
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->sns_conference] continue=false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/ 
break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action answer()
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
set(conference_enforce_security=false)
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
conference(49331YYYYYY-${domain_name}@default)
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:119 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_ROUTING -> CS_EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State ROUTING going to sleep
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] mod_sofia.c:226 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:157 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard EXECUTE
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
set(outside_call=true)
2010-04-03 16:56:07.596335 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 answer()
2010-04-03 16:56:07.596335 [DEBUG] sofia_glue.c:2594 AUDIO RTP 
[sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 17138 -> 
XXX.XX.XX.X6 port 19188 codec: 8 ms: 20
2010-04-03 16:56:07.596335 [DEBUG] switch_rtp.c:1182 Starting 
timer [soft] 160 bytes per 20ms
2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2774 Set 2833 
dtmf send payload to 101
2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2779 Set 2833 
dtmf receive payload to 101
2010-04-03 16:56:07.599335 [DEBUG] mod_sofia.c:636 Local SDP 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6:
v=0
o=FreeSWITCH 1270289429 1270289430 IN IP4 XXX.XX.XX.X7
s=FreeSWITCH
c=IN IP4 XXX.XX.XX.X7
t=0 0
m=audio 17138 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

2010-04-03 16:56:07.599335 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.599335 [NOTICE] mod_dptools.c:719 Channel 
[sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] has been answered
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
set(conference_enforce_security=false)
2010-04-03 16:56:07.599335 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET 
[conference_enforce_security]=[false]
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
conference(49331YYYYYY-XXX.XX.XX.X7 at default)
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'mute' bound to '0'.
2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1616 max len 1
2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1620 min len 1
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'deaf mute' bound to '*'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy up' bound to '9'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy equ' bound to '8'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy dn' bound to '7'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk up' bound to '3'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk zero' bound to '2'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk dn' bound to '1'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen up' bound to '6'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen zero' bound to '5'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen dn' bound to '4'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'hangup' bound to '#'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:4990 Raw 
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5035 Raw 
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 16:56:07.599335 [DEBUG] switch_core_codec.c:122 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:994 Setup 
timer success interval: 20  samples: 160
2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:2202 Setup 
timer soft success interval: 20  samples: 160
2010-04-03 16:56:07.617340 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
2010-04-03 16:56:07.839332 [NOTICE] sofia.c:481 Hangup 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
2010-04-03 16:56:07.839332 [DEBUG] switch_channel.c:2071 Send 
signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [KILL]
2010-04-03 16:56:07.839332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.848341 [DEBUG] mod_conference.c:2473 Channel 
leaving conference, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] switch_core_codec.c:146 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State EXECUTE going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_HANGUP
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State HANGUP
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:408 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 200 
from the other leg
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:414 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 hanging up, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:46 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard HANGUP, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State HANGUP going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:333 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_HANGUP -> CS_REPORTING
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_REPORTING
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State REPORTING
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:53 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard REPORTING, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State REPORTING going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:327 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_REPORTING -> CS_DESTROY
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1161 
Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on 
external entities
2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1179 
Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Ended
2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1181 
Close Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:428 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State DESTROY
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:341 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:60 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State DESTROY going to sleep
2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1361 Write 
Lock ON
2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1364 Write 
Lock OFF


-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                        E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From anthony.minessale at gmail.com  Sat Apr  3 09:16:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 10:16:14 -0600
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <008801cad346$3511a240$9f34e6c0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
	<008801cad346$3511a240$9f34e6c0$@com>
Message-ID: <w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.


On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  I deleted the openzap directory. I ran ?svn up? to make sure I was
> current. Then ran ?make current? and got:
>
>
>
> making uninstall mod_vmd
>
>
>
> making uninstall mod_xml_curl
>
>
>
> making uninstall mod_xml_ldap
>
>
>
> WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....
>
>
>
> making uninstall mod_yaml
>
> cd libs/openzap && autoconf
>
> /bin/sh: line 0: cd: libs/openzap: No such file or directory
>
> make: *** [libs/openzap/Makefile] Error 1
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Saturday, April 03, 2010 8:20 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Error in 'svn up'
>
>
>
> yes, it was merged from external to internal.
> delete it and update again.
>
>  On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:
>
> The current version of FS I?m using is 17135. I tried to ?make current? and
> it errored out in the svn up portion:
>
>
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked out from
> a different repository
>
>
>
> Should I just delete the libs/openzap directory?
>
>
>
> openzap is commented out in modules.conf.xml.
>
>
>
> Thanks, Lars
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From brian at freeswitch.org  Sat Apr  3 09:19:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 11:19:04 -0500
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>

sofia profile xxxxx siptrace on (replace xxx with the profile)

Then try again.

/b

On Apr 3, 2010, at 10:14 AM, Jan Riedinger wrote:

> However, if I dial in via a voip carrier, the calls are disconnect with 
> disconnect cause "facility rejected" by the voip carrier. It seems that 
> Freeswitch tries to change some parameters of the call setup, which is 
> refused. You find below the console log output. Do you have an idea, 
> what I have to change to get it working?



From anthony.minessale at gmail.com  Sat Apr  3 09:23:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 10:23:36 -0600
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <l2h191c3a031004030923x6668e77ena2996e9a3006daaa@mail.gmail.com>

run sofia profile internal siptrace on
and repeat so you can see the sip traffic.


On Sat, Apr 3, 2010 at 9:14 AM, Jan Riedinger <riedinger at sns.eu> wrote:

> If I dial in to a conference from my Cisco IP Phone 7940 all is working
> as it should.
>
> However, if I dial in via a voip carrier, the calls are disconnect with
> disconnect cause "facility rejected" by the voip carrier. It seems that
> Freeswitch tries to change some parameters of the call setup, which is
> refused. You find below the console log output. Do you have an idea,
> what I have to change to get it working?
>
> I know that it is working in prinicple, because I can setup outgoing
> conferences via the voip carrier.
>
> Thank you very much in advance
>     Jan
>
>
>
> =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.04.03 16:56:10
> =~=~=~=~=~=~=~=~=~=~=~=
>  [36m2010-04-03 16:56:07.593334 [NOTICE] switch_channel.c:669 New
> Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> [27446b50-dc2a-4088-83f2-7b7cc8f014e8]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4164 Remote SDP:
> v=0
>
> o=- 1270306567 1270306567 IN IP4 XXX.XX.XX.X6
>
> s=-
>
> c=IN IP4 XXX.XX.XX.X6
>
> t=0 0
>
> m=audio 19188 RTP/AVP 18 4 8 0 101
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[GSM:3:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[GSM:3:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3524 Set 2833
> dtmf send/recv payload to 101
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4310
> (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:83
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:117
> (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT ->
> CS_ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State INIT going to sleep
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:140
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:77 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard ROUTING
>  [m [32m2010-04-03 16:56:07.593334 [INFO] mod_dialplan_xml.c:418
> Processing 49XXXXXXX6->49331YYYYYY in context public
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->unloop] continue=false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS)
> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL)
> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->outside_call] continue=true
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Absolute
> Condition [outside_call]
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> set(outside_call=true)
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->call_debug] continue=true
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL)
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> ...
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->sns_conference] continue=false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS)
> [sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/
> break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action answer()
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> set(conference_enforce_security=false)
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> conference(49331YYYYYY-${domain_name}@default)
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:119 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_ROUTING -> CS_EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State ROUTING going to sleep
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] mod_sofia.c:226
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:157 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard EXECUTE
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> set(outside_call=true)
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] mod_dptools.c:816
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 answer()
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] sofia_glue.c:2594 AUDIO RTP
> [sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 17138 ->
> XXX.XX.XX.X6 port 19188 codec: 8 ms: 20
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] switch_rtp.c:1182 Starting
> timer [soft] 160 bytes per 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2774 Set 2833
> dtmf send payload to 101
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2779 Set 2833
> dtmf receive payload to 101
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_sofia.c:636 Local SDP
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6:
> v=0
> o=FreeSWITCH 1270289429 1270289430 IN IP4 XXX.XX.XX.X7
> s=FreeSWITCH
> c=IN IP4 XXX.XX.XX.X7
> t=0 0
> m=audio 17138 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
>
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [36m2010-04-03 16:56:07.599335 [NOTICE] mod_dptools.c:719 Channel
> [sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] has been answered
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> set(conference_enforce_security=false)
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_dptools.c:816
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET
> [conference_enforce_security]=[false]
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> conference(49331YYYYYY-XXX.XX.XX.X7 at default)
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'mute' bound to '0'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1616 max len 1
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1620 min len 1
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'deaf mute' bound to '*'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy up' bound to '9'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy equ' bound to '8'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy dn' bound to '7'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk up' bound to '3'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk zero' bound to '2'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk dn' bound to '1'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen up' bound to '6'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen zero' bound to '5'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen dn' bound to '4'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'hangup' bound to '#'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:4990 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5035 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_codec.c:122
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:994 Setup
> timer success interval: 20  samples: 160
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:2202 Setup
> timer soft success interval: 20  samples: 160
>  [m [33m2010-04-03 16:56:07.617340 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
>  [m [36m2010-04-03 16:56:07.839332 [NOTICE] sofia.c:481 Hangup
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
>  [m [33m2010-04-03 16:56:07.839332 [DEBUG] switch_channel.c:2071 Send
> signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [KILL]
>  [m [33m2010-04-03 16:56:07.839332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.848341 [DEBUG] mod_conference.c:2473 Channel
> leaving conference, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_codec.c:146
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State EXECUTE going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_HANGUP
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State HANGUP
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:408
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 200
> from the other leg
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:414 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 hanging up, cause:
> FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:46 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard HANGUP, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State HANGUP going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:333 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_HANGUP -> CS_REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:53 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard REPORTING, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State REPORTING going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:327 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_REPORTING -> CS_DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1161
> Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on
> external entities
>  [m [36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1179
> Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Ended
>  [m [36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1181
> Close Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:428 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:341
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:60 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State DESTROY going to sleep
>  [m [33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1361 Write
> Lock ON
>  [m [33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1364 Write
> Lock OFF
>
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                        E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From frank at carmickle.com  Sat Apr  3 09:33:05 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 3 Apr 2010 12:33:05 -0400
Subject: [Freeswitch-users] Conference Dial In Isn't
	Working	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <20100403163305.GE5746@base.carmickle.com>

On Sat, Apr 03, Jan Riedinger wrote:
> If I dial in to a conference from my Cisco IP Phone 7940 all is working 
> as it should.
> 
> However, if I dial in via a voip carrier, the calls are disconnect with 
> disconnect cause "facility rejected" by the voip carrier. It seems that 
> Freeswitch tries to change some parameters of the call setup, which is 
> refused. You find below the console log output. Do you have an idea, 
> what I have to change to get it working?

Make sure you answer the channel first.  But with out full logging we don't know what's going on.  Do as BKW says.

--FC


From riedinger at sns.eu  Sat Apr  3 10:00:34 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 03 Apr 2010 19:00:34 +0200
Subject: [Freeswitch-users] Conference Dial In Isn't
	Working	(mod_conference)
In-Reply-To: <E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
References: <4BB75B73.7040501@sns.eu>
	<E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
Message-ID: <4BB77432.8080508@sns.eu>

Here is the missing debug information.

Thanks
    Jan

freeswitch at ...> recv 839 bytes from udp/[XXX.XX.XX.X6]:5060 at 
16:48:12.118650:
   ------------------------------------------------------------------------
   INVITE sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   Contact: <sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Cisco-Guid: 2962728880-1061163487-2855600256-2188435832
   Content-Type: application/sdp
   Content-Length:   264

   v=0
   o=- 1270313292 1270313292 IN IP4 XXX.XX.XX.X6
   s=-
   c=IN IP4 XXX.XX.XX.X6
   t=0 0
   m=audio 22284 RTP/AVP 18 4 8 0 101
   a=rtpmap:18 G729/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 392 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.118936:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.116629 [NOTICE] switch_channel.c:669 New Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
[d1242848-5588-4083-acea-062fd74e5f66]
2010-04-03 18:48:12.116629 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
2010-04-03 18:48:12.116629 [DEBUG] sofia.c:4164 Remote SDP:
v=0
o=- 1270313292 1270313292 IN IP4 XXX.XX.XX.X6
s=-
c=IN IP4 XXX.XX.XX.X6
t=0 0
m=audio 22284 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_NEW
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[GSM:3:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[GSM:3:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:2354 Set Codec 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf 
send/recv payload to 101
2010-04-03 18:48:12.119630 [DEBUG] sofia.c:4310 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:320 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State NEW
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_INIT
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State INIT
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:83 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:117 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT -> CS_ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State INIT going to sleep
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State ROUTING
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:140 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:77 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard ROUTING
2010-04-03 18:48:12.119630 [INFO] mod_dialplan_xml.c:418 Processing 
49XXXXXXXX6->49331YYYYYY in context public
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->unloop] continue=false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->outside_call] continue=true
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Absolute Condition 
[outside_call]
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
set(outside_call=true)
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->call_debug] continue=true
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
...
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->sns_conference] continue=false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/ 
break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action answer()
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
set(conference_enforce_security=false)
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
conference(49331YYYYYY-${domain_name}@default)
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:119 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_ROUTING -> 
CS_EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State ROUTING going to sleep
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:348 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:226 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:157 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard EXECUTE
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 set(outside_call=true)
2010-04-03 18:48:12.119630 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 answer()
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:2594 AUDIO RTP 
[sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 21986 -> 
XXX.XX.XX.X6 port 22284 codec: 8 ms: 20
2010-04-03 18:48:12.119630 [DEBUG] switch_rtp.c:1182 Starting timer 
[soft] 160 bytes per 20ms
2010-04-03 18:48:12.122631 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send 
payload to 101
2010-04-03 18:48:12.122631 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf 
receive payload to 101
2010-04-03 18:48:12.122631 [DEBUG] mod_sofia.c:636 Local SDP 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6:
v=0
o=FreeSWITCH 1270291306 1270291307 IN IP4 XXX.XX.XX.X7
s=FreeSWITCH
c=IN IP4 XXX.XX.XX.X7
t=0 0
m=audio 21986 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

2010-04-03 18:48:12.122631 [DEBUG] switch_core_session.c:638 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.122631 [NOTICE] mod_dptools.c:719 Channel 
[sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6] has been answered
send 1068 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.124920:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   Contact: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   Remote-Party-ID: "49331YYYYYY" 
<sip:49331YYYYYY at XXX.XX.XX.X7>;party=calling;privacy=off;screen=no

   v=0
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
set(conference_enforce_security=false)
   o=FreeSWITCH 1270291306 1270291307 IN IP4 XXX.XX.XX.X7
   s=FreeSWITCH
   c=IN IP4 XXX.XX.XX.X7
   t=0 0
   m=audio 21986 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
   ------------------------------------------------------------------------
2010-04-03 18:48:12.122631 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
2010-04-03 18:48:12.122631 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SET 
[conference_enforce_security]=[false]
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
conference(49331YYYYYY-XXX.XX.XX.X7 at default)
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'mute' bound to '0'.
2010-04-03 18:48:12.122631 [DEBUG] switch_ivr.c:1616 max len 1
2010-04-03 18:48:12.122631 [DEBUG] switch_ivr.c:1620 min len 1
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'deaf mute' bound to '*'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy up' bound to '9'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy equ' bound to '8'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy dn' bound to '7'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk up' bound to '3'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk zero' bound to '2'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk dn' bound to '1'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen up' bound to '6'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen zero' bound to '5'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen dn' bound to '4'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'hangup' bound to '#'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:4990 Raw Codec 
Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5035 Raw Codec 
Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 18:48:12.125638 [DEBUG] switch_core_codec.c:122 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
2010-04-03 18:48:12.125638 [DEBUG] switch_core_session.c:638 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.125638 [DEBUG] mod_conference.c:2202 Setup timer 
soft success interval: 20  samples: 160
2010-04-03 18:48:12.125638 [DEBUG] mod_conference.c:994 Setup timer 
success interval: 20  samples: 160
recv 445 bytes from udp/[XXX.XX.XX.X6]:5060 at 16:48:12.144223:
   ------------------------------------------------------------------------
   ACK sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 ACK
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.143633 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
recv 446 bytes from udp/[XXX.XX.XX.X6]:5060 at 16:48:12.311539:
   ------------------------------------------------------------------------
   BYE sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP XXX.XX.XX.X6:5060
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 2 BYE
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Reason: Q.850;cause=29;text="Facility rejected"
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.308666 [NOTICE] sofia.c:481 Hangup 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
2010-04-03 18:48:12.308666 [DEBUG] switch_channel.c:2071 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [KILL]
2010-04-03 18:48:12.308666 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
send 493 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.311906:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP XXX.XX.XX.X6:5060
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 2 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.326660 [DEBUG] mod_conference.c:2473 Channel leaving 
conference, cause: FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_codec.c:146 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:348 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State EXECUTE going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_HANGUP
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State HANGUP
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:408 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 
200 from the other leg
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:414 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 hanging up, cause: FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:46 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard HANGUP, cause: 
FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State HANGUP going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:333 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_HANGUP -> 
CS_REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:53 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard REPORTING, cause: 
FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State REPORTING going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:327 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_REPORTING -> 
CS_DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1161 Session 8 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on external 
entities
2010-04-03 18:48:12.329664 [NOTICE] switch_core_session.c:1179 Session 8 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Ended
2010-04-03 18:48:12.329664 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:428 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State DESTROY
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:341 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:60 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State DESTROY going to sleep
2010-04-03 18:48:12.347672 [DEBUG] mod_conference.c:1361 Write Lock ON
2010-04-03 18:48:12.347672 [DEBUG] mod_conference.c:1364 Write Lock OFF

Brian West schrieb:
> sofia profile xxxxx siptrace on (replace xxx with the profile)
>
> Then try again.
>
> /b
>
> On Apr 3, 2010, at 10:14 AM, Jan Riedinger wrote:
>
>   
>> However, if I dial in via a voip carrier, the calls are disconnect with 
>> disconnect cause "facility rejected" by the voip carrier. It seems that 
>> Freeswitch tries to change some parameters of the call setup, which is 
>> refused. You find below the console log output. Do you have an idea, 
>> what I have to change to get it working?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                        E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From brian at freeswitch.org  Sat Apr  3 10:05:27 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 12:05:27 -0500
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB77432.8080508@sns.eu>
References: <4BB75B73.7040501@sns.eu>
	<E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
	<4BB77432.8080508@sns.eu>
Message-ID: <FA3A06CA-92D8-4BEC-9AE9-7E881D45E3E3@freeswitch.org>

  Reason: Q.850;cause=29;text="Facility rejected"

Contact your provider their MERA switch sent you a BYE.

/b

On Apr 3, 2010, at 12:00 PM, Jan Riedinger wrote:

> Here is the missing debug information.
> 
> Thanks
>    Jan



From larclap at yahoo.com  Sat Apr  3 11:22:50 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 11:22:50 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>
	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
Message-ID: <00b901cad35a$ab6857e0$023907a0$@com>

Sorry to be a nuisance with this. No, I do not use openzap.

 

I tried to execute 'make oz-reconf' in an attempt to get around the error,
but no go.

 

[root at fs freeswitch]# make oz-reconf

cd libs/openzap && make clean

make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'

make[1]: *** No rule to make target `clean'.  Stop.

make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'

make: *** [oz-reconf] Error 2

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 9:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.



On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.

On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From mrene_lists at avgs.ca  Sat Apr  3 11:25:44 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 3 Apr 2010 14:25:44 -0400
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00b901cad35a$ab6857e0$023907a0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>
	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
	<00b901cad35a$ab6857e0$023907a0$@com>
Message-ID: <D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>

Re-run ./configure from the top level directory

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-03, at 2:22 PM, Lars Zeb wrote:

> Sorry to be a nuisance with this. No, I do not use openzap.
>
> I tried to execute ?make oz-reconf? in an attempt to get around the  
> error, but no go.
>
> [root at fs freeswitch]# make oz-reconf
> cd libs/openzap && make clean
> make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'
> make[1]: *** No rule to make target `clean'.  Stop.
> make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'
> make: *** [oz-reconf] Error 2
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Anthony Minessale
> Sent: Saturday, April 03, 2010 9:16 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Error in 'svn up'
>
> if you actually want to use openzap after you delete it and svn up,  
> issue
>
> make oz-reconf
>
> then make as usual.
>
>
> On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:
> I deleted the openzap directory. I ran ?svn up? to make sure I was  
> current. Then ran ?make current? and got:
>
> making uninstall mod_vmd
>
> making uninstall mod_xml_curl
>
> making uninstall mod_xml_ldap
>
> WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping  
> it.....
>
> making uninstall mod_yaml
> cd libs/openzap && autoconf
> /bin/sh: line 0: cd: libs/openzap: No such file or directory
> make: *** [libs/openzap/Makefile] Error 1
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Anthony Minessale
> Sent: Saturday, April 03, 2010 8:20 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Error in 'svn up'
>
> yes, it was merged from external to internal.
> delete it and update again.
>
> On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com>  
> wrote:
> The current version of FS I?m using is 17135. I tried to ?make  
> current? and it errored out in the svn up portion:
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked  
> out from a different repository
>
> Should I just delete the libs/openzap directory?
>
> openzap is commented out in modules.conf.xml.
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From jim at k4gvo.com  Sat Apr  3 12:17:44 2010
From: jim at k4gvo.com (Jim)
Date: Sat, 03 Apr 2010 15:17:44 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
Message-ID: <4BB79458.8080601@k4gvo.com>

I obviously need to set a <variable name="toll_allow" 
value="domestic,international,local"/> somewhere but I can't figure out 
what file to put it in.  The examples all show it in the 
directory/default/xxxx.xml files but those appear to be sip only.  In 
any event creating files in that directory for my extension did nothing 
to help the problem.

The only places I have the extension mentioned is in  the openzap.conf 
file and dialplan/default/00_incoming-1.xml.  Adding a <variables ... 
element to the latter does nothing.

How do I set that variable?  Or where?

Thanks,
Jim.


From gavin.henry at gmail.com  Sat Apr  3 12:30:05 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 3 Apr 2010 20:30:05 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
Message-ID: <r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>

On 3 April 2010 05:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Nothing is wrong with it,? there is no revision that broke it or there would
> be 100,000 people saying it was broken.

Unless I was the first that hit a weird configuration. That's the
nature of building software yourself
and these types of e-mails should be encouraged, not frowned upon.

> Change you perspective to looking for the problem on your end and you will
> have more luck finding your problem.

Of course, I know. But sometimes many eyes help and it's easier to ask
a quick question.

> Backup your config by moving it out of the way, re-install and try the
> defaults.

Cheers.

> We use FS with nat like this 12 hours a day.

Good to know.

Thanks all!

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From anthony.minessale at gmail.com  Sat Apr  3 12:51:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 13:51:49 -0600
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
Message-ID: <j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>

I didn't frown upon you I am being serious and straightforward with you.
I am giving you advice I have learned the hard way from personal experience
on troubleshooting.
The facts and pointers lead to a quicker solution and eliminate variables.


On Sat, Apr 3, 2010 at 1:30 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 3 April 2010 05:18, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> > Nothing is wrong with it,  there is no revision that broke it or there
> would
> > be 100,000 people saying it was broken.
>
> Unless I was the first that hit a weird configuration. That's the
> nature of building software yourself
> and these types of e-mails should be encouraged, not frowned upon.
>
> > Change you perspective to looking for the problem on your end and you
> will
> > have more luck finding your problem.
>
> Of course, I know. But sometimes many eyes help and it's easier to ask
> a quick question.
>
> > Backup your config by moving it out of the way, re-install and try the
> > defaults.
>
> Cheers.
>
> > We use FS with nat like this 12 hours a day.
>
> Good to know.
>
> Thanks all!
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From larclap at yahoo.com  Sat Apr  3 13:09:29 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 13:09:29 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>
	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
Message-ID: <00e501cad369$91318950$b3949bf0$@com>

 

configure: creating ./config.status

config.status: creating Makefile

config.status: creating doc/Makefile

config.status: creating test-data/Makefile

config.status: creating test-data/local/Makefile

config.status: creating test-data/itu/Makefile

config.status: creating src/Makefile

config.status: creating src/g722_1.h

config.status: creating tests/Makefile

config.status: creating g722_1.spec

config.status: creating tests/regression_tests.sh

config.status: creating src/config.h

config.status: src/config.h is unchanged

config.status: executing depfiles commands

configure: configuring in libs/silk

configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch
CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS=''
--cache-file=/dev/null --srcdir=.

./configure.gnu: line 3: ./configure: No such file or directory

configure: error: /bin/sh './configure.gnu' failed for libs/silk

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu
Rene
Sent: Saturday, April 03, 2010 11:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

Re-run ./configure from the top level directory

 

Mathieu Rene

Avant-Garde Solutions Inc

Office: + 1 (514) 664-1044 x100

Cell: +1 (514) 664-1044 x200

mrene at avgs.ca

 

 

 

 

On 2010-04-03, at 2:22 PM, Lars Zeb wrote:





Sorry to be a nuisance with this. No, I do not use openzap.

 

I tried to execute 'make oz-reconf' in an attempt to get around the error,
but no go.

 

[root at fs freeswitch]# make oz-reconf

cd libs/openzap && make clean

make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'

make[1]: *** No rule to make target `clean'.  Stop.

make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'

make: *** [oz-reconf] Error 2

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 9:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.




On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.

On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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

 

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From gavin.henry at gmail.com  Sat Apr  3 13:34:20 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 3 Apr 2010 21:34:20 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
Message-ID: <x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>

On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> I didn't frown upon you I am being serious and straightforward with you.
> I am giving you advice I have learned the hard way from personal experience
> on troubleshooting.
> The facts and pointers lead to a quicker solution and eliminate variables.

I know Anthony, and appreciate it! Just mentioning in case others get
scared of posting
things if they get shouted at :-) We have this problem in other projects.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From brian at freeswitch.org  Sat Apr  3 13:43:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 15:43:33 -0500
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00e501cad369$91318950$b3949bf0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>
	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
	<00e501cad369$91318950$b3949bf0$@com>
Message-ID: <23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>

You have to rebootstrap after you update... I missed the part to do libs/silk in configure.in

/b

On Apr 3, 2010, at 3:09 PM, Lars Zeb wrote:

>  
> configure: creating ./config.status
> config.status: creating Makefile
> config.status: creating doc/Makefile
> config.status: creating test-data/Makefile
> config.status: creating test-data/local/Makefile
> config.status: creating test-data/itu/Makefile
> config.status: creating src/Makefile
> config.status: creating src/g722_1.h
> config.status: creating tests/Makefile
> config.status: creating g722_1.spec
> config.status: creating tests/regression_tests.sh
> config.status: creating src/config.h
> config.status: src/config.h is unchanged
> config.status: executing depfiles commands
> configure: configuring in libs/silk
> configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch  CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS='' --cache-file=/dev/null --srcdir=.
> ./configure.gnu: line 3: ./configure: No such file or directory
> configure: error: /bin/sh './configure.gnu' failed for libs/silk
>  

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From larclap at yahoo.com  Sat Apr  3 15:08:47 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 15:08:47 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>	<00e501cad369$91318950$b3949bf0$@com>
	<23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>
Message-ID: <011301cad37a$3d9fbcb0$b8df3610$@com>

That did it, thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Saturday, April 03, 2010 1:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

You have to rebootstrap after you update... I missed the part to do
libs/silk in configure.in

 

/b

 

On Apr 3, 2010, at 3:09 PM, Lars Zeb wrote:





 

configure: creating ./config.status

config.status: creating Makefile

config.status: creating doc/Makefile

config.status: creating test-data/Makefile

config.status: creating test-data/local/Makefile

config.status: creating test-data/itu/Makefile

config.status: creating src/Makefile

config.status: creating src/g722_1.h

config.status: creating tests/Makefile

config.status: creating g722_1.spec

config.status: creating tests/regression_tests.sh

config.status: creating src/config.h

config.status: src/config.h is unchanged

config.status: executing depfiles commands

configure: configuring in libs/silk

configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch
CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS=''
--cache-file=/dev/null --srcdir=.

./configure.gnu: line 3: ./configure: No such file or directory

configure: error: /bin/sh './configure.gnu' failed for libs/silk

 

 

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From jayesh.voip at gmail.com  Sat Apr  3 20:47:58 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Sun, 4 Apr 2010 09:17:58 +0530
Subject: [Freeswitch-users]  domain-wise context
Message-ID: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>

Hi,
Sorry for sounding so impatient, the anxiety only grew because before
posting it to the list I spent a week on reading all the documentation
available on and around this topic FS site and mailing lists. I really
appreciate and am thankful for your suggestions.
I'll try out the suggestions given by you. Is there a way that we can
compare the domain name in the dialplan using regular expressions. Is there
a value in condition tag that can be used to compare this, something like
"extension_number".
Or should I store the domain value in some variable(something like
sip_h_<domain_name>) and compare that variable to take the call to a
different context?

Thanks,

--- Jayesh

---------- Forwarded message ----------

> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 3 Apr 2010 09:11:50 -0600
> Subject: Re: [Freeswitch-users] domain-wise context
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension
>
> The first one transfers the call to the desired exten/dialplan/context and
> the 2nd one executes the specified extension in a similar manner and returns
> to the same point in the dp.
>
> You make one inbound context and use routing logic from there to decide
> which context to transfer to.
>
> Next time, please have a little more patience, I don't like it when people
> reply to themselves on the list asking why nobody answered when their
> question is only unanswered for 2 days especially during a holiday weekend.
>
>
>
> On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> I expected at east one reply saying that the question is stupid, and the
>> solution is simple !!
>> Any folks who can help me understand only how to achieve this in FS which
>> is acheivable in asterisk as follows:
>> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
>> the call goes to context "mydomain" in dialplan)
>> domain = yourdomain.com, yourdomain (if any call has domain as "
>> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>>
>> These calls can come from anywhere, in my case it comes from an Opensips
>> instance !!
>>
>> Thanks for any replies :)
>>
>> --- Jayesh
>>
>>
>> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>>
>>> Hi All,
>>> I am quite very new to freeswitch and I am kind of playing with it to
>>> understand it better.
>>> I am primarily using Opensips as registrar and SIP Proxy and intend to
>>> use FS as media server handling voicemails, IVR, Announcements, MeetMe etc.
>>> My Opensips is a multi-domain setup and I wish to have all the configuration
>>> of media-capabilties segregated domain-wise in the FS too.
>>>
>>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>>> redirect that call to FS IP address keeping the URI intact. I add the
>>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>>> go to voicemail I do the same thing.
>>> The requirement is I want to maintain the dialplans for each domains
>>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>>> the call should go to dialplan context domain1 and similarly if call from
>>> Opensips comes to FS with domain2 the call handling should be mentioned in
>>> the domain2 context.
>>>
>>> The problem is; I am not able to send the calls to respective contexts
>>> according to their domains when they come from Opensips. I've read the
>>> examples on multi-domain setup and have tried taking some help from that
>>> example, but whenever the call comes from Opensips to FS, it tries to go
>>> into the context that is defined in the SIP Profile. If i don't mention
>>> anything in the SIP Profile, it tries to search for default context.
>>> I have tried the following:
>>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>>> 2) mentioned parameters in domain1.xml as follows:
>>> <include>
>>> <domain name = "domain1.com"
>>> <params>
>>> <param name="user_context" value "domain1.com"/>
>>> </params>
>>> </domain>
>>> </include>
>>> 3) Similarly done for file domain2.xml.
>>>
>>> But I am just not able to get the calls to the required context according
>>> to the domain value in the r-uri. In asterisk something like this in
>>> sip.conf worked fine for me:
>>> domain=domain1.com, domain1.com
>>> domain=domain2.com, domain2.com
>>>  Can someone please help me understanding where I am going wrong or have
>>> I mis-understood something?
>>>
>>> Thanks in advance !!
>>>
>>> --- Jayesh
>>>
>>
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From brian at freeswitch.org  Sat Apr  3 20:57:19 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 22:57:19 -0500
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>
References: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>
Message-ID: <1A7D6A7C-B8A5-46B5-AFB1-C94ED55C8F30@freeswitch.org>

see user_context variable on the user.

/b

On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:

> Hi,
> Sorry for sounding so impatient, the anxiety only grew because before posting it to the list I spent a week on reading all the documentation available on and around this topic FS site and mailing lists. I really appreciate and am thankful for your suggestions.
> I'll try out the suggestions given by you. Is there a way that we can compare the domain name in the dialplan using regular expressions. Is there a value in condition tag that can be used to compare this, something like "extension_number".
> Or should I store the domain value in some variable(something like sip_h_<domain_name>) and compare that variable to take the call to a different context?
> 
> Thanks,
> 
> --- Jayesh



From mayamatakeshi at gmail.com  Sat Apr  3 21:31:57 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 4 Apr 2010 13:31:57 +0900
Subject: [Freeswitch-users] Contrib modules
Message-ID: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>

Hi,
I have updated (svn) to trunk (r. 17188) and now I don't see the contrib
folder anymore (I was using mod_xml_odbc from there).
What happened?

regards,
takeshi
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From brian at freeswitch.org  Sat Apr  3 21:37:50 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 23:37:50 -0500
Subject: [Freeswitch-users] Contrib modules
In-Reply-To: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>
References: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>
Message-ID: <21FDFE0D-4743-4B01-9F75-3C898FEBB15C@freeswitch.org>

They are being split out into a repo.  That i'm not sure is complete yet.

/b

On Apr 3, 2010, at 11:31 PM, mayamatakeshi wrote:

> Hi,
> I have updated (svn) to trunk (r. 17188) and now I don't see the contrib folder anymore (I was using mod_xml_odbc from there).
> What happened?
> 
> regards,
> takeshi



From msc at freeswitch.org  Sat Apr  3 23:29:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 3 Apr 2010 23:29:23 -0700
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <4BB79458.8080601@k4gvo.com>
References: <4BB79458.8080601@k4gvo.com>
Message-ID: <s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>

Variables set in the directory/default/xxxx.xml files apply to users who
make authenticated calls through FS. Generally those will be SIP phones.

Let's back up a step. What problem are you trying to solve, i.e., why is it
that you need to set the toll_allow variable? What endpoint is making an
openzap call?

-MC

On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com> wrote:

> I obviously need to set a <variable name="toll_allow"
> value="domestic,international,local"/> somewhere but I can't figure out
> what file to put it in.  The examples all show it in the
> directory/default/xxxx.xml files but those appear to be sip only.  In
> any event creating files in that directory for my extension did nothing
> to help the problem.
>
> The only places I have the extension mentioned is in  the openzap.conf
> file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
> element to the latter does nothing.
>
> How do I set that variable?  Or where?
>
> Thanks,
> Jim.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Sun Apr  4 00:08:38 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sun, 4 Apr 2010 09:08:38 +0200
Subject: [Freeswitch-users] cepstral PB
Message-ID: <20100404090838.414c58ce@anubis.defcon1>

FS svn-17188M
Debian Lenny
=================

Hi list,

I've installed cepstral voices (in /opt/swift, and also changed its
rights recursively to freeswitch:freeswitch) but I had a complain
at FS start that some cepstral libraries were missing.

Eventually, I was obliged to simlink /opt/swift/lib/lib* into 
/usr/local/freeswitch/lib to have it working correctly, is it normal, 
or did I miss something?

(I followed the wiki, so all libs were added to ld.so.cache, path in
mod_cepstral source Makefile are good, SWIFT_HOME is defined, module
flite loading is commented, etc- so I said to myself: Nantidiou!)

-- 
Sorry never means having your say to love.


From jim at k4gvo.com  Sun Apr  4 04:48:57 2010
From: jim at k4gvo.com (Jim)
Date: Sun, 04 Apr 2010 07:48:57 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
References: <4BB79458.8080601@k4gvo.com>
	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
Message-ID: <4BB87CA9.3050403@k4gvo.com>

Michael Collins wrote:
> Variables set in the directory/default/xxxx.xml files apply to users 
> who make authenticated calls through FS. Generally those will be SIP 
> phones.
>
> Let's back up a step. What problem are you trying to solve, i.e., why 
> is it that you need to set the toll_allow variable? What endpoint is 
> making an openzap call?
>
> -MC
Hi, Michael,

I have multiple sip phone that are working fine.  When I dial a 10 digit 
number they connect with my sip provider and place the call.  The 
openzap configured phone gets dial tone and can call other extensions, 
however when I dial a 10 digit number it gives me a busy.  In the log I 
see it appears to be failing on the toll_allow test:

Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call] 
continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call] 
${toll_allow}() =~ /local/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call] 
continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call] 
${toll_allow}() =~ /domestic/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing 
[default->international.example.com] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) 
[international.example.com] ${toll_allow}() =~ /international/ 
break=on-false

This is the area when it should be placing the call, I belive.  When 
placing a call from a sip phone I see:

Dialplan: sofia/internal/1003 at 192.168.2.51 parsing [default->local_call] 
continue=false
Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call] 
${toll_allow}(domestic,international,local) =~ /local/ break=on-false
Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call] 
destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(continue_on_fail=true)
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
bridge(sofia/gateway/${default_gateway}/7707190068)

I simply want this extension to be able to dial out.  The configuration 
is 99% default. 

Thanks,
Jim.
>
> On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com 
> <mailto:jim at k4gvo.com>> wrote:
>
>     I obviously need to set a <variable name="toll_allow"
>     value="domestic,international,local"/> somewhere but I can't
>     figure out
>     what file to put it in.  The examples all show it in the
>     directory/default/xxxx.xml files but those appear to be sip only.  In
>     any event creating files in that directory for my extension did
>     nothing
>     to help the problem.
>
>     The only places I have the extension mentioned is in  the openzap.conf
>     file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
>     element to the latter does nothing.
>
>     How do I set that variable?  Or where?
>
>     Thanks,
>     Jim.
>
>     _______________________________________________
>     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 vfclists at googlemail.com  Sun Apr  4 06:16:34 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 4 Apr 2010 14:16:34 +0100
Subject: [Freeswitch-users] Freeswitch for Windows - command history too
	short?
Message-ID: <n2i7abab2411004040616w5a1eef47j392099111bdcc52c@mail.gmail.com>

I am using svn 17048 for Windows and the command history is too short
and doesn't persist between restarts.

Are there some configuration settings to fix that?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From jeff at jefflenk.com  Sun Apr  4 09:07:49 2010
From: jeff at jefflenk.com (Jeff Lenk )
Date: Sun, 4 Apr 2010 16:07:49 +0000
Subject: [Freeswitch-users] Freeswitch for Windows - command history too
 short?
Message-ID: <COL117-DS119D271516A01C4A3F96FEB91A0@phx.gbl>

No there is no config setting for that yet - feature does not exist yet


-----Original Message-----
From: Frank Church
Sent: 4/4/2010 1:16:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch for Windows - command history too short?
I am using svn 17048 for Windows and the command history is too short
and doesn't persist between restarts.

Are there some configuration settings to fix that?

--
Frank Church

=======================
http://devblog.brahmancreations.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 lloyd.aloysius at gmail.com  Sun Apr  4 09:38:58 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 12:38:58 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
Message-ID: <g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>

Hi All,

All Aastra Phones Behind the NAT stop working after update to the most
recent version.

freeswitch at internal> version
FreeSWITCH Version 1.0.head (svn-17188)

Aastra 57i
Aastra 9133i

Also I can confirm the Xlite is working without any problem.

Thanks
Lloyd



On Sat, Apr 3, 2010 at 4:34 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> > I didn't frown upon you I am being serious and straightforward with you.
> > I am giving you advice I have learned the hard way from personal
> experience
> > on troubleshooting.
> > The facts and pointers lead to a quicker solution and eliminate
> variables.
>
> I know Anthony, and appreciate it! Just mentioning in case others get
> scared of posting
> things if they get shouted at :-) We have this problem in other projects.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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 max.clark at gmail.com  Sun Apr  4 11:12:16 2010
From: max.clark at gmail.com (Max Clark)
Date: Sun, 4 Apr 2010 11:12:16 -0700
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <20100402092310.GA18680@jdc.jasonjgw.net>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
Message-ID: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>

Clarification - for G729 does freeswitch need to be in "bypass media"
or "proxy media"? My understanding was that G729 would work with
"proxy media" enabled and without the new fangled module?

-Max

On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
> Frank Church <vfclists at googlemail.com> wrote:
>> I am just trialling Freeswitch with Linksys adapters, whose default
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>> When I change that setting to 'yes' the calls don't go through. I am
>> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lloyd.aloysius at gmail.com  Sun Apr  4 11:13:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:13:46 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
Message-ID: <t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>

The following version working without any problem.

freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17155)

------------------------------------------------------ Additional Bug
Informations
----------------------------------------------------------------

Also in freeswitch at internal> version
FreeSWITCH Version 1.0.head (svn-17188)


Here is the Dial plan issues.

Extension 201 - Aastra 57i

Extension 202 - Xlite

Extension 202 -> 201 working great.

2010-04-04 13:54:21.012078 [INFO] mod_dialplan_xml.c:418 Processing 202->201
in context dev.abc.ca


Extension 201 -> 202

*2010-04-04 13:55:08.711996 [INFO] mod_dialplan_xml.c:418 Processing
201->202 in context public*

I do not know why it is looking into the public context.

Thanks
Lloyd



On Sun, Apr 4, 2010 at 12:38 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> All Aastra Phones Behind the NAT stop working after update to the most
> recent version.
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (svn-17188)
>
> Aastra 57i
> Aastra 9133i
>
> Also I can confirm the Xlite is working without any problem.
>
> Thanks
> Lloyd
>
>
>
> On Sat, Apr 3, 2010 at 4:34 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> > I didn't frown upon you I am being serious and straightforward with you.
>> > I am giving you advice I have learned the hard way from personal
>> experience
>> > on troubleshooting.
>> > The facts and pointers lead to a quicker solution and eliminate
>> variables.
>>
>> I know Anthony, and appreciate it! Just mentioning in case others get
>> scared of posting
>> things if they get shouted at :-) We have this problem in other projects.
>>
>> --
>> http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.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 brian at freeswitch.org  Sun Apr  4 11:18:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:18:57 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
Message-ID: <CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>

because its not authenticated.

/b

On Apr 4, 2010, at 1:13 PM, Aloysius Lloyd wrote:

> I do not know why it is looking into the public context.



From brian at freeswitch.org  Sun Apr  4 11:19:17 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:19:17 -0500
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
Message-ID: <3ABB861A-D39E-4574-B954-0EAC338323C3@freeswitch.org>

proxy media isn't needed.

/b

On Apr 4, 2010, at 1:12 PM, Max Clark wrote:

> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
> 
> -Max



From david.ponzone at gmail.com  Sun Apr  4 11:35:39 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 4 Apr 2010 20:35:39 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
Message-ID: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>

No, FreeSWITCH does NOT need to be in bypass media or proxy media.
You just need the regular passthrough module: mod_g729 and to allow  
G729 as inbound and outbound codecs in vars.xml.

To summarize:
-normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the  
RTP path, it relays the audio stream between endpoints, but can still  
detect DTMFs
-proxy media enabled: FreeSWITCH relays the audio stream  
transparently, DTMF detection is impossible. In this mode, FS is  
really a "dumb" transparent RTP-forwarder (this is required to get T38  
working between the 2 endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 04/04/2010 ? 20:12, Max Clark a ?crit :

> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
> wrote:
>> Frank Church <vfclists at googlemail.com> wrote:
>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>> When I change that setting to 'yes' the calls don't go through. I am
>>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you really
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at gmail.com  Sun Apr  4 11:45:36 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:45:36 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
Message-ID: <s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>

There is no settings changed.

 Aastra 57i phone is registered to the freeswitch.

Aastra 9133i no more registered to freeswitch.


Is there any major change between 17155 & 17188

Thanks
Lloyd



On Sun, Apr 4, 2010 at 2:18 PM, Brian West <brian at freeswitch.org> wrote:

> because its not authenticated.
>
> /b
>
> On Apr 4, 2010, at 1:13 PM, Aloysius Lloyd wrote:
>
> > I do not know why it is looking into the public context.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 11:49:42 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:49:42 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
Message-ID: <79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>

Its an aastra... I don't trust it.

/b

On Apr 4, 2010, at 1:45 PM, Aloysius Lloyd wrote:

> 
> There is no settings changed.
> 
>  Aastra 57i phone is registered to the freeswitch.
> 
> Aastra 9133i no more registered to freeswitch.
> 
> 
> Is there any major change between 17155 & 17188
> 
> Thanks
> Lloyd
> 



From lloyd.aloysius at gmail.com  Sun Apr  4 11:57:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:57:29 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
Message-ID: <z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>

There is no firmware update on Aastra. Only change in FreeSWITCH.


Snom 320 throwing the following Error.

2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
refer to rfc3711





On Sun, Apr 4, 2010 at 2:49 PM, Brian West <brian at freeswitch.org> wrote:

> Its an aastra... I don't trust it.
>
> /b
>
> On Apr 4, 2010, at 1:45 PM, Aloysius Lloyd wrote:
>
> >
> > There is no settings changed.
> >
> >  Aastra 57i phone is registered to the freeswitch.
> >
> > Aastra 9133i no more registered to freeswitch.
> >
> >
> > Is there any major change between 17155 & 17188
> >
> > Thanks
> > Lloyd
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 12:05:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 14:05:03 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
Message-ID: <32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>

That snom error means you don't have the SRTP set to optional... because a=crypto in an RTP/AVP is invalid.

Again I don't trust the aastra's cuz I have personally seen them fuck up in production with no changes to FreeSWITCH.  Check the invites... mine started sending 0.0.0.0 in the sdp on initial invites.

/b

On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:

> There is no firmware update on Aastra. Only change in FreeSWITCH.
> 
> 
> Snom 320 throwing the following Error.
> 
> 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP, refer to rfc3711



From lloyd.aloysius at gmail.com  Sun Apr  4 12:15:07 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 15:15:07 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
Message-ID: <y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>

Thank you for the information.

I have deployed hundreds of Aastra Phones so far no major problems and
complains. There are some firmware issues in earlier versions.

In my scenario FreeSWITCH 17155 working and 17188 is not working.

Thanks
Lloyd

On Sun, Apr 4, 2010 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:

> That snom error means you don't have the SRTP set to optional... because
> a=crypto in an RTP/AVP is invalid.
>
> Again I don't trust the aastra's cuz I have personally seen them fuck up in
> production with no changes to FreeSWITCH.  Check the invites... mine started
> sending 0.0.0.0 in the sdp on initial invites.
>
> /b
>
> On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:
>
> > There is no firmware update on Aastra. Only change in FreeSWITCH.
> >
> >
> > Snom 320 throwing the following Error.
> >
> > 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
> refer to rfc3711
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Sun Apr  4 12:19:44 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 15:19:44 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
Message-ID: <l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>

I just notice in the console log.

2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl
"domains". Falling back to Digest auth.
2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl
"domains". Falling back to Digest auth.

Snom and Aastra - I have the same behavior.

But Xlite - Not giving the same message and working.

Thanks
Lloyd





On Sun, Apr 4, 2010 at 3:15 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thank you for the information.
>
> I have deployed hundreds of Aastra Phones so far no major problems and
> complains. There are some firmware issues in earlier versions.
>
> In my scenario FreeSWITCH 17155 working and 17188 is not working.
>
> Thanks
> Lloyd
>
>
> On Sun, Apr 4, 2010 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> That snom error means you don't have the SRTP set to optional... because
>> a=crypto in an RTP/AVP is invalid.
>>
>> Again I don't trust the aastra's cuz I have personally seen them fuck up
>> in production with no changes to FreeSWITCH.  Check the invites... mine
>> started sending 0.0.0.0 in the sdp on initial invites.
>>
>> /b
>>
>> On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:
>>
>> > There is no firmware update on Aastra. Only change in FreeSWITCH.
>> >
>> >
>> > Snom 320 throwing the following Error.
>> >
>> > 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
>> refer to rfc3711
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Apr  4 12:24:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 14:24:34 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
Message-ID: <1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>

I'm going to not be able to tell you unless you show me the register and reply to register.. please turn on the sip trace and give me that and I can see what is going on.

/b

On Apr 4, 2010, at 2:19 PM, Aloysius Lloyd wrote:

> I just notice in the console log.
> 
> 2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl "domains". Falling back to Digest auth.
> 2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl "domains". Falling back to Digest auth.
> 
> Snom and Aastra - I have the same behavior.
> 
> But Xlite - Not giving the same message and working.
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Sun Apr  4 13:05:26 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 16:05:26 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
Message-ID: <t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>

Brian,

Could you please guide me , how to capture the SIP Trace.

1. One Phone Register
2. Other One Not Register
3. Both cannot make any calls.
4.  Multi Tenant Environment.

Thanks
Lloyd


On Sun, Apr 4, 2010 at 3:24 PM, Brian West <brian at freeswitch.org> wrote:

> I'm going to not be able to tell you unless you show me the register and
> reply to register.. please turn on the sip trace and give me that and I can
> see what is going on.
>
> /b
>
> On Apr 4, 2010, at 2:19 PM, Aloysius Lloyd wrote:
>
> > I just notice in the console log.
> >
> > 2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by
> acl "domains". Falling back to Digest auth.
> > 2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by
> acl "domains". Falling back to Digest auth.
> >
> > Snom and Aastra - I have the same behavior.
> >
> > But Xlite - Not giving the same message and working.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 13:10:42 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:10:42 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
Message-ID: <69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>

sofia profile xxxx siptrace on

/b
PS:xxxx replace with your profilename 

On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:

> Brian,
> 
> Could you please guide me , how to capture the SIP Trace. 
> 
> 1. One Phone Register
> 2. Other One Not Register
> 3. Both cannot make any calls.
> 4.  Multi Tenant Environment.
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Sun Apr  4 13:27:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 16:27:29 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
Message-ID: <w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>

Please find the following trace.


1. Snom 320 - Xlite Call Failed

http://pastebin.freeswitch.org/12624

<http://pastebin.freeswitch.org/12624>2. Aastra 57i - Dial 9999

http://pastebin.freeswitch.org/12626

3. Phone Not Registering.

http://pastebin.freeswitch.org/12627


On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:

> sofia profile xxxx siptrace on
>
> /b
> PS:xxxx replace with your profilename
>
> On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
>
> > Brian,
> >
> > Could you please guide me , how to capture the SIP Trace.
> >
> > 1. One Phone Register
> > 2. Other One Not Register
> > 3. Both cannot make any calls.
> > 4.  Multi Tenant Environment.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 13:35:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:35:14 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
Message-ID: <B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>


On Apr 4, 2010, at 3:27 PM, Aloysius Lloyd wrote:

> Please find the following trace.
> 
> 
> 1. Snom 320 - Xlite Call Failed
> 
> http://pastebin.freeswitch.org/12624

INCOMPATIBLE_DESTINATION (turn SRTP off or set it to optional for the SAVP option.)

Fix the option I told you about already.  This is listed in the FAQ on the wiki already.

> 
> 2. Aastra 57i - Dial 9999
> 
> http://pastebin.freeswitch.org/12626

Your phone isn't even getting autenticated sounds like you have put some ACL's in place to allow it in without auth.

201->9999 in context public


> 
> 3. Phone Not Registering.
> 
> http://pastebin.freeswitch.org/12627

Enable rport if you notice the register comes from port 1026 and we challenge to 5060 because thats what the phone put in the contact field.  Enable rport on the phone will fix this.

/b



> 
> 
> On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:
> sofia profile xxxx siptrace on
> 
> /b
> PS:xxxx replace with your profilename
> 
> On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
> 
> > Brian,
> >
> > Could you please guide me , how to capture the SIP Trace.
> >
> > 1. One Phone Register
> > 2. Other One Not Register
> > 3. Both cannot make any calls.
> > 4.  Multi Tenant Environment.
> >
> > Thanks
> > Lloyd
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr  4 13:37:40 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:37:40 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
Message-ID: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>

Anyone else get spam from them?

/b



From lloyd.aloysius at gmail.com  Sun Apr  4 14:11:59 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 17:11:59 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
Message-ID: <z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>

Thanks Brian.

This is simple direct install and no ACL. I am going to Install From
Scratch.

Thanks
Lloyd

On Sun, Apr 4, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Apr 4, 2010, at 3:27 PM, Aloysius Lloyd wrote:
>
> > Please find the following trace.
> >
> >
> > 1. Snom 320 - Xlite Call Failed
> >
> > http://pastebin.freeswitch.org/12624
>
> INCOMPATIBLE_DESTINATION (turn SRTP off or set it to optional for the SAVP
> option.)
>
> Fix the option I told you about already.  This is listed in the FAQ on the
> wiki already.
>
> >
> > 2. Aastra 57i - Dial 9999
> >
> > http://pastebin.freeswitch.org/12626
>
> Your phone isn't even getting autenticated sounds like you have put some
> ACL's in place to allow it in without auth.
>
> 201->9999 in context public
>
>
> >
> > 3. Phone Not Registering.
> >
> > http://pastebin.freeswitch.org/12627
>
> Enable rport if you notice the register comes from port 1026 and we
> challenge to 5060 because thats what the phone put in the contact field.
>  Enable rport on the phone will fix this.
>
> /b
>
>
>
> >
> >
> > On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:
> > sofia profile xxxx siptrace on
> >
> > /b
> > PS:xxxx replace with your profilename
> >
> > On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
> >
> > > Brian,
> > >
> > > Could you please guide me , how to capture the SIP Trace.
> > >
> > > 1. One Phone Register
> > > 2. Other One Not Register
> > > 3. Both cannot make any calls.
> > > 4.  Multi Tenant Environment.
> > >
> > > Thanks
> > > Lloyd
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Sun Apr  4 14:13:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 17:13:24 -0400
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
Message-ID: <q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>

I recived one - Sun, Apr 4, 2010 at 4:26 PM

Lloyd

On Sun, Apr 4, 2010 at 4:37 PM, Brian West <brian at freeswitch.org> wrote:

> Anyone else get spam from them?
>
> /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
>
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From brian at freeswitch.org  Sun Apr  4 14:25:46 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 16:25:46 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
Message-ID: <E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>

be sure to nuke your configs... cuz we won't when you reinstall.

/b

On Apr 4, 2010, at 4:11 PM, Aloysius Lloyd wrote:

> Thanks Brian.
> 
> This is simple direct install and no ACL. I am going to Install From Scratch.
> 
> Thanks
> Lloyd



From brian at freeswitch.org  Sun Apr  4 14:28:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 16:28:12 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>
Message-ID: <3D15397A-8F23-4585-8EE6-801AAD427022@freeswitch.org>

Ok that confirms they are spammers and should be SHOT!

/b

On Apr 4, 2010, at 4:13 PM, Aloysius Lloyd wrote:

> I recived one - Sun, Apr 4, 2010 at 4:26 PM
> 
> Lloyd
> 
> On Sun, Apr 4, 2010 at 4:37 PM, Brian West <brian at freeswitch.org> wrote:
> Anyone else get spam from them?
> 
> /b

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From max.clark at gmail.com  Sun Apr  4 15:00:34 2010
From: max.clark at gmail.com (Max Clark)
Date: Sun, 4 Apr 2010 15:00:34 -0700
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
Message-ID: <p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>

How would one detect T38 and convert the session into proxy media?

On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone at gmail.com> wrote:
> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> You just need the regular passthrough module: mod_g729 and to allow G729 as
> inbound and outbound codecs in vars.xml.
> To summarize:
> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> path, it relays the audio stream between endpoints, but can still detect
> DTMFs
> -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF
> detection is impossible. In this mode, FS is really a "dumb" transparent
> RTP-forwarder (this is required to get T38 working between the 2 endpoints)
> -bypass media enabled: FreeSWITCH is not in the RTP path
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>
> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From linux4michelle at tamay-dogan.net  Sun Apr  4 15:16:47 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Mon, 5 Apr 2010 00:16:47 +0200
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
Message-ID: <20100404221647.GN3737@tamay-dogan.net>

Hello Brian West,

Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
> Anyone else get spam from them?

YesNo because my server/spamassassin has rejected it.

Missing spamfilter?

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    24V Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strasbourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From brian at freeswitch.org  Sun Apr  4 15:23:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 17:23:07 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <20100404221647.GN3737@tamay-dogan.net>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<20100404221647.GN3737@tamay-dogan.net>
Message-ID: <B2D535ED-E3BB-4846-A0DB-557C4E38D78B@freeswitch.org>

I shouldn't have to deal with it.  They should be responsible or DIE.  It slipped thru... but it should be known they are probably a shady operation and shouldn't be trusted.

/b

On Apr 4, 2010, at 5:16 PM, Michelle Konzack wrote:

> Hello Brian West,
> 
> Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
>> Anyone else get spam from them?
> 
> YesNo because my server/spamassassin has rejected it.
> 
> Missing spamfilter?
> 
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    24V Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant



From brian at freeswitch.org  Sun Apr  4 20:07:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 22:07:05 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <20100404221647.GN3737@tamay-dogan.net>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<20100404221647.GN3737@tamay-dogan.net>
Message-ID: <256C90FA-AE87-4D2D-9FA8-71DD9434EB89@freeswitch.org>

Well it seems they harvested our mailing list... so every please be on the look out for it.

/b

On Apr 4, 2010, at 5:16 PM, Michelle Konzack wrote:

> Hello Brian West,
> 
> Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
>> Anyone else get spam from them?
> 
> YesNo because my server/spamassassin has rejected it.
> 
> Missing spamfilter?



From msc at freeswitch.org  Sun Apr  4 20:11:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 4 Apr 2010 20:11:04 -0700
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <4BB87CA9.3050403@k4gvo.com>
References: <4BB79458.8080601@k4gvo.com>
	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
	<4BB87CA9.3050403@k4gvo.com>
Message-ID: <h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>

I looked in mod_openzap.c and I didn't see any references to channel
variables. However, you have context and dialplan options. I suggest that
you create a dialplan context just for your FXS port(s). Try this. Create
conf/dialplan/fxs-ports.xml:

<include>
  <context name="fxs-ports">
    <extension name="outbound">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set"
data="toll_allow=local,domestic,international"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>
  </context>
</include>

Then in your openzap.conf.xml change the context for the analog span(s) with
the FXS ports:
<param name="context" value="default"/>

Restart FS after making these changes and then give it a shot. You should
see the call from the analog phone going into context "fxs-ports" and then
get transferred over to the default context where it will act like your SIP
phones because we manually set the ${toll_allow} chan var.

-MC

On Sun, Apr 4, 2010 at 4:48 AM, Jim <jim at k4gvo.com> wrote:

> Michael Collins wrote:
> > Variables set in the directory/default/xxxx.xml files apply to users
> > who make authenticated calls through FS. Generally those will be SIP
> > phones.
> >
> > Let's back up a step. What problem are you trying to solve, i.e., why
> > is it that you need to set the toll_allow variable? What endpoint is
> > making an openzap call?
> >
> > -MC
> Hi, Michael,
>
> I have multiple sip phone that are working fine.  When I dial a 10 digit
> number they connect with my sip provider and place the call.  The
> openzap configured phone gets dial tone and can call other extensions,
> however when I dial a 10 digit number it gives me a busy.  In the log I
> see it appears to be failing on the toll_allow test:
>
> Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call]
> continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call]
> ${toll_allow}() =~ /local/ break=on-false
> Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call]
> continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call]
> ${toll_allow}() =~ /domestic/ break=on-false
> Dialplan: OpenZAP/1:1/7707190068 parsing
> [default->international.example.com] continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL)
> [international.example.com] ${toll_allow}() =~ /international/
> break=on-false
>
> This is the area when it should be placing the call, I belive.  When
> placing a call from a sip phone I see:
>
> Dialplan: sofia/internal/1003 at 192.168.2.51 parsing [default->local_call]
> continue=false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call]
> ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call]
> destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(effective_caller_id_number=${outbound_caller_id_number})
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(effective_caller_id_name=${outbound_caller_id_name})
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> bridge(sofia/gateway/${default_gateway}/7707190068)
>
> I simply want this extension to be able to dial out.  The configuration
> is 99% default.
>
> Thanks,
> Jim.
> >
> > On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com
> > <mailto:jim at k4gvo.com>> wrote:
> >
> >     I obviously need to set a <variable name="toll_allow"
> >     value="domestic,international,local"/> somewhere but I can't
> >     figure out
> >     what file to put it in.  The examples all show it in the
> >     directory/default/xxxx.xml files but those appear to be sip only.  In
> >     any event creating files in that directory for my extension did
> >     nothing
> >     to help the problem.
> >
> >     The only places I have the extension mentioned is in  the
> openzap.conf
> >     file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
> >     element to the latter does nothing.
> >
> >     How do I set that variable?  Or where?
> >
> >     Thanks,
> >     Jim.
> >
> >     _______________________________________________
> >     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
>
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From msc at freeswitch.org  Sun Apr  4 20:16:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 4 Apr 2010 20:16:05 -0700
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100404090838.414c58ce@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
Message-ID: <h2v87f2f3b91004042016wc562e5fds46911c2d9bf43652@mail.gmail.com>

On Sun, Apr 4, 2010 at 12:08 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> FS svn-17188M
> Debian Lenny
> =================
>
> Hi list,
>
> I've installed cepstral voices (in /opt/swift, and also changed its
> rights recursively to freeswitch:freeswitch) but I had a complain
> at FS start that some cepstral libraries were missing.
>
> Eventually, I was obliged to simlink /opt/swift/lib/lib* into
> /usr/local/freeswitch/lib to have it working correctly, is it normal,
> or did I miss something?
>
> That's "normal" even though it's not desired. Getting Cepstral working
properly again would be nice but I don't think it's high on the priority
list.
-MC


> (I followed the wiki, so all libs were added to ld.so.cache, path in
> mod_cepstral source Makefile are good, SWIFT_HOME is defined, module
> flite loading is commented, etc- so I said to myself: Nantidiou!)
>
> --
> Sorry never means having your say to love.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 20:19:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 22:19:48 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100404090838.414c58ce@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
Message-ID: <009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>

Their exists a file in /etc/ called ld.so.conf  if you happen to add  /opt/swift/lib/lib into that then run ldconfig it would work.

muhahahahaha linux is great.

/b

On Apr 4, 2010, at 2:08 AM, Jean-Yves F. Barbier wrote:

> 
> Eventually, I was obliged to simlink /opt/swift/lib/lib* into 
> /usr/local/freeswitch/lib to have it working correctly, is it normal, 
> or did I miss something?



From hungngm at bkav.com.vn  Sun Apr  4 20:27:26 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Mon, 05 Apr 2010 10:27:26 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>



 

Hi Anthony,

Can you discuss some details in how polycom or snom can do this and x-lite not.

If can, I want to edit some open source soffphone like officeSIP to do this.

Best Regards.

 

Anthony Minessale [anthony.minessale at gmail.com]

 

We already do it.
X-Lite does not support it.
If you try it with a phone like snom or polycom you will see it works just like
that.


 

2010/4/3 Nguy??n M???nh H??ng < hungngm at bkav.com.vn >
 

     

    Hi Seven Du.

    

    Thanks to yours suggetion.

    

    I have an ideal, it is: when the call between caller and agent is set, the
    caller_id is determined. So, i want to edit code to sent the agent
    information (the call_id and call_id_number) which will be displayed againt
    in the agent's softphone (as Xlite..) when the call is happening.

    

    I read some documents but i still can't determine: It's maybe yes or maybe
    to do this and where to do this.

    

    Can you give me some comments.

    

    Best Regard.

    

     

    Seven Du [ dujinfang at gmail.com ]

    

    

    ??As discussed in the list, it's not a freeswitch problem but a reality
    of life. 



    Think about customer A and B calls in one after another, then if
    FreeSWITCH call agent X with caller id A and Y with caller id B, and
    angent Y answers before X, then

    1) if bridge Y with A with the FIFO rule, then the caller id is wrong
    2) if bridge Y with B, the caller id is right but it breaks the rule
    of FIFO - A should be served before B!! And what even worse is that
    if X never answer A then A never can be served which is really
    unfair!!

    Of course you don't want 1), and you don't need mod_fifo if you want
    behavior 2), you just need some dialplan trick or some simple Lua
    script I think. Also FreeSWITCH is designed to be easily extended with
    almost any languages so feel free to implement anything.

    2010/3/31 Nguy??n M???nh H??ng : 


    > Hi Mike and Seven Du.
    > Thanks to yours help.
    > I known the mechanism of mod_fifo.
    >>>
   
   
   
   
    http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
    .
    > What a pity, It can't solve this problem. I can't use freeswitch for my
    call
    > center.
    > Hope new version can solve 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


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




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

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From anthony.minessale at gmail.com  Sun Apr  4 21:41:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 4 Apr 2010 23:41:56 -0500
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>
References: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>
Message-ID: <i2w191c3a031004042141g11c18e92wcd63e833fe1f3aa0@mail.gmail.com>

its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when
the call is bridged.
X-lite does not update anything when it receives them.

That's about it.


2010/4/4 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi Anthony,
>
> Can you discuss some details in how polycom or snom can do this and x-lite
> not.
>
> If can, I want to edit some open source soffphone like officeSIP to do
> this.
>
> Best Regards.
>
> Anthony Minessale <anthony.minessale at gmail.com> [
> anthony.minessale at gmail.com]
>
>
> We already do it.
> X-Lite does not support it.
> If you try it with a phone like snom or polycom you will see it works just
> like that.
>
>
> 2010/4/3 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>
>
>>  Hi Seven Du.
>>
>> Thanks to yours suggetion.
>>
>> I have an ideal, it is: when the call between caller and agent is set, the
>> caller_id is determined. So, i want to edit code to sent the agent
>> information (the call_id and  call_id_number) which will be displayed
>> againt in the agent's softphone (as Xlite..) when the call is happening.
>>
>> I read some documents but i still can't determine: It's maybe yes or maybe
>> to do this and where to do this.
>>
>> Can you give me some comments.
>>
>> Best Regard.
>>
>> Seven Du <dujinfang at gmail.com> [dujinfang at gmail.com]
>>
>>
>> ?As discussed in the list, it's not a freeswitch problem but a reality of
>> life.
>>
>>
>> Think about customer A and B calls in one after another, then if
>> FreeSWITCH call agent X with caller id A and Y with caller id B, and
>> angent Y answers before X, then
>>
>> 1) if bridge Y with A with the FIFO rule, then the caller id is wrong
>> 2) if bridge Y with B, the caller id is right but it breaks the rule
>> of FIFO - A should be served before B!! And what even worse is that
>> if X never answer A then A never can be served which is really
>> unfair!!
>>
>> Of course you don't want 1), and you don't need mod_fifo if you want
>> behavior 2), you just need some dialplan trick or some simple Lua
>> script I think. Also FreeSWITCH is designed to be easily extended with
>> almost any languages so feel free to implement anything.
>>
>> 2010/3/31 Nguy?n M?nh H?ng :
>>
>> > Hi Mike and Seven Du.
>> > Thanks to yours help.
>> > I known the mechanism of mod_fifo.
>> >>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
>> .
>> > What a pity, It can't solve this problem. I can't use freeswitch for my
>> call
>> > center.
>> > Hope new version can solve 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jayesh.voip at gmail.com  Sun Apr  4 22:43:26 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Mon, 5 Apr 2010 11:13:26 +0530
Subject: [Freeswitch-users] domain-wise context
Message-ID: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>

Hi Brian,
I've tried user_context variable. Does user_context apply to only registered
users in freeswitch?
Because in my case the users are registered on a different software(OSips).
Also out of curiosity, does the OSips needs to be defined as a Gateway in FS
for handling the calls in a standard fashion?

Thanks,

--- Jayesh


> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 3 Apr 2010 22:57:19 -0500
> Subject: Re: [Freeswitch-users] domain-wise context
> see user_context variable on the user.
>
> /b
>
> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>
> > Hi,
> > Sorry for sounding so impatient, the anxiety only grew because before
> posting it to the list I spent a week on reading all the documentation
> available on and around this topic FS site and mailing lists. I really
> appreciate and am thankful for your suggestions.
> > I'll try out the suggestions given by you. Is there a way that we can
> compare the domain name in the dialplan using regular expressions. Is there
> a value in condition tag that can be used to compare this, something like
> "extension_number".
> > Or should I store the domain value in some variable(something like
> sip_h_<domain_name>) and compare that variable to take the call to a
> different context?
> >
> > Thanks,
> >
> > --- Jayesh
>
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From david.ponzone at gmail.com  Mon Apr  5 01:11:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 10:11:09 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>
Message-ID: <A90BAE74-B676-40EA-8B4C-D18509059905@gmail.com>

As far as I know, you can't, because T38 is not advertised at first.
T38 starts with regular codecs in the SDP, and then later, a T38  
REINVITE is negotiated.

I guess the easiest way to handle that is:
-enable proxy media based on DID for inbound fax
-enable proxy media based on CLID for outbound fax

This requires to have specific extensions in your dialplan.

For outbound, you can also have a dedicated SIP profile for T38 ATAs  
so you can enable proxy-media for the whole profile.
For inbound, you may do the same if your ITSP/gateway can send you the  
fax DIDs on a specific trunk (so to a specific SIP profile).

Be aware that T38 is a PITA, and that you really need to validate that  
your T38 device is compatible with the other endpoint, which is  
probably a gateway.
If this gateway is yours, that's fine because you control it, and you  
can rely on it for your T38 service.
If it's not yours, you could have issues if some day, your iTSP  
decices to change it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 00:00, Max Clark a ?crit :

> How would one detect T38 and convert the session into proxy media?
>
> On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> You just need the regular passthrough module: mod_g729 and to allow  
>> G729 as
>> inbound and outbound codecs in vars.xml.
>> To summarize:
>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
>> the RTP
>> path, it relays the audio stream between endpoints, but can still  
>> detect
>> DTMFs
>> -proxy media enabled: FreeSWITCH relays the audio stream  
>> transparently, DTMF
>> detection is impossible. In this mode, FS is really a "dumb"  
>> transparent
>> RTP-forwarder (this is required to get T38 working between the 2  
>> endpoints)
>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>
>> Clarification - for G729 does freeswitch need to be in "bypass media"
>> or "proxy media"? My understanding was that G729 would work with
>> "proxy media" enabled and without the new fangled module?
>>
>> -Max
>>
>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
>> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at googlemail.com  Mon Apr  5 02:49:13 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 10:49:13 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
Message-ID: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>

Have you reviewed http://pastebin.freeswitch.org/12617 ?
It has the G729 set in the codecs section.
In this one it seems the call does not get to the external gateway.
Freeswitch stops the call before calling the external gateway


I have checked it again a few times using the bypass_media,
proxy_media settings.

And with those settings the call ends as soon as ringing starts or as
sonn as the call is answered.

I will do another one just to confirm




On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> You just need the regular passthrough module: mod_g729 and to allow G729 as
> inbound and outbound codecs in vars.xml.
> To summarize:
> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> path, it relays the audio stream between endpoints, but can still detect
> DTMFs
> -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF
> detection is impossible. In this mode, FS is really a "dumb" transparent
> RTP-forwarder (this is required to get T38 working between the 2 endpoints)
> -bypass media enabled: FreeSWITCH is not in the RTP path
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>
> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From david.ponzone at gmail.com  Mon Apr  5 03:02:27 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 12:02:27 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
Message-ID: <FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>

Frank,

again:

     
------------------------------------------------------------------------
recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
     
------------------------------------------------------------------------
    INVITE sip:02074379497 at 192.168.4.156 SIP/2.0
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Remote-Party-ID: Booth1 <sip: 
1001 at 192.168.4.156>;screen=yes;party=calling
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    Max-Forwards: 70
    Proxy-Authorization: Digest  
username="1001",realm="192.168.4.156",nonce="9567
e05b-d541-4a5e-9e47-2152eb90a199",uri="sip: 
02074379497 at 192.168.4.156",algorithm=
MD5 
,response 
="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d
a1954"
    Contact: Booth1 <sip:1001 at 192.168.4.154:5060>
    Expires: 240
    User-Agent: Linksys/SPA2102-3.2.8(d)
    Content-Length: 260
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp

    v=0
    o=- 2612067 2612067 IN IP4 192.168.4.154
    s=-
    c=IN IP4 192.168.4.154
    t=0 0
    m=audio 16392 RTP/AVP 18 100 101
    a=rtpmap:18 G729a/8000
    a=rtpmap:100 NSE/8000
    a=fmtp:100 192-193
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv

     
------------------------------------------------------------------------
send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
     
------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M
    Content-Length: 0

     
------------------------------------------------------------------------
2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001 at 192.168.4.156 
  [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8]
2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001 at 192.168.1.133 
  [CS_NEW] [INCOMPATIBLE_DESTINATION]

send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
     
------------------------------------------------------------------------
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
    To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
    Call-ID: 1c27844d-e299e5e9 at 192.168.4.154

That is the answer from FS to the phone, just after receiving the  
INVITE that contains only G729 and NSE (??) in the SDP.

If you're sure you enabled G729 in vars.xml, you should check you  
restarted your SIP profile.
If you are not sure how to do it right, check my previous emails, or  
restart FS .

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 11:49, Frank Church a ?crit :

> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> You just need the regular passthrough module: mod_g729 and to allow  
>> G729 as
>> inbound and outbound codecs in vars.xml.
>> To summarize:
>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
>> the RTP
>> path, it relays the audio stream between endpoints, but can still  
>> detect
>> DTMFs
>> -proxy media enabled: FreeSWITCH relays the audio stream  
>> transparently, DTMF
>> detection is impossible. In this mode, FS is really a "dumb"  
>> transparent
>> RTP-forwarder (this is required to get T38 working between the 2  
>> endpoints)
>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>
>> Clarification - for G729 does freeswitch need to be in "bypass media"
>> or "proxy media"? My understanding was that G729 would work with
>> "proxy media" enabled and without the new fangled module?
>>
>> -Max
>>
>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
>> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 oseslija at gmail.com  Mon Apr  5 03:34:56 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 12:34:56 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
Message-ID: <n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>

G729a is invalid as a codec name. FS used to allow it but not anymore afaik.
You should change codec name to G729 (I assume you're using Linksys product;
there is a setting to change under SIP tab).

Ognjen

On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists at googlemail.com>wrote:

> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > You just need the regular passthrough module: mod_g729 and to allow G729
> as
> > inbound and outbound codecs in vars.xml.
> > To summarize:
> > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> > path, it relays the audio stream between endpoints, but can still detect
> > DTMFs
> > -proxy media enabled: FreeSWITCH relays the audio stream transparently,
> DTMF
> > detection is impossible. In this mode, FS is really a "dumb" transparent
> > RTP-forwarder (this is required to get T38 working between the 2
> endpoints)
> > -bypass media enabled: FreeSWITCH is not in the RTP path
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >
> > Clarification - for G729 does freeswitch need to be in "bypass media"
> > or "proxy media"? My understanding was that G729 would work with
> > "proxy media" enabled and without the new fangled module?
> >
> > -Max
> >
> > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
> >
> > Frank Church <vfclists at googlemail.com> wrote:
> >
> > I am just trialling Freeswitch with Linksys adapters, whose default
> >
> > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >
> > When I change that setting to 'yes' the calls don't go through. I am
> >
> > using the latest Windows SVN.
> >
> > FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass
> media
> >
> > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> > really
> >
> > need it.
> >
> > My recommendation would be to use a codec other than G.729 unless you
> have a
> >
> > compelling reason, for example a carrier that only supports G.729.
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 david.ponzone at gmail.com  Mon Apr  5 03:46:38 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 12:46:38 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
Message-ID: <31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>

Ognjen,

very good point, but I used to think that for G729 (and all payload id  
smaller than 97), FS was relying on the payload id, and not the name.

Am I wrong ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :

> G729a is invalid as a codec name. FS used to allow it but not  
> anymore afaik.
> You should change codec name to G729 (I assume you're using Linksys  
> product; there is a setting to change under SIP tab).
>
> Ognjen
>
> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church  
> <vfclists at googlemail.com> wrote:
> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > You just need the regular passthrough module: mod_g729 and to  
> allow G729 as
> > inbound and outbound codecs in vars.xml.
> > To summarize:
> > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
> the RTP
> > path, it relays the audio stream between endpoints, but can still  
> detect
> > DTMFs
> > -proxy media enabled: FreeSWITCH relays the audio stream  
> transparently, DTMF
> > detection is impossible. In this mode, FS is really a "dumb"  
> transparent
> > RTP-forwarder (this is required to get T38 working between the 2  
> endpoints)
> > -bypass media enabled: FreeSWITCH is not in the RTP path
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou  
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est  
> susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
> message s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
> de ce
> > message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
> >
> >
> >
> > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >
> > Clarification - for G729 does freeswitch need to be in "bypass  
> media"
> > or "proxy media"? My understanding was that G729 would work with
> > "proxy media" enabled and without the new fangled module?
> >
> > -Max
> >
> > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
> wrote:
> >
> > Frank Church <vfclists at googlemail.com> wrote:
> >
> > I am just trialling Freeswitch with Linksys adapters, whose default
> >
> > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >
> > When I change that setting to 'yes' the calls don't go through. I am
> >
> > using the latest Windows SVN.
> >
> > FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
> bypass media
> >
> > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
> you
> > really
> >
> > need it.
> >
> > My recommendation would be to use a codec other than G.729 unless  
> you have a
> >
> > compelling reason, for example a carrier that only supports G.729.
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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

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From sos at sokhapkin.dyndns.org  Mon Apr  5 03:51:50 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 5 Apr 2010 06:51:50 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
	<FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>
Message-ID: <201004050651.50465.sos@sokhapkin.dyndns.org>

G729a is incorrect codec name, it must be changed to G729 in SPA settings.

On Monday 05 April 2010, David Ponzone wrote:
> Frank,
> 
> again:
> 
> 
> ------------------------------------------------------------------------
> recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
> 
> ------------------------------------------------------------------------
>     INVITE sip:02074379497 at 192.168.4.156 SIP/2.0
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
>     To: <sip:02074379497 at 192.168.4.156>
>     Remote-Party-ID: Booth1 <sip:
> 1001 at 192.168.4.156>;screen=yes;party=calling
>     Call-ID: 68993a3f-e38f80af at 192.168.4.154
>     CSeq: 102 INVITE
>     Max-Forwards: 70
>     Proxy-Authorization: Digest
> username="1001",realm="192.168.4.156",nonce="9567
> e05b-d541-4a5e-9e47-2152eb90a199",uri="sip:
> 02074379497 at 192.168.4.156",algorithm=
> MD5
> ,response
> ="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d
> a1954"
>     Contact: Booth1 <sip:1001 at 192.168.4.154:5060>
>     Expires: 240
>     User-Agent: Linksys/SPA2102-3.2.8(d)
>     Content-Length: 260
>     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>     Supported: x-sipura
>     Content-Type: application/sdp
> 
>     v=0
>     o=- 2612067 2612067 IN IP4 192.168.4.154
>     s=-
>     c=IN IP4 192.168.4.154
>     t=0 0
>     m=audio 16392 RTP/AVP 18 100 101
>     a=rtpmap:18 G729a/8000
>     a=rtpmap:100 NSE/8000
>     a=fmtp:100 192-193
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=ptime:30
>     a=sendrecv
> 
> 
> ------------------------------------------------------------------------
> send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
> 
> ------------------------------------------------------------------------
>     SIP/2.0 100 Trying
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
>     To: <sip:02074379497 at 192.168.4.156>
>     Call-ID: 68993a3f-e38f80af at 192.168.4.154
>     CSeq: 102 INVITE
>     User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M
>     Content-Length: 0
> 
> 
> ------------------------------------------------------------------------
> 2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel
>  sofia/internal/1001 at 192.168.4.156 [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8]
> 2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup
>  sofia/internal/1001 at 192.168.1.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 
> send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
> 
> ------------------------------------------------------------------------
>     SIP/2.0 488 Not Acceptable Here
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
>     To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
>     Call-ID: 1c27844d-e299e5e9 at 192.168.4.154
> 
> That is the answer from FS to the phone, just after receiving the
> INVITE that contains only G729 and NSE (??) in the SDP.
> 
> If you're sure you enabled G729 in vars.xml, you should check you
> restarted your SIP profile.
> If you are not sure how to do it right, check my previous emails, or
> restart FS .
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> Le 05/04/2010 ? 11:49, Frank Church a ?crit :
> > Have you reviewed http://pastebin.freeswitch.org/12617 ?
> > It has the G729 set in the codecs section.
> > In this one it seems the call does not get to the external gateway.
> > Freeswitch stops the call before calling the external gateway
> >
> >
> > I have checked it again a few times using the bypass_media,
> > proxy_media settings.
> >
> > And with those settings the call ends as soon as ringing starts or as
> > sonn as the call is answered.
> >
> > I will do another one just to confirm
> >
> > On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> >> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> >> You just need the regular passthrough module: mod_g729 and to allow
> >> G729 as
> >> inbound and outbound codecs in vars.xml.
> >> To summarize:
> >> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
> >> the RTP
> >> path, it relays the audio stream between endpoints, but can still
> >> detect
> >> DTMFs
> >> -proxy media enabled: FreeSWITCH relays the audio stream
> >> transparently, DTMF
> >> detection is impossible. In this mode, FS is really a "dumb"
> >> transparent
> >> RTP-forwarder (this is required to get T38 working between the 2
> >> endpoints)
> >> -bypass media enabled: FreeSWITCH is not in the RTP path
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et
> >> ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> >> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est
> >> susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
> >> message s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
> >> de ce
> >> message, merci de le d?truire imm?diatement et d'avertir
> >> l'exp?diteur.
> >>
> >>
> >>
> >> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >>
> >> Clarification - for G729 does freeswitch need to be in "bypass media"
> >> or "proxy media"? My understanding was that G729 would work with
> >> "proxy media" enabled and without the new fangled module?
> >>
> >> -Max
> >>
> >> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >>
> >> Frank Church <vfclists at googlemail.com> wrote:
> >>
> >> I am just trialling Freeswitch with Linksys adapters, whose default
> >>
> >> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >>
> >> When I change that setting to 'yes' the calls don't go through. I am
> >>
> >> using the latest Windows SVN.
> >>
> >> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
> >> bypass media
> >>
> >> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
> >> you
> >> really
> >>
> >> need it.
> >>
> >> My recommendation would be to use a codec other than G.729 unless
> >> you have a
> >>
> >> compelling reason, for example a carrier that only supports G.729.
> >>
> >>
> >> _______________________________________________
> >>
> >> FreeSWITCH-users mailing list
> >>
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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://devblog.brahmancreations.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 sos at sokhapkin.dyndns.org  Mon Apr  5 03:54:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 5 Apr 2010 06:54:11 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
Message-ID: <201004050654.11705.sos@sokhapkin.dyndns.org>

FS looks at codec name too.

On Monday 05 April 2010, David Ponzone wrote:
> Ognjen,
> 
> very good point, but I used to think that for G729 (and all payload id
> smaller than 97), FS was relying on the payload id, and not the name.
> 
> Am I wrong ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
> > G729a is invalid as a codec name. FS used to allow it but not
> > anymore afaik.
> > You should change codec name to G729 (I assume you're using Linksys
> > product; there is a setting to change under SIP tab).
> >
> > Ognjen
> >
> > On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
> > <vfclists at googlemail.com> wrote:
> > Have you reviewed http://pastebin.freeswitch.org/12617 ?
> > It has the G729 set in the codecs section.
> > In this one it seems the call does not get to the external gateway.
> > Freeswitch stops the call before calling the external gateway
> >
> >
> > I have checked it again a few times using the bypass_media,
> > proxy_media settings.
> >
> > And with those settings the call ends as soon as ringing starts or as
> > sonn as the call is answered.
> >
> > I will do another one just to confirm
> >
> > On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > > You just need the regular passthrough module: mod_g729 and to
> >
> > allow G729 as
> >
> > > inbound and outbound codecs in vars.xml.
> > > To summarize:
> > > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
> >
> > the RTP
> >
> > > path, it relays the audio stream between endpoints, but can still
> >
> > detect
> >
> > > DTMFs
> > > -proxy media enabled: FreeSWITCH relays the audio stream
> >
> > transparently, DTMF
> >
> > > detection is impossible. In this mode, FS is really a "dumb"
> >
> > transparent
> >
> > > RTP-forwarder (this is required to get T38 working between the 2
> >
> > endpoints)
> >
> > > -bypass media enabled: FreeSWITCH is not in the RTP path
> > > David Ponzone  Direction Technique
> > > email: david.ponzone at ipeva.fr
> > > tel:      01 74 03 18 97
> > > gsm:   06 66 98 76 34
> > > Service Client IPeva
> > > tel:      0811 46 26 26
> > > www.ipeva.fr  -   www.ipeva-studio.com
> > > Ce message et toutes les pi?ces jointes sont confidentiels et
> >
> > ?tablis ?
> >
> > > l'intention exclusive de ses destinataires. Toute utilisation ou
> >
> > diffusion
> >
> > > non autoris?e est interdite. Tout message ?lectronique est
> >
> > susceptible
> >
> > > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
> >
> > message s'il
> >
> > > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
> >
> > de ce
> >
> > > message, merci de le d?truire imm?diatement et d'avertir
> >
> > l'exp?diteur.
> >
> > > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> > >
> > > Clarification - for G729 does freeswitch need to be in "bypass
> >
> > media"
> >
> > > or "proxy media"? My understanding was that G729 would work with
> > > "proxy media" enabled and without the new fangled module?
> > >
> > > -Max
> > >
> > > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
> >
> > wrote:
> > > Frank Church <vfclists at googlemail.com> wrote:
> > >
> > > I am just trialling Freeswitch with Linksys adapters, whose default
> > >
> > > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> > >
> > > When I change that setting to 'yes' the calls don't go through. I am
> > >
> > > using the latest Windows SVN.
> > >
> > > FreeSWITCH only supports G.729 in pass-through mode (i.e., with
> >
> > bypass media
> >
> > > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
> >
> > you
> >
> > > really
> > >
> > > need it.
> > >
> > > My recommendation would be to use a codec other than G.729 unless
> >
> > you have a
> >
> > > compelling reason, for example a carrier that only supports G.729.
> > >
> > >
> > > _______________________________________________
> > >
> > > FreeSWITCH-users mailing list
> > >
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s
> > >
> > > http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > --
> > Frank Church
> >
> > =======================
> > http://devblog.brahmancreations.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 mrene_lists at avgs.ca  Mon Apr  5 03:59:07 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 5 Apr 2010 06:59:07 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <201004050654.11705.sos@sokhapkin.dyndns.org>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
Message-ID: <26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>

Yup, thats why we even have a param called "NDLB-allow-bad-iananame"  
in sofia profiles.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:

> FS looks at codec name too.
>
> On Monday 05 April 2010, David Ponzone wrote:
>> Ognjen,
>>
>> very good point, but I used to think that for G729 (and all payload  
>> id
>> smaller than 97), FS was relying on the payload id, and not the name.
>>
>> Am I wrong ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>> d'avertir l'exp?diteur.
>>
>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>>> G729a is invalid as a codec name. FS used to allow it but not
>>> anymore afaik.
>>> You should change codec name to G729 (I assume you're using Linksys
>>> product; there is a setting to change under SIP tab).
>>>
>>> Ognjen
>>>
>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>>> <vfclists at googlemail.com> wrote:
>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>>> It has the G729 set in the codecs section.
>>> In this one it seems the call does not get to the external gateway.
>>> Freeswitch stops the call before calling the external gateway
>>>
>>>
>>> I have checked it again a few times using the bypass_media,
>>> proxy_media settings.
>>>
>>> And with those settings the call ends as soon as ringing starts or  
>>> as
>>> sonn as the call is answered.
>>>
>>> I will do another one just to confirm
>>>
>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>  
>>> wrote:
>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>>>> You just need the regular passthrough module: mod_g729 and to
>>>
>>> allow G729 as
>>>
>>>> inbound and outbound codecs in vars.xml.
>>>> To summarize:
>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>>>
>>> the RTP
>>>
>>>> path, it relays the audio stream between endpoints, but can still
>>>
>>> detect
>>>
>>>> DTMFs
>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>>>
>>> transparently, DTMF
>>>
>>>> detection is impossible. In this mode, FS is really a "dumb"
>>>
>>> transparent
>>>
>>>> RTP-forwarder (this is required to get T38 working between the 2
>>>
>>> endpoints)
>>>
>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>
>>> ?tablis ?
>>>
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>
>>> diffusion
>>>
>>>> non autoris?e est interdite. Tout message ?lectronique est
>>>
>>> susceptible
>>>
>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>>>
>>> message s'il
>>>
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
>>>
>>> de ce
>>>
>>>> message, merci de le d?truire imm?diatement et d'avertir
>>>
>>> l'exp?diteur.
>>>
>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>>>
>>>> Clarification - for G729 does freeswitch need to be in "bypass
>>>
>>> media"
>>>
>>>> or "proxy media"? My understanding was that G729 would work with
>>>> "proxy media" enabled and without the new fangled module?
>>>>
>>>> -Max
>>>>
>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>>>
>>> wrote:
>>>> Frank Church <vfclists at googlemail.com> wrote:
>>>>
>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>>
>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>>
>>>> When I change that setting to 'yes' the calls don't go through. I  
>>>> am
>>>>
>>>> using the latest Windows SVN.
>>>>
>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>>>
>>> bypass media
>>>
>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
>>>
>>> you
>>>
>>>> really
>>>>
>>>> need it.
>>>>
>>>> My recommendation would be to use a codec other than G.729 unless
>>>
>>> you have a
>>>
>>>> compelling reason, for example a carrier that only supports G.729.
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> FreeSWITCH-users mailing list
>>>>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-user
>>>> s
>>>>
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-user
>>>> s http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-user
>>>> s http://www.freeswitch.org
>>>
>>> --
>>> Frank Church
>>>
>>> =======================
>>> http://devblog.brahmancreations.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



From oseslija at gmail.com  Mon Apr  5 04:14:41 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 13:14:41 +0200
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>
References: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>
Message-ID: <q2r4468a6771004050414xc4759078h58c9e434c0eb1c5d@mail.gmail.com>

user_context applies to authenticated users. That means if a user
successfully gets its INVITE authenticated either by IP or SIP auth method
its call will end up in ${user_context}.

Ognjen

On Mon, Apr 5, 2010 at 7:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi Brian,
> I've tried user_context variable. Does user_context apply to only
> registered users in freeswitch?
> Because in my case the users are registered on a different software(OSips).
> Also out of curiosity, does the OSips needs to be defined as a Gateway in FS
> for handling the calls in a standard fashion?
>
> Thanks,
>
> --- Jayesh
>
>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Sat, 3 Apr 2010 22:57:19 -0500
>> Subject: Re: [Freeswitch-users] domain-wise context
>> see user_context variable on the user.
>>
>> /b
>>
>> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>>
>> > Hi,
>> > Sorry for sounding so impatient, the anxiety only grew because before
>> posting it to the list I spent a week on reading all the documentation
>> available on and around this topic FS site and mailing lists. I really
>> appreciate and am thankful for your suggestions.
>> > I'll try out the suggestions given by you. Is there a way that we can
>> compare the domain name in the dialplan using regular expressions. Is there
>> a value in condition tag that can be used to compare this, something like
>> "extension_number".
>> > Or should I store the domain value in some variable(something like
>> sip_h_<domain_name>) and compare that variable to take the call to a
>> different context?
>> >
>> > Thanks,
>> >
>> > --- Jayesh
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jayesh.voip at gmail.com  Mon Apr  5 05:24:05 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Mon, 5 Apr 2010 17:54:05 +0530
Subject: [Freeswitch-users] domain-wise context
Message-ID: <m2h4fe8b8271004050524kbda3bbc2hf86948274c934da7@mail.gmail.com>

>
> Hi,
>
   Now i know, why user_context is not working for me. My calls are coming
from Opensips and are not authenticated by Freeswitch.

--- Jayesh

>
> ---------- Forwarded message ----------
> From: Ognjen Seslija <oseslija at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 5 Apr 2010 13:14:41 +0200
> Subject: Re: [Freeswitch-users] domain-wise context
> user_context applies to authenticated users. That means if a user
> successfully gets its INVITE authenticated either by IP or SIP auth method
> its call will end up in ${user_context}.
>
> Ognjen
>
> On Mon, Apr 5, 2010 at 7:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi Brian,
>> I've tried user_context variable. Does user_context apply to only
>> registered users in freeswitch?
>> Because in my case the users are registered on a different
>> software(OSips). Also out of curiosity, does the OSips needs to be defined
>> as a Gateway in FS for handling the calls in a standard fashion?
>>
>> Thanks,
>>
>> --- Jayesh
>>
>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Sat, 3 Apr 2010 22:57:19 -0500
>>> Subject: Re: [Freeswitch-users] domain-wise context
>>> see user_context variable on the user.
>>>
>>> /b
>>>
>>> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>>>
>>> > Hi,
>>> > Sorry for sounding so impatient, the anxiety only grew because before
>>> posting it to the list I spent a week on reading all the documentation
>>> available on and around this topic FS site and mailing lists. I really
>>> appreciate and am thankful for your suggestions.
>>> > I'll try out the suggestions given by you. Is there a way that we can
>>> compare the domain name in the dialplan using regular expressions. Is there
>>> a value in condition tag that can be used to compare this, something like
>>> "extension_number".
>>> > Or should I store the domain value in some variable(something like
>>> sip_h_<domain_name>) and compare that variable to take the call to a
>>> different context?
>>> >
>>> > Thanks,
>>> >
>>> > --- Jayesh
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim at k4gvo.com  Mon Apr  5 05:53:50 2010
From: jim at k4gvo.com (Jim)
Date: Mon, 05 Apr 2010 08:53:50 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>
References: <4BB79458.8080601@k4gvo.com>	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>	<4BB87CA9.3050403@k4gvo.com>
	<h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>
Message-ID: <4BB9DD5E.8030706@k4gvo.com>

I never gets around to reading that file.  It looks like it parses 
conf/dialplan/default.xml, conf/dialplan/default/*.xml and then stops.  
It seems to be matching something in the 99999_enum.xml file and never 
getting any other file.  I read up on enum but I don't really know what 
it's supposed to do.

This is the last bit of parsing he does:

Dialplan: OpenZAP/1:1/7707190068 parsing 
[default->international.example.com] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) 
[international.example.com] ${toll_allow}() =~ /international/ 
break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing [default->enum] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (PASS) [enum] 
${module_exists(mod_enum)}(true) =~ /true/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 Regex (PASS) [enum] 
destination_number(7707190068) =~ /^(.*)$/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 Action transfer(7707190068 enum)


Thanks,
Jim.
Michael Collins wrote:
> I looked in mod_openzap.c and I didn't see any references to channel 
> variables. However, you have context and dialplan options. I suggest 
> that you create a dialplan context just for your FXS port(s). Try 
> this. Create conf/dialplan/fxs-ports.xml:
>
> <include>
>   <context name="fxs-ports">
>     <extension name="outbound">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="set" 
> data="toll_allow=local,domestic,international"/>
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Then in your openzap.conf.xml change the context for the analog 
> span(s) with the FXS ports:
> <param name="context" value="default"/>
>
> Restart FS after making these changes and then give it a shot. You 
> should see the call from the analog phone going into context 
> "fxs-ports" and then get transferred over to the default context where 
> it will act like your SIP phones because we manually set the 
> ${toll_allow} chan var.
>
> -MC
>
> On Sun, Apr 4, 2010 at 4:48 AM, Jim <jim at k4gvo.com 
> <mailto:jim at k4gvo.com>> wrote:
>
>     Michael Collins wrote:
>     > Variables set in the directory/default/xxxx.xml files apply to users
>     > who make authenticated calls through FS. Generally those will be SIP
>     > phones.
>     >
>     > Let's back up a step. What problem are you trying to solve,
>     i.e., why
>     > is it that you need to set the toll_allow variable? What endpoint is
>     > making an openzap call?
>     >
>     > -MC
>     Hi, Michael,
>
>     I have multiple sip phone that are working fine.  When I dial a 10
>     digit
>     number they connect with my sip provider and place the call.  The
>     openzap configured phone gets dial tone and can call other extensions,
>     however when I dial a 10 digit number it gives me a busy.  In the
>     log I
>     see it appears to be failing on the toll_allow test:
>
>     Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call]
>     continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call]
>     ${toll_allow}() =~ /local/ break=on-false
>     Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call]
>     continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call]
>     ${toll_allow}() =~ /domestic/ break=on-false
>     Dialplan: OpenZAP/1:1/7707190068 parsing
>     [default->international.example.com
>     <http://international.example.com>] continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL)
>     [international.example.com <http://international.example.com>]
>     ${toll_allow}() =~ /international/
>     break=on-false
>
>     This is the area when it should be placing the call, I belive.  When
>     placing a call from a sip phone I see:
>
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> parsing [default->local_call]
>     continue=false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Regex (PASS) [local_call]
>     ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Regex (PASS) [local_call]
>     destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(effective_caller_id_number=${outbound_caller_id_number})
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(effective_caller_id_name=${outbound_caller_id_name})
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(continue_on_fail=true)
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     bridge(sofia/gateway/${default_gateway}/7707190068)
>
>     I simply want this extension to be able to dial out.  The
>     configuration
>     is 99% default.
>
>     Thanks,
>     Jim.
>     >
>     > On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com
>     <mailto:jim at k4gvo.com>
>     > <mailto:jim at k4gvo.com <mailto:jim at k4gvo.com>>> wrote:
>     >
>     >     I obviously need to set a <variable name="toll_allow"
>     >     value="domestic,international,local"/> somewhere but I can't
>     >     figure out
>     >     what file to put it in.  The examples all show it in the
>     >     directory/default/xxxx.xml files but those appear to be sip
>     only.  In
>     >     any event creating files in that directory for my extension did
>     >     nothing
>     >     to help the problem.
>     >
>     >     The only places I have the extension mentioned is in  the
>     openzap.conf
>     >     file and dialplan/default/00_incoming-1.xml.  Adding a
>     <variables ...
>     >     element to the latter does nothing.
>     >
>     >     How do I set that variable?  Or where?
>     >
>     >     Thanks,
>     >     Jim.
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto: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
>     <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 12ukwn at gmail.com  Mon Apr  5 06:07:24 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 5 Apr 2010 15:07:24 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
Message-ID: <20100405150724.214af8a2@anubis.defcon1>

Le Sun, 4 Apr 2010 22:19:48 -0500,
Brian West <brian at freeswitch.org> a ?crit :

> Their exists a file in /etc/ called ld.so.conf  if you happen to
> add  /opt/swift/lib/lib into that then run ldconfig it would work.
> 
> muhahahahaha linux is great.

As I wrote in my 1st post: I followed the wiki (and I'm not a Linux rooky),
so I asked here; may be I should reformulate my question: why having to
symlink all ceptsral libs into /usr/local/freswitch/lib while already having
them cached in the system libs cache (/etc/ld.so.cache)?

This is the thing I don't understand...

-- 
For internal use only.


From Russell.Mosemann at cune.org  Mon Apr  5 08:54:08 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Apr 2010 10:54:08 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100405150724.214af8a2@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
Message-ID: <769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>

Jean-Yves F. Barbier wrote:

> so I asked here; may be I should reformulate my question: why having to
> symlink all ceptsral libs into /usr/local/freswitch/lib while already
> having them cached in the system libs cache (/etc/ld.so.cache)?

What does "ldd freeswitch" say?

--
Russell Mosemann



From fraserredmond at gmail.com  Mon Apr  5 09:12:42 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 17:12:42 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
Message-ID: <j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>

I've taken another stab at this one way audio problem today.

I've run a wireshark capture, and looking at the RTP analysis it only has
the down-stream, it doesn't record anything being sent upstream at all.

Below is the SIP graph, which shows RTP coming down, but none going up. But
I don't know enough about SIP to know whether something is missing.

Any suggestions of what I should try now?

Would the dtmf's be sent in the sip packets, or in the rtp?

To preempt the easy answers and save some time:
- ports are open on EC2 config,
- iptables turned off for the test,
- RTP port range uncommented in switch.conf.xml,
- softphone stun was set to stun.freeswitch.org

Cheers,
Fraser


|Time     | 192.168.1.8                           |
|         |                   | 184.73.226.197    |
|6.488    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.615    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         407 Proxy Authentication Required          |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|6.738    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.869    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.070    |         183 Session Progress SDP ( g711U
telephone-eve...          |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.264    |         RTP (g711U)                   |RTP Num packets:520
Duration:10.793s SSRC:0x5433093E
|         |(44172)  <------------------  (30432)  |
|18.090   |         200 OK SDP ( g711U telephone-event)          |SIP Status
|         |(25829)  <------------------  (5060)   |
|18.112   |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.750   |         BYE       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.872   |         200 OK    |                   |SIP Status
|         |(25829)  <------------------  (5060)   |






On Sat, Apr 3, 2010 at 4:49 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I've got a FreeSwitch server up on Amazon EC2, ports wide open for my
> office external-IP, server iptables disabled, and changed the FreeSwitch ACL
> domains to "allow", so it's all wide open for now.
>
> In the office I'm trying to connect to the server from Bria/X-lite. I've
> entered a stun server (stun.freeswitch.org) and I can now call to the
> server, but not from the server. I read this page:
>    http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc
> which suggested adding this variable to the user config:
>
> http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction
>
> With that on I can now call to and from the server. However with or without
> that although I can hear audio from the server, audio to the server isn't
> arriving (doesn't appear in recordings), and dtmf doesn't get received
> either.
>
> When I hang up from the client, I see in the CLI that it gets that
> instruction, so it hasn't started the call and lost all contact with the
> softphone, it's receiving some instructions, but not the audio and dtmf.
>
> The problem is that both the server and client are each behind NAT, so
> either could be having the problem (on EC2 the auto-NAT doesn't work, so
> I've specified the external rtp and sip ip's.. I've also turned on
> aggressive-NAT in case that helps. Also I'm connecting to the server by a
> sub-domain (A-name) rather than IP.)
>
> I've got almost the same setup working fine on the internal network (same
> dialplan and directory, and all the config is the same if it can be), so
> it's got to be something to do with the NAT's.
>
> Any suggestions on what the problem might be, or how to find it?
>
> Cheers,
> Fraser
>
>
>
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From brian at freeswitch.org  Mon Apr  5 09:21:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 11:21:33 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
Message-ID: <4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>

Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace.

/b

On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:

> I've taken another stab at this one way audio problem today.
> 
> I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.
> 
> Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.
> 
> Any suggestions of what I should try now?
> 
> Would the dtmf's be sent in the sip packets, or in the rtp?
> 
> To preempt the easy answers and save some time:
> - ports are open on EC2 config, 
> - iptables turned off for the test,
> - RTP port range uncommented in switch.conf.xml,
> - softphone stun was set to stun.freeswitch.org
> 
> Cheers,
> Fraser

-------------- next part --------------
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From 12ukwn at gmail.com  Mon Apr  5 09:24:14 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 05 Apr 2010 18:24:14 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
Message-ID: <4BBA0EAE.4020003@gmail.com>

Russell Mosemann wrote:
>> so I asked here; may be I should reformulate my question: why having to
>> symlink all ceptsral libs into /usr/local/freswitch/lib while already
>> having them cached in the system libs cache (/etc/ld.so.cache)?
> 
> What does "ldd freeswitch" say?
linux-gate.so.1 =>  (0xb7faf000)
libm.so.6 => /lib/i686/cmov/libm.so.6 (0xb7f5d000)
libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 (0xb7db3000)
libuuid.so.1 => /lib/libuuid.so.1 (0xb7dae000)
librt.so.1 => /lib/i686/cmov/librt.so.1 (0xb7da5000)
libdl.so.2 => /lib/i686/cmov/libdl.so.2 (0xb7da1000)
libcrypt.so.1 => /lib/i686/cmov/libcrypt.so.1 (0xb7d6f000)
libpthread.so.0 => /lib/i686/cmov/libpthread.so.0 (0xb7d56000)
libssl.so.0.9.8 => /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7d0f000)
libcrypto.so.0.9.8 => /usr/lib/i686/cmov/libcrypto.so.0.9.8 (0xb7bbb000)
libncurses.so.5 => /lib/libncurses.so.5 (0xb7b89000)
libc.so.6 => /lib/i686/cmov/libc.so.6 (0xb7a2e000)
/lib/ld-linux.so.2 (0xb7fb0000)
libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0xb7940000)
libgcc_s.so.1 => /lib/libgcc_s.so.1 (0xb7933000)
libodbc.so.1 => /usr/lib/libodbc.so.1 (0xb78d2000)
libz.so.1 => /usr/lib/libz.so.1 (0xb78bd000)
libltdl.so.3 => /usr/lib/libltdl.so.3 (0xb78b6000)

-- 
We don't have to protect the environment -- the Second Coming is at hand.
		-- James Watt, noted ecologist


From Russell.Mosemann at cune.org  Mon Apr  5 09:44:06 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Apr 2010 11:44:06 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <4BBA0EAE.4020003@gmail.com>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>	<20100405150724.214af8a2@anubis.defcon1><769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
Message-ID: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>

Jean-Yves F. Barbier wrote:
> >> so I asked here; may be I should reformulate my question: why having to
> >> symlink all ceptsral libs into /usr/local/freswitch/lib while already
> >> having them cached in the system libs cache (/etc/ld.so.cache)?

When you say that the libs are in the system libs cache, do you mean that the libs are configured in /etc/ld.conf (or /etc/ld.conf.d/, as appropriate), and when you enter "ldconfig -v", the libs are listed in the output under their home directory (was it /opt/something?)?

On rare occasion, I have seen software not find a library for some strange reason when there is a libsomething.so.2.0 but there is no symbolic link for libsomething.so or libsomething.so.0. Adding a symbolic link worked. I don't know if this is the same kind of situation.

--
Russell Mosemann



From vfclists at googlemail.com  Mon Apr  5 10:00:19 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 18:00:19 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
Message-ID: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>

I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
were completely garbled, even the ringing. It could be affecting
something on the provider end.

Could it be that the provider has a different G729 codec that is not
compatible with the actual G729a the Linksys is sending?



On 5 April 2010 11:59, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Yup, thats why we even have a param called "NDLB-allow-bad-iananame"
> in sofia profiles.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
>
>> FS looks at codec name too.
>>
>> On Monday 05 April 2010, David Ponzone wrote:
>>> Ognjen,
>>>
>>> very good point, but I used to think that for G729 (and all payload
>>> id
>>> smaller than 97), FS was relying on the payload id, and not the name.
>>>
>>> Am I wrong ?
>>>
>>> David Ponzone ?Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel: ? ? ?01 74 03 18 97
>>> gsm: ? 06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel: ? ? ?0811 46 26 26
>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
>>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>>> d'avertir l'exp?diteur.
>>>
>>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>>>> G729a is invalid as a codec name. FS used to allow it but not
>>>> anymore afaik.
>>>> You should change codec name to G729 (I assume you're using Linksys
>>>> product; there is a setting to change under SIP tab).
>>>>
>>>> Ognjen
>>>>
>>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>>>> <vfclists at googlemail.com> wrote:
>>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>>>> It has the G729 set in the codecs section.
>>>> In this one it seems the call does not get to the external gateway.
>>>> Freeswitch stops the call before calling the external gateway
>>>>
>>>>
>>>> I have checked it again a few times using the bypass_media,
>>>> proxy_media settings.
>>>>
>>>> And with those settings the call ends as soon as ringing starts or
>>>> as
>>>> sonn as the call is answered.
>>>>
>>>> I will do another one just to confirm
>>>>
>>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>
>>>> wrote:
>>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>>>>> You just need the regular passthrough module: mod_g729 and to
>>>>
>>>> allow G729 as
>>>>
>>>>> inbound and outbound codecs in vars.xml.
>>>>> To summarize:
>>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>>>>
>>>> the RTP
>>>>
>>>>> path, it relays the audio stream between endpoints, but can still
>>>>
>>>> detect
>>>>
>>>>> DTMFs
>>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>>>>
>>>> transparently, DTMF
>>>>
>>>>> detection is impossible. In this mode, FS is really a "dumb"
>>>>
>>>> transparent
>>>>
>>>>> RTP-forwarder (this is required to get T38 working between the 2
>>>>
>>>> endpoints)
>>>>
>>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>>>>> David Ponzone ?Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel: ? ? ?01 74 03 18 97
>>>>> gsm: ? 06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel: ? ? ?0811 46 26 26
>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>>
>>>> ?tablis ?
>>>>
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>
>>>> diffusion
>>>>
>>>>> non autoris?e est interdite. Tout message ?lectronique est
>>>>
>>>> susceptible
>>>>
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>>>>
>>>> message s'il
>>>>
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
>>>>
>>>> de ce
>>>>
>>>>> message, merci de le d?truire imm?diatement et d'avertir
>>>>
>>>> l'exp?diteur.
>>>>
>>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>>>>
>>>>> Clarification - for G729 does freeswitch need to be in "bypass
>>>>
>>>> media"
>>>>
>>>>> or "proxy media"? My understanding was that G729 would work with
>>>>> "proxy media" enabled and without the new fangled module?
>>>>>
>>>>> -Max
>>>>>
>>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>>>>
>>>> wrote:
>>>>> Frank Church <vfclists at googlemail.com> wrote:
>>>>>
>>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>>>
>>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>>>
>>>>> When I change that setting to 'yes' the calls don't go through. I
>>>>> am
>>>>>
>>>>> using the latest Windows SVN.
>>>>>
>>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>>>>
>>>> bypass media
>>>>
>>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
>>>>
>>>> you
>>>>
>>>>> really
>>>>>
>>>>> need it.
>>>>>
>>>>> My recommendation would be to use a codec other than G.729 unless
>>>>
>>>> you have a
>>>>
>>>>> compelling reason, for example a carrier that only supports G.729.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-user
>>>>> s
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-user
>>>>> s http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-user
>>>>> s http://www.freeswitch.org
>>>>
>>>> --
>>>> Frank Church
>>>>
>>>> =======================
>>>> http://devblog.brahmancreations.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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From brian at freeswitch.org  Mon Apr  5 10:09:10 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:09:10 -0500
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
Message-ID: <6602B9C0-96E3-4BC6-A76E-05DD3555D46D@freeswitch.org>

Visit www.freeswitch.org, Click G729 at the top and buy some licenses.  Problem solved ;)  It supports the project and solves a problem all at the same time.

Thanks,
Brian
PS: In depth install instructions are coming shortly for G729.

On Apr 5, 2010, at 12:00 PM, Frank Church wrote:

> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
> were completely garbled, even the ringing. It could be affecting
> something on the provider end.
> 
> Could it be that the provider has a different G729 codec that is not
> compatible with the actual G729a the Linksys is sending?
> 



From fraserredmond at gmail.com  Mon Apr  5 10:28:17 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 18:28:17 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
Message-ID: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>

Thanks Brian. Sorry, should have done a full sip trace before, but here is
one now: <http://pastebin.freeswitch.org/12632>

Calling an IVR dialplan:
http://pastebin.freeswitch.org/12634

Calling from one extn to another.
http://pastebin.freeswitch.org/12633
(With this one, the source/calling softphone gets a message on it saying put
on hold by the other user - not sure if that helps.)

For what it's worth, at a couple of points when I was running the trace I
was pressing keys to generate dtmf, and nothing changed on the screen - no
activity at all.

Also, I've been able to remote desktop into a computer on another network,
and install x-lite and it can connect to our internal server and works fine,
but it can't do dtmf on the EC2 server either (so it's definitely a problem
on the server end somehow, not my local network's NAT.)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 5:21 PM, Brian West <brian at freeswitch.org> wrote:

> Most likely its in the RTP stream as RFC2833 which is the reason you're not
> getting anything plus I need a FULL sip trace not this abbreviated trace.
>
> /b
>
> On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
>
> I've taken another stab at this one way audio problem today.
>
> I've run a wireshark capture, and looking at the RTP analysis it only has
> the down-stream, it doesn't record anything being sent upstream at all.
>
> Below is the SIP graph, which shows RTP coming down, but none going up. But
> I don't know enough about SIP to know whether something is missing.
>
> Any suggestions of what I should try now?
>
> Would the dtmf's be sent in the sip packets, or in the rtp?
>
> To preempt the easy answers and save some time:
> - ports are open on EC2 config,
> - iptables turned off for the test,
> - RTP port range uncommented in switch.conf.xml,
> - softphone stun was set to stun.freeswitch.org
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 10:34:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:34:09 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
Message-ID: <7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>

sofia loglevel all 0
sofia profile xx siptrace on

replace xx with profile.  What you have provided is NOT a sip trace.

Thanks,
Brian

On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:

> Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
> 
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
> 
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
> 
> For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
> 
> Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
> 
> Cheers,
> Fraser

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From brian at freeswitch.org  Mon Apr  5 10:34:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:34:36 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
Message-ID: <A1E647FA-30C1-49FA-9D87-BDE8B353A53F@freeswitch.org>

Also have you checked your firewall?  service iptables stop and test .

/b

On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:

> Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
> 
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
> 
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
> 
> For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
> 
> Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
> 
> Cheers,
> Fraser
> 

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From lloyd.aloysius at gmail.com  Mon Apr  5 10:46:14 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 5 Apr 2010 13:46:14 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
Message-ID: <s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>

Brian,

I reinstall the OS , FreeSWITCH and Setup the configurations.

It is same problem.
.

1. Multi Tenant Setup is not working [ Aastra Phones , 57i Register and
9143is Not Registering and No way to make a calls ] .

But Xlite Working without any problem.

2. When I use the Aasta Phones connecting Default Extension 1000 it is
working.

3. My Local Router LinkSys WRT54GL + Tomato 1.27

Multi Tenant Setup working fine before. Please let me know if you need a SSH
access.

Thanks
Lloyd






On Sun, Apr 4, 2010 at 5:25 PM, Brian West <brian at freeswitch.org> wrote:

> be sure to nuke your configs... cuz we won't when you reinstall.
>
> /b
>
> On Apr 4, 2010, at 4:11 PM, Aloysius Lloyd wrote:
>
> > Thanks Brian.
> >
> > This is simple direct install and no ACL. I am going to Install From
> Scratch.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Mon Apr  5 10:53:54 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 5 Apr 2010 19:53:54 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
Message-ID: <20100405195354.376ea693@anubis.defcon1>

Le Mon, 5 Apr 2010 11:44:06 -0500,
"Russell Mosemann" <Russell.Mosemann at cune.org> a ?crit :

> When you say that the libs are in the system libs cache, do you mean that
> the libs are configured in /etc/ld.conf (or /etc/ld.conf.d/, as
> appropriate), and when you enter "ldconfig -v", the libs are listed in
> the output under their home directory (was it /opt/something?)?

Yep (/etc/ld.so.conf.d/CEPSTRAL.conf, containing '/opt/swift/lib' +
ldconfig), and what is even stranger is that it was like that for a week,
and the last rebuild put everything back in place (I just tested and removed
the symlinks.)

> On rare occasion, I have seen software not find a library for some
> strange reason when there is a libsomething.so.2.0 but there is no
> symbolic link for libsomething.so or libsomething.so.0. Adding a symbolic
> link worked. I don't know if this is the same kind of situation.

May be, may be not, on rare occasions I've seen uncanny interactions into
Linux (but tons less than in m$ (all alpha) products:)

Thanks anyway!

> Jean-Yves F. Barbier wrote:
> > >> so I asked here; may be I should reformulate my question: why having
> > >> to symlink all ceptsral libs into /usr/local/freswitch/lib while
> > >> already having them cached in the system libs cache
> > >> (/etc/ld.so.cache)?
 

-- 
Old programmers never die, they just branch to a new address.


From fraserredmond at gmail.com  Mon Apr  5 10:53:41 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 18:53:41 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
Message-ID: <r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>

Ok, sorry, try this:
http://pastebin.freeswitch.org/12635

And, yes, as I said in my earlier message, the firewall is off - checked it
over and over again :-)

Amazon EC2 has it's own firewall system as well, which can't be turned off,
but I've set it to enable all ports (0-65535)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 6:34 PM, Brian West <brian at freeswitch.org> wrote:

> sofia loglevel all 0
> sofia profile xx siptrace on
>
> replace xx with profile.  What you have provided is NOT a sip trace.
>
> Thanks,
> Brian
>
> On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:
>
> Thanks Brian. Sorry, should have done a full sip trace before, but here is
> one now: <http://pastebin.freeswitch.org/12632>
>
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
>
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying
> put on hold by the other user - not sure if that helps.)
>
> For what it's worth, at a couple of points when I was running the trace I
> was pressing keys to generate dtmf, and nothing changed on the screen - no
> activity at all.
>
> Also, I've been able to remote desktop into a computer on another network,
> and install x-lite and it can connect to our internal server and works fine,
> but it can't do dtmf on the EC2 server either (so it's definitely a problem
> on the server end somehow, not my local network's NAT.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 10:56:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:56:45 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
Message-ID: <A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>

Then you're config is clearly wrong.

/b

On Apr 5, 2010, at 12:46 PM, Aloysius Lloyd wrote:

> Brian,
> 
> I reinstall the OS , FreeSWITCH and Setup the configurations.
> 
> It is same problem.
> .
> 
> 1. Multi Tenant Setup is not working [ Aastra Phones , 57i Register and 9143is Not Registering and No way to make a calls ] . 
> 
> But Xlite Working without any problem.
> 
> 2. When I use the Aasta Phones connecting Default Extension 1000 it is working.
> 
> 3. My Local Router LinkSys WRT54GL + Tomato 1.27
> 
> Multi Tenant Setup working fine before. Please let me know if you need a SSH access.
> 
> Thanks
> Lloyd
> 
> 



From brian at freeswitch.org  Mon Apr  5 11:08:16 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 13:08:16 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
Message-ID: <19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>

While its valid to have host names in the SDP we do NOT and WILL NOT support it please input IP addresses into the rtp-ip and ext-rtp-ip fields.  You'll find that it WORKS BETTER.

/b

On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:

> Ok, sorry, try this:
> http://pastebin.freeswitch.org/12635
> 
> And, yes, as I said in my earlier message, the firewall is off - checked it over and over again :-) 
> 
> Amazon EC2 has it's own firewall system as well, which can't be turned off, but I've set it to enable all ports (0-65535)
> 
> Cheers,
> Fraser

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From david.ponzone at gmail.com  Mon Apr  5 11:16:01 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 20:16:01 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
Message-ID: <E34DB2E8-E29B-48A2-8DF3-887DCD3AAF1D@gmail.com>

Frank,

can you please try that with another phone or softphone ?
You will then know if it's the Linksys which is faulty or your provider.

Also, you can try to upgrade the ATA to the latest firmware.
I recently bought a PAP2T, and the firmware on that was not the latest  
one (it was v4, but v5 is the current version).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 19:00, Frank Church a ?crit :

> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
> were completely garbled, even the ringing. It could be affecting
> something on the provider end.
>
> Could it be that the provider has a different G729 codec that is not
> compatible with the actual G729a the Linksys is sending?
>
>
>
> On 5 April 2010 11:59, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>> Yup, thats why we even have a param called "NDLB-allow-bad-iananame"
>> in sofia profiles.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
>>
>>> FS looks at codec name too.
>>>
>>> On Monday 05 April 2010, David Ponzone wrote:
>>>> Ognjen,
>>>>
>>>> very good point, but I used to think that for G729 (and all payload
>>>> id
>>>> smaller than 97), FS was relying on the payload id, and not the  
>>>> name.
>>>>
>>>> Am I wrong ?
>>>>
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>>>> ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>>> diffusion non autoris?e est interdite. Tout message ?lectronique  
>>>> est
>>>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
>>>> titre
>>>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous  
>>>> n'?tes
>>>> pas destinataire de ce message, merci de le d?truire  
>>>> imm?diatement et
>>>> d'avertir l'exp?diteur.
>>>>
>>>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>>>>> G729a is invalid as a codec name. FS used to allow it but not
>>>>> anymore afaik.
>>>>> You should change codec name to G729 (I assume you're using  
>>>>> Linksys
>>>>> product; there is a setting to change under SIP tab).
>>>>>
>>>>> Ognjen
>>>>>
>>>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>>>>> <vfclists at googlemail.com> wrote:
>>>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>>>>> It has the G729 set in the codecs section.
>>>>> In this one it seems the call does not get to the external  
>>>>> gateway.
>>>>> Freeswitch stops the call before calling the external gateway
>>>>>
>>>>>
>>>>> I have checked it again a few times using the bypass_media,
>>>>> proxy_media settings.
>>>>>
>>>>> And with those settings the call ends as soon as ringing starts or
>>>>> as
>>>>> sonn as the call is answered.
>>>>>
>>>>> I will do another one just to confirm
>>>>>
>>>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>
>>>>> wrote:
>>>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy  
>>>>>> media.
>>>>>> You just need the regular passthrough module: mod_g729 and to
>>>>>
>>>>> allow G729 as
>>>>>
>>>>>> inbound and outbound codecs in vars.xml.
>>>>>> To summarize:
>>>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>>>>>
>>>>> the RTP
>>>>>
>>>>>> path, it relays the audio stream between endpoints, but can still
>>>>>
>>>>> detect
>>>>>
>>>>>> DTMFs
>>>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>>>>>
>>>>> transparently, DTMF
>>>>>
>>>>>> detection is impossible. In this mode, FS is really a "dumb"
>>>>>
>>>>> transparent
>>>>>
>>>>>> RTP-forwarder (this is required to get T38 working between the 2
>>>>>
>>>>> endpoints)
>>>>>
>>>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>>>>>> David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>>>
>>>>> ?tablis ?
>>>>>
>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>>
>>>>> diffusion
>>>>>
>>>>>> non autoris?e est interdite. Tout message ?lectronique est
>>>>>
>>>>> susceptible
>>>>>
>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>>>>>
>>>>> message s'il
>>>>>
>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas  
>>>>>> destinataire
>>>>>
>>>>> de ce
>>>>>
>>>>>> message, merci de le d?truire imm?diatement et d'avertir
>>>>>
>>>>> l'exp?diteur.
>>>>>
>>>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>>>>>
>>>>>> Clarification - for G729 does freeswitch need to be in "bypass
>>>>>
>>>>> media"
>>>>>
>>>>>> or "proxy media"? My understanding was that G729 would work with
>>>>>> "proxy media" enabled and without the new fangled module?
>>>>>>
>>>>>> -Max
>>>>>>
>>>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>>>>>
>>>>> wrote:
>>>>>> Frank Church <vfclists at googlemail.com> wrote:
>>>>>>
>>>>>> I am just trialling Freeswitch with Linksys adapters, whose  
>>>>>> default
>>>>>>
>>>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>>>>
>>>>>> When I change that setting to 'yes' the calls don't go through. I
>>>>>> am
>>>>>>
>>>>>> using the latest Windows SVN.
>>>>>>
>>>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>>>>>
>>>>> bypass media
>>>>>
>>>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH  
>>>>>> if
>>>>>
>>>>> you
>>>>>
>>>>>> really
>>>>>>
>>>>>> need it.
>>>>>>
>>>>>> My recommendation would be to use a codec other than G.729 unless
>>>>>
>>>>> you have a
>>>>>
>>>>>> compelling reason, for example a carrier that only supports G. 
>>>>>> 729.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>>> freeswitch-user
>>>>>> s
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>>> freeswitch-user
>>>>>> s http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>>> freeswitch-user
>>>>>> s http://www.freeswitch.org
>>>>>
>>>>> --
>>>>> Frank Church
>>>>>
>>>>> =======================
>>>>> http://devblog.brahmancreations.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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 fraserredmond at gmail.com  Mon Apr  5 11:20:26 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 19:20:26 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
	<19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>
Message-ID: <k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>

Thanks Brian, it's annoying that there's some places that say to only have
IP's and elsewhere it says domain is ok.

I'll change it over and try again now, but I had tried putting those back to
IP's (I've tried everything!) in some of my earlier testing the other day it
hadn't helped. But I'll double-check now. Do you want another trace using
IP's if it's still not working?

Cheers,
Fraser




On Mon, Apr 5, 2010 at 7:08 PM, Brian West <brian at freeswitch.org> wrote:

> While its valid to have host names in the SDP we do NOT and WILL NOT
> support it please input IP addresses into the rtp-ip and ext-rtp-ip fields.
>  You'll find that it WORKS BETTER.
>
> /b
>
> On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:
>
> Ok, sorry, try this:
> http://pastebin.freeswitch.org/12635
>
> And, yes, as I said in my earlier message, the firewall is off - checked it
> over and over again :-)
>
> Amazon EC2 has it's own firewall system as well, which can't be turned off,
> but I've set it to enable all ports (0-65535)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 11:29:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 13:29:14 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
	<19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>
	<k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>
Message-ID: <CA0F4C45-9506-4B61-B9E2-D397DA578915@freeswitch.org>

Plus you can't trust devices will resolve that... which is part of what I suspect is going on.

/b

On Apr 5, 2010, at 1:20 PM, Fraser Redmond wrote:

> Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok.
> 
> I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working?
> 
> Cheers,
> Fraser



From oseslija at gmail.com  Mon Apr  5 11:29:29 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 20:29:29 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
Message-ID: <k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>

Try to set ptime from 0.030 (default) to 0.020 (20ms).

Ognjen

On Mon, Apr 5, 2010 at 7:00 PM, Frank Church <vfclists at googlemail.com>wrote:

> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
> were completely garbled, even the ringing. It could be affecting
> something on the provider end.
>
> Could it be that the provider has a different G729 codec that is not
> compatible with the actual G729a the Linksys is sending?
>
>
>
> On 5 April 2010 11:59, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> > Yup, thats why we even have a param called "NDLB-allow-bad-iananame"
> > in sofia profiles.
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mrene at avgs.ca
> >
> >
> >
> >
> > On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
> >
> >> FS looks at codec name too.
> >>
> >> On Monday 05 April 2010, David Ponzone wrote:
> >>> Ognjen,
> >>>
> >>> very good point, but I used to think that for G729 (and all payload
> >>> id
> >>> smaller than 97), FS was relying on the payload id, and not the name.
> >>>
> >>> Am I wrong ?
> >>>
> >>> David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>>
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>>
> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> >>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> >>> diffusion non autoris?e est interdite. Tout message ?lectronique est
> >>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> >>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> >>> pas destinataire de ce message, merci de le d?truire imm?diatement et
> >>> d'avertir l'exp?diteur.
> >>>
> >>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
> >>>> G729a is invalid as a codec name. FS used to allow it but not
> >>>> anymore afaik.
> >>>> You should change codec name to G729 (I assume you're using Linksys
> >>>> product; there is a setting to change under SIP tab).
> >>>>
> >>>> Ognjen
> >>>>
> >>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
> >>>> <vfclists at googlemail.com> wrote:
> >>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> >>>> It has the G729 set in the codecs section.
> >>>> In this one it seems the call does not get to the external gateway.
> >>>> Freeswitch stops the call before calling the external gateway
> >>>>
> >>>>
> >>>> I have checked it again a few times using the bypass_media,
> >>>> proxy_media settings.
> >>>>
> >>>> And with those settings the call ends as soon as ringing starts or
> >>>> as
> >>>> sonn as the call is answered.
> >>>>
> >>>> I will do another one just to confirm
> >>>>
> >>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>
> >>>> wrote:
> >>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> >>>>> You just need the regular passthrough module: mod_g729 and to
> >>>>
> >>>> allow G729 as
> >>>>
> >>>>> inbound and outbound codecs in vars.xml.
> >>>>> To summarize:
> >>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
> >>>>
> >>>> the RTP
> >>>>
> >>>>> path, it relays the audio stream between endpoints, but can still
> >>>>
> >>>> detect
> >>>>
> >>>>> DTMFs
> >>>>> -proxy media enabled: FreeSWITCH relays the audio stream
> >>>>
> >>>> transparently, DTMF
> >>>>
> >>>>> detection is impossible. In this mode, FS is really a "dumb"
> >>>>
> >>>> transparent
> >>>>
> >>>>> RTP-forwarder (this is required to get T38 working between the 2
> >>>>
> >>>> endpoints)
> >>>>
> >>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
> >>>>> David Ponzone  Direction Technique
> >>>>> email: david.ponzone at ipeva.fr
> >>>>> tel:      01 74 03 18 97
> >>>>> gsm:   06 66 98 76 34
> >>>>> Service Client IPeva
> >>>>> tel:      0811 46 26 26
> >>>>> www.ipeva.fr  -   www.ipeva-studio.com
> >>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
> >>>>
> >>>> ?tablis ?
> >>>>
> >>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
> >>>>
> >>>> diffusion
> >>>>
> >>>>> non autoris?e est interdite. Tout message ?lectronique est
> >>>>
> >>>> susceptible
> >>>>
> >>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
> >>>>
> >>>> message s'il
> >>>>
> >>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
> >>>>
> >>>> de ce
> >>>>
> >>>>> message, merci de le d?truire imm?diatement et d'avertir
> >>>>
> >>>> l'exp?diteur.
> >>>>
> >>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >>>>>
> >>>>> Clarification - for G729 does freeswitch need to be in "bypass
> >>>>
> >>>> media"
> >>>>
> >>>>> or "proxy media"? My understanding was that G729 would work with
> >>>>> "proxy media" enabled and without the new fangled module?
> >>>>>
> >>>>> -Max
> >>>>>
> >>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
> >>>>
> >>>> wrote:
> >>>>> Frank Church <vfclists at googlemail.com> wrote:
> >>>>>
> >>>>> I am just trialling Freeswitch with Linksys adapters, whose default
> >>>>>
> >>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >>>>>
> >>>>> When I change that setting to 'yes' the calls don't go through. I
> >>>>> am
> >>>>>
> >>>>> using the latest Windows SVN.
> >>>>>
> >>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
> >>>>
> >>>> bypass media
> >>>>
> >>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
> >>>>
> >>>> you
> >>>>
> >>>>> really
> >>>>>
> >>>>> need it.
> >>>>>
> >>>>> My recommendation would be to use a codec other than G.729 unless
> >>>>
> >>>> you have a
> >>>>
> >>>>> compelling reason, for example a carrier that only supports G.729.
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>>
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> >>>>> freeswitch-user
> >>>>> s
> >>>>>
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> >>>>> freeswitch-user
> >>>>> s http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> >>>>> freeswitch-user
> >>>>> s http://www.freeswitch.org
> >>>>
> >>>> --
> >>>> Frank Church
> >>>>
> >>>> =======================
> >>>> http://devblog.brahmancreations.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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 lloyd.aloysius at gmail.com  Mon Apr  5 11:31:08 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 5 Apr 2010 14:31:08 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
Message-ID: <p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>

Brian,

I do not think so.

If the config is wrong why the xlite is working for the Multitenat
Extension. The same extension is registering and not able to make call . In
another phone even not registering.

Thanks
Lloyd




On Mon, Apr 5, 2010 at 1:56 PM, Brian West <brian at freeswitch.org> wrote:

> Then you're config is clearly wrong.
>
> /b
>
> On Apr 5, 2010, at 12:46 PM, Aloysius Lloyd wrote:
>
> > Brian,
> >
> > I reinstall the OS , FreeSWITCH and Setup the configurations.
> >
> > It is same problem.
> > .
> >
> > 1. Multi Tenant Setup is not working [ Aastra Phones , 57i Register and
> 9143is Not Registering and No way to make a calls ] .
> >
> > But Xlite Working without any problem.
> >
> > 2. When I use the Aasta Phones connecting Default Extension 1000 it is
> working.
> >
> > 3. My Local Router LinkSys WRT54GL + Tomato 1.27
> >
> > Multi Tenant Setup working fine before. Please let me know if you need a
> SSH access.
> >
> > Thanks
> > Lloyd
> >
> >
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraserredmond at gmail.com  Mon Apr  5 11:31:21 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 19:31:21 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
	<19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>
	<k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>
Message-ID: <n2uc32ee1cf1004051131udb886742sbdbbebf79bf8c20a@mail.gmail.com>

That works! Awesome! Thank you so much :-)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 7:20 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks Brian, it's annoying that there's some places that say to only have
> IP's and elsewhere it says domain is ok.
>
> I'll change it over and try again now, but I had tried putting those back
> to IP's (I've tried everything!) in some of my earlier testing the other day
> it hadn't helped. But I'll double-check now. Do you want another trace using
> IP's if it's still not working?
>
> Cheers,
> Fraser
>
>
>
>
> On Mon, Apr 5, 2010 at 7:08 PM, Brian West <brian at freeswitch.org> wrote:
>
>> While its valid to have host names in the SDP we do NOT and WILL NOT
>> support it please input IP addresses into the rtp-ip and ext-rtp-ip fields.
>>  You'll find that it WORKS BETTER.
>>
>> /b
>>
>> On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:
>>
>> Ok, sorry, try this:
>> http://pastebin.freeswitch.org/12635
>>
>> And, yes, as I said in my earlier message, the firewall is off - checked
>> it over and over again :-)
>>
>> Amazon EC2 has it's own firewall system as well, which can't be turned
>> off, but I've set it to enable all ports (0-65535)
>>
>> Cheers,
>> Fraser
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Apr  5 11:40:51 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 13:40:51 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
Message-ID: <A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>

Well based on your logs its clearly setup wrong... did you wipe the config? and start over and how are you doing multi-tenant?  Can you explain to me the steps you took to make a multi-tenant setup?

/b

On Apr 5, 2010, at 1:31 PM, Aloysius Lloyd wrote:

> 
> Brian,
>  
> I do not think so.
>  
> If the config is wrong why the xlite is working for the Multitenat Extension. The same extension is registering and not able to make call . In another phone even not registering.
>  
> Thanks
> Lloyd
> 



From brian at freeswitch.org  Mon Apr  5 11:41:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 13:41:22 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <n2uc32ee1cf1004051131udb886742sbdbbebf79bf8c20a@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
	<19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>
	<k2wc32ee1cf1004051120gf69a1ed1ic52794ed824ebc0d@mail.gmail.com>
	<n2uc32ee1cf1004051131udb886742sbdbbebf79bf8c20a@mail.gmail.com>
Message-ID: <8BD90D48-F591-48E6-80FC-E860CE499F54@freeswitch.org>

Your very welcome.  You might think about adding a wiki entry in the FAQ or the setup guide that outlines that.

/b

On Apr 5, 2010, at 1:31 PM, Fraser Redmond wrote:

> That works! Awesome! Thank you so much :-)
> 
> Cheers,
> Fraser
> 
> 
> 
> 
> On Mon, Apr 5, 2010 at 7:20 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok.
> 
> I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working?
> 
> Cheers,
> Fraser
> 

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From justin at ejtown.org  Mon Apr  5 12:47:34 2010
From: justin at ejtown.org (Justin B Newman)
Date: Mon, 5 Apr 2010 15:47:34 -0400
Subject: [Freeswitch-users] mod_xml_cdr DNS
Message-ID: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>

I have mod_xml_cdr set with a url to POST entries to.

The DNS record is something along the lines of:

cdr.mydomain.com.		600	IN	CNAME	mydomain.com.
mydomain.com.		600	IN	A	10.10.10.1
mydomain.com.		600	IN	A	10.10.10.2

Only one of the two web servers is receiving any POSTs.

I tried adding "curl_easy_setopt(curl_handle,
CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail.

Any ideas?

-jbn


From vfclists at googlemail.com  Mon Apr  5 13:21:44 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 21:21:44 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <6602B9C0-96E3-4BC6-A76E-05DD3555D46D@freeswitch.org>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<6602B9C0-96E3-4BC6-A76E-05DD3555D46D@freeswitch.org>
Message-ID: <w2u7abab2411004051321n8b04bca3z1f379df710d24c0b@mail.gmail.com>

On 5 April 2010 18:09, Brian West <brian at freeswitch.org> wrote:
> Visit www.freeswitch.org, Click G729 at the top and buy some licenses. ?Problem solved ;) ?It supports the project and solves a problem all at the same time.
>

I am experimenting with Freeswitch for embedded bundled product
offered freely to a service provider's customers, which is already
running well on an Asterisk VM. I don't see them forking up for
licenses on every installation. The product and the adapters that are
bundled with it is a loss leader, sending out an installer costs them
even more.

I am looking at Freeswitch because of its native Windows support, as
it avoids the VM route which makes the installation more complicated
for less savvy or impatient users.

For the Freeswitch developers I would say that there is a large
installed base of Linksys adapters,their clones and other VoIP
adapters working well with Asterisk's passthru. If the Linksys G729
passthru is failing because of an additional letter 'a' where it is
not expected, then it should be fixed or reverted. If the
"NDLB-allow-bad-iananame" is what fixes it should be documented
upfront.

After 3 days playing with Freeswitch and checking the documentation I
see Freeswitch as the way forward for a development oriented user, on
both Linux and Windows, but minor end user issues like these ought to
be avoided, especially if the existing alternatives work well. It is
only a developer or an enthusiast who would put in the effort of
switching, because they can see the advantage.


PS. After using "NDLB-allow-bad-iananame" in both the internal and
external profiles the problem still exists. Does it work on its own,
or does it permit the use of 'G729a' in the codec parameters in
vars.xml?

> Thanks,
> Brian
> PS: In depth install instructions are coming shortly for G729.
>
> On Apr 5, 2010, at 12:00 PM, Frank Church wrote:
>
>> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
>> were completely garbled, even the ringing. It could be affecting
>> something on the provider end.
>>
>> Could it be that the provider has a different G729 codec that is not
>> compatible with the actual G729a the Linksys is sending?
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From vfclists at googlemail.com  Mon Apr  5 13:43:55 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 21:43:55 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
Message-ID: <x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>

I can't see a ptime in the Linksys settings, is that what is labelled
as 'RTP Packet Size' on the SIP page?

It is the only thing that has the value 0.030

On 5 April 2010 19:29, Ognjen Seslija <oseslija at gmail.com> wrote:
> Try to set ptime from 0.030 (default) to 0.020 (20ms).
>
> Ognjen
>
> On Mon, Apr 5, 2010 at 7:00 PM, Frank Church <vfclists at googlemail.com>
> wrote:
>>
>> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
>> were completely garbled, even the ringing. It could be affecting
>> something on the provider end.
>>
>> Could it be that the provider has a different G729 codec that is not
>> compatible with the actual G729a the Linksys is sending?
>>
>>
>>
>> On 5 April 2010 11:59, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>> > Yup, thats why we even have a param called "NDLB-allow-bad-iananame"
>> > in sofia profiles.
>> >
>> > Mathieu Rene
>> > Avant-Garde Solutions Inc
>> > Office: + 1 (514) 664-1044 x100
>> > Cell: +1 (514) 664-1044 x200
>> > mrene at avgs.ca
>> >
>> >
>> >
>> >
>> > On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
>> >
>> >> FS looks at codec name too.
>> >>
>> >> On Monday 05 April 2010, David Ponzone wrote:
>> >>> Ognjen,
>> >>>
>> >>> very good point, but I used to think that for G729 (and all payload
>> >>> id
>> >>> smaller than 97), FS was relying on the payload id, and not the name.
>> >>>
>> >>> Am I wrong ?
>> >>>
>> >>> David Ponzone ?Direction Technique
>> >>> email: david.ponzone at ipeva.fr
>> >>> tel: ? ? ?01 74 03 18 97
>> >>> gsm: ? 06 66 98 76 34
>> >>>
>> >>> Service Client IPeva
>> >>> tel: ? ? ?0811 46 26 26
>> >>> www.ipeva.fr ?- ? www.ipeva-studio.com
>> >>>
>> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> >>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>> >>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> >>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
>> >>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>> >>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>> >>> d'avertir l'exp?diteur.
>> >>>
>> >>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>> >>>> G729a is invalid as a codec name. FS used to allow it but not
>> >>>> anymore afaik.
>> >>>> You should change codec name to G729 (I assume you're using Linksys
>> >>>> product; there is a setting to change under SIP tab).
>> >>>>
>> >>>> Ognjen
>> >>>>
>> >>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>> >>>> <vfclists at googlemail.com> wrote:
>> >>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>> >>>> It has the G729 set in the codecs section.
>> >>>> In this one it seems the call does not get to the external gateway.
>> >>>> Freeswitch stops the call before calling the external gateway
>> >>>>
>> >>>>
>> >>>> I have checked it again a few times using the bypass_media,
>> >>>> proxy_media settings.
>> >>>>
>> >>>> And with those settings the call ends as soon as ringing starts or
>> >>>> as
>> >>>> sonn as the call is answered.
>> >>>>
>> >>>> I will do another one just to confirm
>> >>>>
>> >>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>
>> >>>> wrote:
>> >>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> >>>>> You just need the regular passthrough module: mod_g729 and to
>> >>>>
>> >>>> allow G729 as
>> >>>>
>> >>>>> inbound and outbound codecs in vars.xml.
>> >>>>> To summarize:
>> >>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>> >>>>
>> >>>> the RTP
>> >>>>
>> >>>>> path, it relays the audio stream between endpoints, but can still
>> >>>>
>> >>>> detect
>> >>>>
>> >>>>> DTMFs
>> >>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>> >>>>
>> >>>> transparently, DTMF
>> >>>>
>> >>>>> detection is impossible. In this mode, FS is really a "dumb"
>> >>>>
>> >>>> transparent
>> >>>>
>> >>>>> RTP-forwarder (this is required to get T38 working between the 2
>> >>>>
>> >>>> endpoints)
>> >>>>
>> >>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> >>>>> David Ponzone ?Direction Technique
>> >>>>> email: david.ponzone at ipeva.fr
>> >>>>> tel: ? ? ?01 74 03 18 97
>> >>>>> gsm: ? 06 66 98 76 34
>> >>>>> Service Client IPeva
>> >>>>> tel: ? ? ?0811 46 26 26
>> >>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>> >>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>> >>>>
>> >>>> ?tablis ?
>> >>>>
>> >>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> >>>>
>> >>>> diffusion
>> >>>>
>> >>>>> non autoris?e est interdite. Tout message ?lectronique est
>> >>>>
>> >>>> susceptible
>> >>>>
>> >>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>> >>>>
>> >>>> message s'il
>> >>>>
>> >>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
>> >>>>
>> >>>> de ce
>> >>>>
>> >>>>> message, merci de le d?truire imm?diatement et d'avertir
>> >>>>
>> >>>> l'exp?diteur.
>> >>>>
>> >>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>> >>>>>
>> >>>>> Clarification - for G729 does freeswitch need to be in "bypass
>> >>>>
>> >>>> media"
>> >>>>
>> >>>>> or "proxy media"? My understanding was that G729 would work with
>> >>>>> "proxy media" enabled and without the new fangled module?
>> >>>>>
>> >>>>> -Max
>> >>>>>
>> >>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>> >>>>
>> >>>> wrote:
>> >>>>> Frank Church <vfclists at googlemail.com> wrote:
>> >>>>>
>> >>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>> >>>>>
>> >>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>> >>>>>
>> >>>>> When I change that setting to 'yes' the calls don't go through. I
>> >>>>> am
>> >>>>>
>> >>>>> using the latest Windows SVN.
>> >>>>>
>> >>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>> >>>>
>> >>>> bypass media
>> >>>>
>> >>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
>> >>>>
>> >>>> you
>> >>>>
>> >>>>> really
>> >>>>>
>> >>>>> need it.
>> >>>>>
>> >>>>> My recommendation would be to use a codec other than G.729 unless
>> >>>>
>> >>>> you have a
>> >>>>
>> >>>>> compelling reason, for example a carrier that only supports G.729.
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>>
>> >>>>> FreeSWITCH-users mailing list
>> >>>>>
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> >>>>> freeswitch-user
>> >>>>> s
>> >>>>>
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> >>>>> freeswitch-user
>> >>>>> s http://www.freeswitch.org
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> >>>>> freeswitch-user
>> >>>>> s http://www.freeswitch.org
>> >>>>
>> >>>> --
>> >>>> Frank Church
>> >>>>
>> >>>> =======================
>> >>>> http://devblog.brahmancreations.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
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From brian at freeswitch.org  Mon Apr  5 13:47:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 15:47:41 -0500
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
Message-ID: <34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>

change it to 0.020

/b

On Apr 5, 2010, at 3:43 PM, Frank Church wrote:

> I can't see a ptime in the Linksys settings, is that what is labelled
> as 'RTP Packet Size' on the SIP page?
> 
> It is the only thing that has the value 0.030



From vfclists at googlemail.com  Mon Apr  5 14:02:53 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 22:02:53 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
Message-ID: <g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>

I have changed it and this time I get this error

2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
usable in passthrough mode!
2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error!

When it is on 0.030 I get the garbled sound.

I guess the 0.020 is more compatible and I now have to set the pass
through mode. Is there a parameter for that?


On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
> change it to 0.020
>
> /b
>
> On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
>
>> I can't see a ptime in the Linksys settings, is that what is labelled
>> as 'RTP Packet Size' on the SIP page?
>>
>> It is the only thing that has the value 0.030
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From jerry.richards at teotech.com  Mon Apr  5 14:20:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Apr 2010 14:20:06 -0700
Subject: [Freeswitch-users] How to Disable FS Session Timers?
Message-ID: <2F343E3FAEC34014B558F5173AED9A4C@greyhawk.tonecommander.com>


How do I disable session timers in Freeswitch?  Currently, FS is including
"Supported: timer" during call establishment.  Is there an FS tag to disable
this?

Best Regards,
Jerry



From oseslija at gmail.com  Mon Apr  5 14:27:14 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 23:27:14 +0200
Subject: [Freeswitch-users] How to Disable FS Session Timers?
In-Reply-To: <2F343E3FAEC34014B558F5173AED9A4C@greyhawk.tonecommander.com>
References: <2F343E3FAEC34014B558F5173AED9A4C@greyhawk.tonecommander.com>
Message-ID: <m2n4468a6771004051427x84362396q6e5d2fe126d9eec9@mail.gmail.com>

enable-timer=false in profile.



On Mon, Apr 5, 2010 at 11:20 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> How do I disable session timers in Freeswitch?  Currently, FS is including
> "Supported: timer" during call establishment.  Is there an FS tag to
> disable
> this?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Mon Apr  5 14:27:14 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 5 Apr 2010 17:27:14 -0400
Subject: [Freeswitch-users] How to Disable FS Session Timers?
In-Reply-To: <2F343E3FAEC34014B558F5173AED9A4C@greyhawk.tonecommander.com>
References: <2F343E3FAEC34014B558F5173AED9A4C@greyhawk.tonecommander.com>
Message-ID: <0C32579C-228B-451F-8FB4-2FAE3AA741F3@avgs.ca>

In the sip profile:
     <!--
	enable/disable session timers
     -->
     <!--<param name="enable-timer" value="false"/>-->

Uncomment that line.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-05, at 5:20 PM, Jerry Richards wrote:

>
> How do I disable session timers in Freeswitch?  Currently, FS is  
> including
> "Supported: timer" during call establishment.  Is there an FS tag to  
> disable
> this?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From jerry.richards at teotech.com  Mon Apr  5 14:29:54 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Apr 2010 14:29:54 -0700
Subject: [Freeswitch-users] Automatic IP Routing Table Updates From
	Freeswitch?
Message-ID: <510B7115A47E46B1A5BADF0522093221@greyhawk.tonecommander.com>

We are running Freeswitch on a CentOS 5.3 machine with two network ports
connected to two different networks.  Is there an example of Freeswitch
dynamically routing SIP traffic (i.e. update CentOS IP Routing Table?) based
on which sip_profile the extension is REGISTERing with.  This would apply to
registration and call routing (and RTP packets).

Best Regards,
Jerry



From oseslija at gmail.com  Mon Apr  5 14:44:33 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 23:44:33 +0200
Subject: [Freeswitch-users] Automatic IP Routing Table Updates From
	Freeswitch?
In-Reply-To: <510B7115A47E46B1A5BADF0522093221@greyhawk.tonecommander.com>
References: <510B7115A47E46B1A5BADF0522093221@greyhawk.tonecommander.com>
Message-ID: <v2q4468a6771004051444w33b2cc3q8aedb2f3a99ef37e@mail.gmail.com>

FS will always send its UA packets depending on profiles configuration (IP A
for network1, IP B for network 2).
It cannot influence routing though, as it's not a routing software.



On Mon, Apr 5, 2010 at 11:29 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> We are running Freeswitch on a CentOS 5.3 machine with two network ports
> connected to two different networks.  Is there an example of Freeswitch
> dynamically routing SIP traffic (i.e. update CentOS IP Routing Table?)
> based
> on which sip_profile the extension is REGISTERing with.  This would apply
> to
> registration and call routing (and RTP packets).
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dule.maillist at gmail.com  Mon Apr  5 14:54:25 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 5 Apr 2010 17:54:25 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
Message-ID: <s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>

Sorry to dig up an old thread, but ran into this same problem, and trying to
understand the RFC as what is actually the expected behaviour.

So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams
section:

For sendrecv RTP
> streams, the payload type numbers indicate the value of the payload
> type field the offerer expects to receive, and would prefer to send
>
>
>
Doesn't that mean if they say 96 and FS says 101, they should be listening
on 96 and FS should be listening on 101? Kinda like how ports work, you
listen on the port that you offer.

Am I completely misinterpreting?

Thanks,
Dan


On Sat, Oct 31, 2009 at 11:58 AM, Brian West <brian at freeswitch.org> wrote:

> We have <param name="rfc2833-pt" value="101"/> on the profile and you
> to set this.
>
> So you can set it to 96 if needed.  But you shouldn't have to do that
> if they say 96 and we say 101 they should be listening on 101 from us
> and we should be listening on 96 from them... thats why its called an
> RTP map.
>
> /b
>
> On Oct 31, 2009, at 10:48 AM, Patrick List wrote:
>
> >> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
> >>
> >>> ok, as written, i come back after some tests with fs and a thomson
> >>> cirpack.
> >
> > No idea if this is useful as I'm a noob with fs. If not please excuse
> > the noise. In the past Asterisk to work properly with Cirpack needed
> > the
> > following patch:
> >
> > diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
> > --- asterisk-1.4.19.org/main/rtp.c    2007-10-08 22:06:33.000000000
> +0200
> > +++ asterisk-1.4.19/main/rtp.c        2007-11-11 13:12:28.000000000 +0100
> > @@ -1383,6 +1383,7 @@
> >       [34] = {1, AST_FORMAT_H263},
> >       [103] = {1, AST_FORMAT_H263_PLUS},
> >       [97] = {1, AST_FORMAT_ILBC},
> > +     [96] = {0, AST_RTP_DTMF},
> >       [99] = {1, AST_FORMAT_H264},
> >       [101] = {0, AST_RTP_DTMF},
> >       [110] = {1, AST_FORMAT_SPEEX},
> >
> > Maybe this helps.
> >
> > Regards,
> > Patrick
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 14:58:49 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 16:58:49 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
	<s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
Message-ID: <AC077181-A885-4F33-851A-0C12E2FE2FC8@freeswitch.org>

Yes thats how I read it... have you downloaded the latest freeswitch?

/b

On Apr 5, 2010, at 4:54 PM, Dan Le wrote:

> Sorry to dig up an old thread, but ran into this same problem, and trying to understand the RFC as what is actually the expected behaviour.
> 
> So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams section:
> 
> For sendrecv RTP
> streams, the payload type numbers indicate the value of the payload
> type field the offerer expects to receive, and would prefer to send
> 
> 
> Doesn't that mean if they say 96 and FS says 101, they should be listening on 96 and FS should be listening on 101? Kinda like how ports work, you listen on the port that you offer.
> 
> Am I completely misinterpreting?
> 
> Thanks,
> Dan

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From lloyd.aloysius at gmail.com  Mon Apr  5 15:08:51 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 5 Apr 2010 18:08:51 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
	<A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
Message-ID: <m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>

Brian,

Further to my previous off list email.

I can confirm now 100% something went wrong between 17155 to 17188.

The same setup working on the following version

freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17155)

But not working in the following version

freeswitch at internal> version
FreeSWITCH Version 1.0.head (svn-17188)

Thanks
Lloyd
On Mon, Apr 5, 2010 at 2:40 PM, Brian West <brian at freeswitch.org> wrote:

> Well based on your logs its clearly setup wrong... did you wipe the config?
> and start over and how are you doing multi-tenant?  Can you explain to me
> the steps you took to make a multi-tenant setup?
>
> /b
>
> On Apr 5, 2010, at 1:31 PM, Aloysius Lloyd wrote:
>
> >
> > Brian,
> >
> > I do not think so.
> >
> > If the config is wrong why the xlite is working for the Multitenat
> Extension. The same extension is registering and not able to make call . In
> another phone even not registering.
> >
> > Thanks
> > Lloyd
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dule.maillist at gmail.com  Mon Apr  5 15:16:37 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 5 Apr 2010 18:16:37 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <AC077181-A885-4F33-851A-0C12E2FE2FC8@freeswitch.org>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
	<s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
	<AC077181-A885-4F33-851A-0C12E2FE2FC8@freeswitch.org>
Message-ID: <v2v914fc92a1004051516g246cf7c0ncf1e2c1eb7b1b250@mail.gmail.com>

Oh okay, it's just from your previous comment, it sounded like you said the
opposite. I.e., if the other side offered 96, FS would listen on 96.

I don't actually have access to the box in question atm, I'm getting this
issue second hand. You wouldn't happen to know roughly when this was changed
(which you seem to be implying)? The box is definitely running a much older
version maybe 1.0.3 or 1.0.4 at the latest.

Dan



On Mon, Apr 5, 2010 at 5:58 PM, Brian West <brian at freeswitch.org> wrote:

> Yes thats how I read it... have you downloaded the latest freeswitch?
>
> /b
>
> On Apr 5, 2010, at 4:54 PM, Dan Le wrote:
>
> Sorry to dig up an old thread, but ran into this same problem, and trying
> to understand the RFC as what is actually the expected behaviour.
>
> So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams
> section:
>
> For sendrecv RTP
>> streams, the payload type numbers indicate the value of the payload
>> type field the offerer expects to receive, and would prefer to send
>>
>>
>>
> Doesn't that mean if they say 96 and FS says 101, they should be listening
> on 96 and FS should be listening on 101? Kinda like how ports work, you
> listen on the port that you offer.
>
> Am I completely misinterpreting?
>
> Thanks,
> Dan
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 15:20:37 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 17:20:37 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
	<A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
	<m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>
Message-ID: <5DA9DF5C-6F60-4CA5-A94D-8DF291BDF751@freeswitch.org>

Considering that their was only exactly ONE patch between 17155 to 17188 that involved sofia and didn't touch any of that code in sofia related to nat (add killgw _all_ to delete all gws 17167)  the rest were mod_skinny, mod_sangoma_codec, swigall, packaging and git related changes.

17187 was making sofia work on OpenBSD you could try to use 17186, And report... 

But looking over the changes between those two I don't see anything that obvious related to sofia and nat that could have broken.

/b


On Apr 5, 2010, at 5:08 PM, Aloysius Lloyd wrote:

> Brian,
> 
> Further to my previous off list email.
> 
> I can confirm now 100% something went wrong between 17155 to 17188. 
> 
> The same setup working on the following version
> 
> freeswitch at internal> version
> FreeSWITCH Version 1.0.trunk (17155)
> 
> But not working in the following version
> 
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (svn-17188)
> 
> Thanks
> Lloyd



From javieraristizabal at gmail.com  Mon Apr  5 16:08:38 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 5 Apr 2010 19:08:38 -0400
Subject: [Freeswitch-users] Question about incoming calls.
Message-ID: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>

Hi Folks.

I have an incoming TFN into my FreeSWITCH box, and i have two trunks.. Is it
possible to send the 50% of the calls to one trunk and the other 50% of the
calls to the other trunk?

Many thanks

-- 
-.-- --- / ... --- -.--

Javier Aristiz?bal
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From brian at freeswitch.org  Mon Apr  5 16:15:20 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 18:15:20 -0500
Subject: [Freeswitch-users] Question about incoming calls.
In-Reply-To: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
References: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
Message-ID: <071ADCCA-D6DA-41FB-83A9-C5753296A605@freeswitch.org>

.... .- ...- .   -.-- --- ..-   -.-. .... . -.-. -.- . -..   --- ..- -   -- --- -.. -.. .. ... - .-. .. -... ..- - --- .-. ..--.. 

-..-. -... 

On Apr 5, 2010, at 6:08 PM, Javier Aristiz?bal wrote:

> -- 
> -.-- --- / ... --- -.--

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From janvb at live.com  Sun Apr  4 12:43:27 2010
From: janvb at live.com (Jan Berger)
Date: Sun, 4 Apr 2010 21:43:27 +0200
Subject: [Freeswitch-users] Freeswitch for Windows - command history
	too	short?
In-Reply-To: <n2i7abab2411004040616w5a1eef47j392099111bdcc52c@mail.gmail.com>
References: <n2i7abab2411004040616w5a1eef47j392099111bdcc52c@mail.gmail.com>
Message-ID: <SNT135-w12547C763F8656D904D5F1D21A0@phx.gbl>


hi,

 

This is not FreeSWITCH, but sounds like your window console settings? 

 

Once you open a FreeSWITCH console left click on the system icon 8topmost left icon in window) and select options/preferences at the bottom of the pop-up menu - there you can alter the buffer length and window appearance.

 

Jan
 
> Date: Sun, 4 Apr 2010 14:16:34 +0100
> From: vfclists at googlemail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch for Windows - command history too short?
> 
> I am using svn 17048 for Windows and the command history is too short
> and doesn't persist between restarts.
> 
> Are there some configuration settings to fix that?
> 
> -- 
> Frank Church
> 
> =======================
> http://devblog.brahmancreations.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
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From uzoechi at yahoo.com  Mon Apr  5 08:00:38 2010
From: uzoechi at yahoo.com (Uzo Uzo)
Date: Mon, 5 Apr 2010 08:00:38 -0700 (PDT)
Subject: [Freeswitch-users] how do I get my DID number from lua script with
	session:getvariable
Message-ID: <714596.69133.qm@web32802.mail.mud.yahoo.com>

I probably am doing this wrong, but here is what I have.
I have a bunch of DID's.  I want my lua script to intercept all calls then
handle them based on which DID was called.

What I have done is create my lua script foo.lua

then in the dialplan/default.xml

I have
<extension name="mylua">
  <condition field="destination_number" "expression="^1002$">
  <action application="lua" data="/usr/local/freeswitch/scripts/foo.lua"/>
</condition>
</extension>

then i forward all my DIDs to extension 1002 at myfreeswitchbox

when I get a call to the DID the lua script intercepts it, is this the best way?

from the lua script,  I can do session:getVariable("caller_id_number")

but that gives me the caller id of the incoming number, which is cool.
which variable can i examine to get the DID number?

thanks.



      


From nazim.aghabayov at gmail.com  Mon Apr  5 13:51:21 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Tue, 06 Apr 2010 01:51:21 +0500
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
Message-ID: <4BBA4D49.5020704@gmail.com>

Hello Justin, have you tried to DNS lookup the cdr.mydomain.com for
serveral times to see if DNS round robin really works?

On 04/06/2010 12:47 AM, Justin B Newman wrote:
> I have mod_xml_cdr set with a url to POST entries to.
>
> The DNS record is something along the lines of:
>
> cdr.mydomain.com.		600	IN	CNAME	mydomain.com.
> mydomain.com.		600	IN	A	10.10.10.1
> mydomain.com.		600	IN	A	10.10.10.2
>
> Only one of the two web servers is receiving any POSTs.
>
> I tried adding "curl_easy_setopt(curl_handle,
> CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail.
>
> Any ideas?
>
> -jbn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From nazim.aghabayov at gmail.com  Mon Apr  5 14:01:13 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Tue, 06 Apr 2010 02:01:13 +0500
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
Message-ID: <4BBA4F99.102@gmail.com>

Most probably RR is not working for CNAME. You should really try:

cdr.mydomain.com.    IN  A    10.10.10.1
cdr.mydomain.com.    IN  A    10.10.10.2



On 04/06/2010 12:47 AM, Justin B Newman wrote:
> I have mod_xml_cdr set with a url to POST entries to.
>
> The DNS record is something along the lines of:
>
> cdr.mydomain.com.		600	IN	CNAME	mydomain.com.
> mydomain.com.		600	IN	A	10.10.10.1
> mydomain.com.		600	IN	A	10.10.10.2
>
> Only one of the two web servers is receiving any POSTs.
>
> I tried adding "curl_easy_setopt(curl_handle,
> CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail.
>
> Any ideas?
>
> -jbn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr  5 16:16:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Apr 2010 16:16:19 -0700
Subject: [Freeswitch-users] Question about incoming calls.
In-Reply-To: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
References: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
Message-ID: <o2z87f2f3b91004051616me5daf67bx374baab736d850cb@mail.gmail.com>

Javier,

Are your trunks from two separate carriers?
-MC

2010/4/5 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Hi Folks.
>
> I have an incoming TFN into my FreeSWITCH box, and i have two trunks.. Is
> it possible to send the 50% of the calls to one trunk and the other 50% of
> the calls to the other trunk?
>
> Many thanks
>
> --
> -.-- --- / ... --- -.--
>
> Javier Aristiz?bal
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From javieraristizabal at gmail.com  Mon Apr  5 16:28:12 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 5 Apr 2010 19:28:12 -0400
Subject: [Freeswitch-users] Question about incoming calls.
In-Reply-To: <o2z87f2f3b91004051616me5daf67bx374baab736d850cb@mail.gmail.com>
References: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
	<o2z87f2f3b91004051616me5daf67bx374baab736d850cb@mail.gmail.com>
Message-ID: <x2pb6f11f301004051628q6d2565eeub962aa5a53777853@mail.gmail.com>

Brian, i will check mod distributor, thanks ;)

Michael, yeah two diff carriers...

Many Thanks

On Mon, Apr 5, 2010 at 7:16 PM, Michael Collins <msc at freeswitch.org> wrote:

> Javier,
>
> Are your trunks from two separate carriers?
> -MC
>
> 2010/4/5 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Hi Folks.
>>
>> I have an incoming TFN into my FreeSWITCH box, and i have two trunks.. Is
>> it possible to send the 50% of the calls to one trunk and the other 50% of
>> the calls to the other trunk?
>>
>> Many thanks
>>
>> --
>> -.-- --- / ... --- -.--
>>
>> Javier Aristiz?bal
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 16:32:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Apr 2010 16:32:36 -0700
Subject: [Freeswitch-users] Question about incoming calls.
In-Reply-To: <x2pb6f11f301004051628q6d2565eeub962aa5a53777853@mail.gmail.com>
References: <k2zb6f11f301004051608z1341548bm2facafbb3bbd02c9@mail.gmail.com>
	<o2z87f2f3b91004051616me5daf67bx374baab736d850cb@mail.gmail.com>
	<x2pb6f11f301004051628q6d2565eeub962aa5a53777853@mail.gmail.com>
Message-ID: <t2r87f2f3b91004051632u3f24fad5if11b70839b152a6a@mail.gmail.com>

2010/4/5 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Brian, i will check mod distributor, thanks ;)
>
> Michael, yeah two diff carriers...
>
> Many Thanks
>

Yeah I'm guessing that this is a carrier challenge. mod_distributor will let
you do fun things once the call reaches your switch, but the carrier is who
sends you the call in the first place.
-MC
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From david.ponzone at gmail.com  Mon Apr  5 16:58:03 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 6 Apr 2010 01:58:03 +0200
Subject: [Freeswitch-users] how do I get my DID number from lua script
	with session:getvariable
In-Reply-To: <714596.69133.qm@web32802.mail.mud.yahoo.com>
References: <714596.69133.qm@web32802.mail.mud.yahoo.com>
Message-ID: <0242C963-51BC-466B-ADC1-5846B869B60A@gmail.com>

try rdnis

But you could put your lua script in your public context, where the  
calls are handled first.
I dont really see the point to do that in default.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 05/04/2010 ? 17:00, Uzo Uzo a ?crit :

> I probably am doing this wrong, but here is what I have.
> I have a bunch of DID's.  I want my lua script to intercept all  
> calls then
> handle them based on which DID was called.
>
> What I have done is create my lua script foo.lua
>
> then in the dialplan/default.xml
>
> I have
> <extension name="mylua">
>  <condition field="destination_number" "expression="^1002$">
>  <action application="lua" data="/usr/local/freeswitch/scripts/ 
> foo.lua"/>
> </condition>
> </extension>
>
> then i forward all my DIDs to extension 1002 at myfreeswitchbox
>
> when I get a call to the DID the lua script intercepts it, is this  
> the best way?
>
> from the lua script,  I can do session:getVariable("caller_id_number")
>
> but that gives me the caller id of the incoming number, which is cool.
> which variable can i examine to get the DID number?
>
> 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 rob4manhere at gmail.com  Mon Apr  5 17:47:30 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 5 Apr 2010 19:47:30 -0500
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <4BBA4F99.102@gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
	<4BBA4F99.102@gmail.com>
Message-ID: <s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>

Sounds like something more appropriate for a load balancer, such as lvs (
http://www.linuxvirtualserver.org/).

DNS round robin also can't detect failure and stopping routing requests to
the node, while a tool like lvs could.


On Mon, Apr 5, 2010 at 4:01 PM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

> Most probably RR is not working for CNAME. You should really try:
>
> cdr.mydomain.com.    IN  A    10.10.10.1
> cdr.mydomain.com.    IN  A    10.10.10.2
>
>
>
> On 04/06/2010 12:47 AM, Justin B Newman wrote:
> > I have mod_xml_cdr set with a url to POST entries to.
> >
> > The DNS record is something along the lines of:
> >
> > cdr.mydomain.com.             600     IN      CNAME   mydomain.com.
> > mydomain.com.         600     IN      A       10.10.10.1
> > mydomain.com.         600     IN      A       10.10.10.2
> >
> > Only one of the two web servers is receiving any POSTs.
> >
> > I tried adding "curl_easy_setopt(curl_handle,
> > CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail.
> >
> > Any ideas?
> >
> > -jbn
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Mon Apr  5 18:06:20 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 6 Apr 2010 09:06:20 +0800
Subject: [Freeswitch-users] how do I get my DID number from lua script
	with session:getvariable
In-Reply-To: <714596.69133.qm@web32802.mail.mud.yahoo.com>
References: <714596.69133.qm@web32802.mail.mud.yahoo.com>
Message-ID: <j2g7d0bfd8c1004051806leb245316x6d8a2aba02769aa7@mail.gmail.com>

you can pass the caller_id_number parameter to the Lua script e.g.
<action application="lua" data="/usr/local/freeswitch/scripts/foo.lua
${caller_id_number}"/>
-nandy


On Mon, Apr 5, 2010 at 11:00 PM, Uzo Uzo <uzoechi at yahoo.com> wrote:

> I probably am doing this wrong, but here is what I have.
> I have a bunch of DID's.  I want my lua script to intercept all calls then
> handle them based on which DID was called.
>
> What I have done is create my lua script foo.lua
>
> then in the dialplan/default.xml
>
> I have
> <extension name="mylua">
>  <condition field="destination_number" "expression="^1002$">
>  <action application="lua" data="/usr/local/freeswitch/scripts/foo.lua"/>
> </condition>
> </extension>
>
> then i forward all my DIDs to extension 1002 at myfreeswitchbox
>
> when I get a call to the DID the lua script intercepts it, is this the best
> way?
>
> from the lua script,  I can do session:getVariable("caller_id_number")
>
> but that gives me the caller id of the incoming number, which is cool.
> which variable can i examine to get the DID number?
>
> 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 steveu at coppice.org  Mon Apr  5 18:16:23 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 06 Apr 2010 09:16:23 +0800
Subject: [Freeswitch-users] How to configure g729 pass though
	on	Freeswitch
In-Reply-To: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>	<201004050654.11705.sos@sokhapkin.dyndns.org>	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
Message-ID: <4BBA8B67.70505@coppice.org>

Hi Frank,

On 04/06/2010 01:00 AM, Frank Church wrote:
> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
> were completely garbled, even the ringing. It could be affecting
> something on the provider end.
>
> Could it be that the provider has a different G729 codec that is not
> compatible with the actual G729a the Linksys is sending?
>
>    
G.729 and G.729A are identical on the wire, which is why there is only a 
single SDP code for them. The only time you would ever see an 
incompatible G.729 is if you are working with an ancient Cisco. They had 
a form of G.729 at one time that packed all the bits in the opposite 
order. Its rare to see that these days, though. Its more likely 
something strange happens with the ptime.

Steve



From bcxml at hotmail.com  Mon Apr  5 18:49:26 2010
From: bcxml at hotmail.com (bcxml)
Date: Mon, 5 Apr 2010 18:49:26 -0700 (PDT)
Subject: [Freeswitch-users]  Problem with IVR app starting too soon
Message-ID: <28146712.post@talk.nabble.com>


I have Freeswitch setup to pass incomming calls off to Microsoft Speech
Server 2007. Everything works fine except for one thing.  When the user
calls, they hear ringing, the call is answered, then they hear the Speech
Server app welcome message, which may or may not be complete.

For example, the application is suppoose to say "Welcome to the IVR
application" at the beginning. The caller may hear the entire prompt or they
may only hear  "to the IVR Application" or perhaps "IVR Application"

Can anyone point me in the right direction ?


Thanks

Brian Campbell
-- 
View this message in context: http://old.nabble.com/Problem-with-IVR-app-starting-too-soon-tp28146712p28146712.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From 12ukwn at gmail.com  Mon Apr  5 18:52:09 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 6 Apr 2010 03:52:09 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
Message-ID: <20100406035209.6e5444d0@anubis.defcon1>

Le Mon, 5 Apr 2010 11:44:06 -0500,
"Russell Mosemann" <Russell.Mosemann at cune.org> a ?crit :

Hehe, I spoke too fast and without even testing after deleting the
cepstral libs symlinks; as a matter of fact TTS don't work anymore
an FS complains about missing cepstral libs...
Here's the trace:

2010-04-06 03:47:29.244980 [NOTICE] mod_sofia.c:1907 Pre-Answer
sofia/internal/1000 at 192.168.1.50! Failed to load library libceplang_en.so
due to: libceplang_en.so: cannot open shared object file: No such file or
directory Failed to load library libceplex_us.so due to: libceplex_us.so:
cannot open shared object file: No such file or directory Failed to load
language / lexical libraries for William Failed to load library
libceplang_en.so due to: libceplang_en.so: cannot open shared object file:
No such file or directory Failed to load library libceplex_us.so due to:
libceplex_us.so: cannot open shared object file: No such file or directory
Failed to load language / lexical libraries for Callie 2010-04-06
03:48:05.457410 [NOTICE] sofia.c:4789 Hangup
sofia/internal/1000 at 192.168.1.50 [CS_EXECUTE] [ORIGINATOR_CANCEL]

I would say that FS is only looking for any lib in its path and not into
the system.

-- 
What happens when you cut back the jungle?  It recedes.


From brian at freeswitch.org  Mon Apr  5 18:59:53 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 20:59:53 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100406035209.6e5444d0@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
	<20100406035209.6e5444d0@anubis.defcon1>
Message-ID: <700C46B8-F74A-429D-AEA0-90112076817C@freeswitch.org>

That would be a wrong assumption.  Do you have the correct versions?

/b

On Apr 5, 2010, at 8:52 PM, Jean-Yves F. Barbier wrote:

> I would say that FS is only looking for any lib in its path and not into
> the system.



From 12ukwn at gmail.com  Mon Apr  5 19:06:06 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 6 Apr 2010 04:06:06 +0200
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <28146712.post@talk.nabble.com>
References: <28146712.post@talk.nabble.com>
Message-ID: <20100406040606.78daf9fd@anubis.defcon1>

Le Mon, 5 Apr 2010 18:49:26 -0700 (PDT),
bcxml <bcxml at hotmail.com> a ?crit :

> 
> I have Freeswitch setup to pass incomming calls off to Microsoft Speech
> Server 2007. Everything works fine except for one thing.  When the user
> calls, they hear ringing, the call is answered, then they hear the Speech
> Server app welcome message, which may or may not be complete.
> 
> For example, the application is suppoose to say "Welcome to the IVR
> application" at the beginning. The caller may hear the entire prompt or
> they may only hear  "to the IVR Application" or perhaps "IVR Application"
> 
> Can anyone point me in the right direction ?
Yes: turn left :D

As it is done into mod_cepstral, insert a 500ms silence before
(and possibly after) the speech takes place.

-- 
  Our comedies are not to be laughed at. -Samuel Goldwyn


From 12ukwn at gmail.com  Mon Apr  5 19:09:17 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 6 Apr 2010 04:09:17 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
Message-ID: <20100406040917.40f7d6ef@anubis.defcon1>

Le Mon, 5 Apr 2010 11:44:06 -0500,
"Russell Mosemann" <Russell.Mosemann at cune.org> a ?crit :

Oops: I also forgot to say I re-symlinked the libs without restarting FS
that silently failed to answer any further call (I'd rather have it dead:(

-- 
Down with categorical imperative!


From 12ukwn at gmail.com  Mon Apr  5 19:12:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 6 Apr 2010 04:12:28 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <700C46B8-F74A-429D-AEA0-90112076817C@freeswitch.org>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
	<20100406035209.6e5444d0@anubis.defcon1>
	<700C46B8-F74A-429D-AEA0-90112076817C@freeswitch.org>
Message-ID: <20100406041228.0aeb7038@anubis.defcon1>

Le Mon, 5 Apr 2010 20:59:53 -0500,
Brian West <brian at freeswitch.org> a ?crit :

> That would be a wrong assumption.  Do you have the correct versions?

What versions of what are you talking about?

if it is about libs, FS was compiled on this computer.


-- 
A woman takes off her claim to respect along with her garments.
		-- Herodotus


From brian at freeswitch.org  Mon Apr  5 19:16:40 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 21:16:40 -0500
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <20100406040606.78daf9fd@anubis.defcon1>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
Message-ID: <ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>

playback silence_stream://1000

/b

On Apr 5, 2010, at 9:06 PM, Jean-Yves F. Barbier wrote:

> Yes: turn left :D
> 
> As it is done into mod_cepstral, insert a 500ms silence before
> (and possibly after) the speech takes place.



From jeff at jefflenk.com  Mon Apr  5 19:30:52 2010
From: jeff at jefflenk.com (Jeff Lenk )
Date: Tue, 6 Apr 2010 02:30:52 +0000
Subject: [Freeswitch-users] Freeswitch for Windows - command history
	too	short?
Message-ID: <COL117-DS1227CF2742AB2003A94E87B9180@phx.gbl>

FS uses its own history logic for its console which are fixed length at this time.


-----Original Message-----
From: Jan Berger
Sent: 4/4/2010 7:43:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch for Windows - command history too short?
hi,

This is not FreeSWITCH, but sounds like your window console settings?

Once you open a FreeSWITCH console left click on the system icon 8topmost left icon in window) and select options/preferences at the bottom of the pop-up menu - there you can alter the buffer length and window appearance.

Jan

> Date: Sun, 4 Apr 2010 14:16:34 +0100
> From: vfclists at googlemail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch for Windows - command history too short?
>
> I am using svn 17048 for Windows and the command history is too short
> and doesn't persist between restarts.
>
> Are there some configuration settings to fix that?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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

________________________________
Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
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From hungngm at bkav.com.vn  Mon Apr  5 20:56:52 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Tue, 06 Apr 2010 10:56:52 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <87E806E5FECF0D28C32C44D7B173658F789757CA@hungngm>



 

Hi Anthony.

What is version of FS has this feature of mod_fifo. I have update the latest
version in svn: FreeSWITCH Version 1.0.trunk (16972M).

I capture packets in an agent of mod_fifo with Wireshark, but i can't see any
SIP UPDATE or SIP INFO packet.

Best Regards.

 

Anthony Minessale [anthony.minessale at gmail.com]

its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when the
call is bridged.
X-lite does not update anything when it receives them.

That's about it.

 


 

     

    Hi Anthony,

    

    Can you discuss some details in how polycom or snom can do this and x-lite
    not.

    

    If can, I want to edit some open source soffphone like officeSIP to do this.
   

    

    Best Regards.

    

     

    Anthony Minessale [ anthony.minessale at gmail.com ]

    

    We already do it.
    X-Lite does not support it.
    If you try it with a phone like snom or polycom you will see it works just
    like that.


     


     

         

        Hi Seven Du.

        

        Thanks to yours suggetion.

        

        I have an ideal, it is: when the call between caller and agent is set,
        the caller_id is determined. So, i want to edit code to sent the agent
        information (the call_id and call_id_number) which will be displayed
        againt in the agent's softphone (as Xlite..) when the call is happening.
       

        

        I read some documents but i still can't determine: It's maybe yes or
        maybe to do this and where to do this.

        

        Can you give me some comments.

        

        Best Regard.

        

         

        Seven Du [ dujinfang at gmail.com ]

        

        

        ??As discussed in the list, it's not a freeswitch problem but a
        reality of life. 



        Think about customer A and B calls in one after another, then if
        FreeSWITCH call agent X with caller id A and Y with caller id B, and
        angent Y answers before X, then

        1) if bridge Y with A with the FIFO rule, then the caller id is wrong
        2) if bridge Y with B, the caller id is right but it breaks the rule
        of FIFO - A should be served before B!! And what even worse is that
        if X never answer A then A never can be served which is really
        unfair!!

        Of course you don't want 1), and you don't need mod_fifo if you want
        behavior 2), you just need some dialplan trick or some simple Lua
        script I think. Also FreeSWITCH is designed to be easily extended with
        almost any languages so feel free to implement anything.


        > Hi Mike and Seven Du.
        > Thanks to yours help.
        > I known the mechanism of mod_fifo.
        >>>
       
       
       
       
       
       
       
       
        http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
        .
        > What a pity, It can't solve this problem. I can't use freeswitch for
        my call
        > center.
        > Hope new version can solve this !!!


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From brian at freeswitch.org  Mon Apr  5 21:03:17 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 23:03:17 -0500
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <87E806E5FECF0D28C32C44D7B173658F789757CA@hungngm>
References: <87E806E5FECF0D28C32C44D7B173658F789757CA@hungngm>
Message-ID: <4B1B9DBF-9819-4711-9261-6E9E283446BD@freeswitch.org>

x-lite doesn't support updates in any way.

/b

On Apr 5, 2010, at 10:56 PM, Nguy?n M?nh H?ng wrote:

> X-lite does not update anything when it receives them



From lakindia89 at gmail.com  Mon Apr  5 21:24:51 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 6 Apr 2010 09:54:51 +0530
Subject: [Freeswitch-users] How to setup a span as PRI_NET
Message-ID: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>

Hi all,
In my office we have a Hard PBX, with some 4 extensions.
We also have sangoma A102 card.
>From the Hard PBX, if 0 is pressed, it is setup in a way that it will go to
outside world.
I've connected that line to span2 of the card.
The span2 in the A102 card, is configured as PRI_NET.

wanrouter status shows connected for the span2.

But if I dial from the extension, I got the following in the sangoma_dchan
log.
2010-04-03 12:49:49
INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 33 39
31 31 34 36 30 30 7d 02 91 81 ]
  Call Ref:0164
  Type:Setup (0x5)
  Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
Kbit/s(16) L1Prot:G.711 A-Law(3)
  Calling Party Number:4439114600(l:10) plan:isdn(1)
type:unknown(0)scr:user, passed(1) pres:allowed(0)
  High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]

2010-04-03 12:49:49
OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
  Call Ref:0164
  Type:Release Compl (0x5a)
  Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
Circuit/Channel Available(34)

The call in not reaching freeswitch ( I enabled debug log. But nothing is
printing in it ).
Can someone suggest how to make this work.

Please ask me if you need more information?, since I don't know what to give
now.
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From a.afzali2003 at gmail.com  Mon Apr  5 22:28:58 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 6 Apr 2010 09:58:58 +0430
Subject: [Freeswitch-users] Stuck at revision 17188
Message-ID: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>

Hi,
 After some issues about svn update in openzap and compile, my FreeSWITCH
version stuck at revision 17188 ! I'm in trouble?

-- afshin
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From jason at jasonjgw.net  Mon Apr  5 22:40:22 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 6 Apr 2010 15:40:22 +1000
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
Message-ID: <20100406054022.GA21075@jdc.jasonjgw.net>

afshin afzali <a.afzali2003 at gmail.com> wrote:
>  After some issues about svn update in openzap and compile, my FreeSWITCH
> version stuck at revision 17188 ! I'm in trouble?

Do a fresh checkout and try building from there.



From brian at freeswitch.org  Mon Apr  5 22:42:17 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 00:42:17 -0500
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
Message-ID: <40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>

well 17188 is the current rev.. what are you having issues with...

/b

On Apr 6, 2010, at 12:28 AM, afshin afzali wrote:

> Hi,
>  After some issues about svn update in openzap and compile, my FreeSWITCH version stuck at revision 17188 ! I'm in trouble?
>  
> -- afshin



From oseslija at gmail.com  Tue Apr  6 00:10:17 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 6 Apr 2010 09:10:17 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
Message-ID: <r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>

There isn't one. Both call legs need to agree on G729. Try to set only G729
in inbound/outbound codec-prefs in sip profile or use disable-transcoding.

Since  this is becoming a neverending saga, join us on IRC for help.

On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists at googlemail.com>wrote:

> I have changed it and this time I get this error
>
> 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
> usable in passthrough mode!
> 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder
> error!
>
> When it is on 0.030 I get the garbled sound.
>
> I guess the 0.020 is more compatible and I now have to set the pass
> through mode. Is there a parameter for that?
>
>
> On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
> > change it to 0.020
> >
> > /b
> >
> > On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
> >
> >> I can't see a ptime in the Linksys settings, is that what is labelled
> >> as 'RTP Packet Size' on the SIP page?
> >>
> >> It is the only thing that has the value 0.030
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 vfclists at googlemail.com  Tue Apr  6 00:42:35 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 6 Apr 2010 08:42:35 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
	<r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
Message-ID: <r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>

Thanks to the help of every one in this thread I have managed to get it
working. The important thing I realized is it is necessary to stop
Freeswitch whenever you change the settings and restart it again. I thought
doing a reloadxml would get the changes to take, but that is not the case.

I have documented my findings on my
<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura>
devblog<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura>

On 6 April 2010 08:10, Ognjen Seslija <oseslija at gmail.com> wrote:
> There isn't one. Both call legs need to agree on G729. Try to set only
G729
> in inbound/outbound codec-prefs in sip profile or use disable-transcoding.
>
> Since  this is becoming a neverending saga, join us on IRC for help.
>

I have got to start using IRC more often, I have never really gotten into
it, but I see that it is necessary when time is important. I prefer mailing
lists though as the keep a better record of what has transpired for all to
see.

Thanks to all.

My next question will be coming up shortly, this time on IRC first.
> On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists at googlemail.com>
> wrote:
>>
>> I have changed it and this time I get this error
>>
>> 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
>> usable in passthrough mode!
>> 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder
>> error!
>>
>> When it is on 0.030 I get the garbled sound.
>>
>> I guess the 0.020 is more compatible and I now have to set the pass
>> through mode. Is there a parameter for that?
>>
>>
>> On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
>> > change it to 0.020
>> >
>> > /b
>> >
>> > On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
>> >
>> >> I can't see a ptime in the Linksys settings, is that what is labelled
>> >> as 'RTP Packet Size' on the SIP page?
>> >>
>> >> It is the only thing that has the value 0.030
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From jason at jasonjgw.net  Tue Apr  6 01:06:16 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 6 Apr 2010 18:06:16 +1000
Subject: [Freeswitch-users] How to configure g729 pass though on
 Freeswitch
In-Reply-To: <r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
References: <31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
	<r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
	<r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
Message-ID: <20100406080616.GA26122@jdc.jasonjgw.net>

Frank Church <vfclists at googlemail.com> wrote:
> Thanks to the help of every one in this thread I have managed to get it
> working. The important thing I realized is it is necessary to stop
> Freeswitch whenever you change the settings and restart it again. I thought
> doing a reloadxml would get the changes to take, but that is not the case.

sofia profile <profile-name> rescan reloadxml
should suffice for a codec change.



From david.ponzone at gmail.com  Tue Apr  6 01:06:22 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 6 Apr 2010 10:06:22 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
	<r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
	<r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
Message-ID: <C509F7B5-B9B3-4FF7-A0DE-E9CD5FA3BA4D@gmail.com>

Frank,

Restarting FS should not be necessary.

If you change a SIP profile xml, or some parameters affecting it (some  
variables in vars.xml for example), you have to restart the profile, I  
think I told you that before:
sofia profile internal restart reloadxml

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 06/04/2010 ? 09:42, Frank Church a ?crit :

> Thanks to the help of every one in this thread I have managed to get  
> it working. The important thing I realized is it is necessary to  
> stop Freeswitch whenever you change the settings and restart it  
> again. I thought doing a reloadxml would get the changes to take,  
> but that is not the case.
>
> I have documented my findings on my devblog
>
> On 6 April 2010 08:10, Ognjen Seslija <oseslija at gmail.com> wrote:
> > There isn't one. Both call legs need to agree on G729. Try to set  
> only G729
> > in inbound/outbound codec-prefs in sip profile or use disable- 
> transcoding.
> >
> > Since  this is becoming a neverending saga, join us on IRC for help.
> >
>
> I have got to start using IRC more often, I have never really gotten  
> into it, but I see that it is necessary when time is important. I  
> prefer mailing lists though as the keep a better record of what has  
> transpired for all to see.
>
> Thanks to all.
>
> My next question will be coming up shortly, this time on IRC first.
> > On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists at googlemail.com 
> >
> > wrote:
> >>
> >> I have changed it and this time I get this error
> >>
> >> 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
> >> usable in passthrough mode!
> >> 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729  
> decoder
> >> error!
> >>
> >> When it is on 0.030 I get the garbled sound.
> >>
> >> I guess the 0.020 is more compatible and I now have to set the pass
> >> through mode. Is there a parameter for that?
> >>
> >>
> >> On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
> >> > change it to 0.020
> >> >
> >> > /b
> >> >
> >> > On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
> >> >
> >> >> I can't see a ptime in the Linksys settings, is that what is  
> labelled
> >> >> as 'RTP Packet Size' on the SIP page?
> >> >>
> >> >> It is the only thing that has the value 0.030
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Frank Church
> >>
> >> =======================
> >> http://devblog.brahmancreations.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
> >
> >
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 vfclists at googlemail.com  Tue Apr  6 01:29:43 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 6 Apr 2010 09:29:43 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <C509F7B5-B9B3-4FF7-A0DE-E9CD5FA3BA4D@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
	<r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
	<r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
	<C509F7B5-B9B3-4FF7-A0DE-E9CD5FA3BA4D@gmail.com>
Message-ID: <m2g7abab2411004060129y66dfce23s2e62ff5a8adcf7c1@mail.gmail.com>

What does reloadxml do on its own?

I thought it was equivalent it executed a restart as well.

On 6 April 2010 09:06, David Ponzone <david.ponzone at gmail.com> wrote:

> Frank,
>
> Restarting FS should not be necessary.
>
> If you change a SIP profile xml, or some parameters affecting it (some
> variables in vars.xml for example), you have to restart the profile, I think
> I told you that before:
> sofia profile internal restart reloadxml
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 06/04/2010 ? 09:42, Frank Church a ?crit :
>
> Thanks to the help of every one in this thread I have managed to get it
> working. The important thing I realized is it is necessary to stop
> Freeswitch whenever you change the settings and restart it again. I thought
> doing a reloadxml would get the changes to take, but that is not the case.
>
> I have documented my findings on my
> <http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura>
> devblog<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura>
>
> On 6 April 2010 08:10, Ognjen Seslija <oseslija at gmail.com> wrote:
> > There isn't one. Both call legs need to agree on G729. Try to set only
> G729
> > in inbound/outbound codec-prefs in sip profile or use
> disable-transcoding.
> >
> > Since  this is becoming a neverending saga, join us on IRC for help.
> >
>
> I have got to start using IRC more often, I have never really gotten into
> it, but I see that it is necessary when time is important. I prefer mailing
> lists though as the keep a better record of what has transpired for all to
> see.
>
> Thanks to all.
>
> My next question will be coming up shortly, this time on IRC first.
> > On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists at googlemail.com>
> > wrote:
> >>
> >> I have changed it and this time I get this error
> >>
> >> 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
> >> usable in passthrough mode!
> >> 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729
> decoder
> >> error!
> >>
> >> When it is on 0.030 I get the garbled sound.
> >>
> >> I guess the 0.020 is more compatible and I now have to set the pass
> >> through mode. Is there a parameter for that?
> >>
> >>
> >> On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
> >> > change it to 0.020
> >> >
> >> > /b
> >> >
> >> > On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
> >> >
> >> >> I can't see a ptime in the Linksys settings, is that what is labelled
> >> >> as 'RTP Packet Size' on the SIP page?
> >> >>
> >> >> It is the only thing that has the value 0.030
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Frank Church
> >>
> >> =======================
> >> http://devblog.brahmancreations.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
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From david.ponzone at gmail.com  Tue Apr  6 01:51:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 6 Apr 2010 10:51:09 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <m2g7abab2411004060129y66dfce23s2e62ff5a8adcf7c1@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
	<k2g4468a6771004051129scdb49373o3af42cfb1bca9cb8@mail.gmail.com>
	<x2j7abab2411004051343tab729810t61bcc29afb2a2aa5@mail.gmail.com>
	<34BDA635-27BD-4E15-B34E-0DFC4267CE2B@freeswitch.org>
	<g2j7abab2411004051402q30c6f253k9f3c07a8ff234927@mail.gmail.com>
	<r2y4468a6771004060010l19cde0f3w6f917ceda71f1f48@mail.gmail.com>
	<r2k7abab2411004060042j3e21d0b2j8394f50e4030b3a1@mail.gmail.com>
	<C509F7B5-B9B3-4FF7-A0DE-E9CD5FA3BA4D@gmail.com>
	<m2g7abab2411004060129y66dfce23s2e62ff5a8adcf7c1@mail.gmail.com>
Message-ID: <DAF2EE86-74E4-4C81-BC7D-E21B46A778DA@gmail.com>

reloadxml reads again all the XML tree into memory (and you can see  
this tree in freeswitch.xml.fsxml).
That is enough for the XML dialplan or for the user directory, because  
the XML tree is parsed from memory when an INVITE or REGISTER hits FS.

But SIP profiles are not "dynamic".
If you change some parameters, you have to restart the profile using  
the new XML tree.
You may think of a SIP profile as a kind of sub-FS instance.

restart will do a complete stop/start on the profile (stopping  
existing calls)
rescan is softer and re-configure the profile without restarting it

I suspect there are situations where rescan is not enough (perhaps  
when you change the IP/port the profile is bound to).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 06/04/2010 ? 10:29, Frank Church a ?crit :

> What does reloadxml do on its own?
>
> I thought it was equivalent it executed a restart as well.
>
> On 6 April 2010 09:06, David Ponzone <david.ponzone at gmail.com> wrote:
> Frank,
>
> Restarting FS should not be necessary.
>
> If you change a SIP profile xml, or some parameters affecting it  
> (some variables in vars.xml for example), you have to restart the  
> profile, I think I told you that before:
> sofia profile internal restart reloadxml
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 06/04/2010 ? 09:42, Frank Church a ?crit :
>
>> Thanks to the help of every one in this thread I have managed to  
>> get it working. The important thing I realized is it is necessary  
>> to stop Freeswitch whenever you change the settings and restart it  
>> again. I thought doing a reloadxml would get the changes to take,  
>> but that is not the case.
>>
>> I have documented my findings on my devblog
>>
>> On 6 April 2010 08:10, Ognjen Seslija <oseslija at gmail.com> wrote:
>> > There isn't one. Both call legs need to agree on G729. Try to set  
>> only G729
>> > in inbound/outbound codec-prefs in sip profile or use disable- 
>> transcoding.
>> >
>> > Since  this is becoming a neverending saga, join us on IRC for  
>> help.
>> >
>>
>> I have got to start using IRC more often, I have never really  
>> gotten into it, but I see that it is necessary when time is  
>> important. I prefer mailing lists though as the keep a better  
>> record of what has transpired for all to see.
>>
>> Thanks to all.
>>
>> My next question will be coming up shortly, this time on IRC first.
>> > On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists at googlemail.com 
>> >
>> > wrote:
>> >>
>> >> I have changed it and this time I get this error
>> >>
>> >> 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only
>> >> usable in passthrough mode!
>> >> 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G. 
>> 729 decoder
>> >> error!
>> >>
>> >> When it is on 0.030 I get the garbled sound.
>> >>
>> >> I guess the 0.020 is more compatible and I now have to set the  
>> pass
>> >> through mode. Is there a parameter for that?
>> >>
>> >>
>> >> On 5 April 2010 21:47, Brian West <brian at freeswitch.org> wrote:
>> >> > change it to 0.020
>> >> >
>> >> > /b
>> >> >
>> >> > On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
>> >> >
>> >> >> I can't see a ptime in the Linksys settings, is that what is  
>> labelled
>> >> >> as 'RTP Packet Size' on the SIP page?
>> >> >>
>> >> >> It is the only thing that has the value 0.030
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Frank Church
>> >>
>> >> =======================
>> >> http://devblog.brahmancreations.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
>> >
>> >
>>
>>
>>
>> -- 
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 velu.technical at gmail.com  Tue Apr  6 02:44:02 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Tue, 6 Apr 2010 15:14:02 +0530
Subject: [Freeswitch-users] Help to call a VXML SIP number
Message-ID: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>

Dear All,
     I have received the XXX at sip.voxeo.net from my VXML application. I have
written the following dial plan to call that number.
  <extension name="VXML">
        <condition field="destination_number" expression="^(999)$">
                <action application="bridge" data="sofia/external/
9996079973 at sip.voxeo.net"/>
                <action application="hangup"/>
        </condition>
  </extension>

I have registered the FreeSWITCH and called the 999 extension. But the call
was just trying. Finally I have got the NO_ANSWER information in console.
  2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: NO_ANSWER

Is there configuration I need to do to call this number?? Please help me??
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From a.afzali2003 at gmail.com  Tue Apr  6 03:07:19 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 6 Apr 2010 14:37:19 +0430
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
	<40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>
Message-ID: <y2jdebff5a81004060307pf81f5092z73406446c3cbf7f1@mail.gmail.com>

No, I don't have any issue with it, I just saw the revision does not change
for some days just like previous ones :)

Regards,


On Tue, Apr 6, 2010 at 10:12 AM, Brian West <brian at freeswitch.org> wrote:

> well 17188 is the current rev.. what are you having issues with...
>
> /b
>
> On Apr 6, 2010, at 12:28 AM, afshin afzali wrote:
>
> > Hi,
> >  After some issues about svn update in openzap and compile, my FreeSWITCH
> version stuck at revision 17188 ! I'm in trouble?
> >
> > -- afshin
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim at k4gvo.com  Tue Apr  6 03:57:00 2010
From: jim at k4gvo.com (Jim)
Date: Tue, 06 Apr 2010 06:57:00 -0400
Subject: [Freeswitch-users] What is the 99999_enum.xml file?
Message-ID: <4BBB137C.7010604@k4gvo.com>

It seems to be preventing FS from parsing any files beyond it.  FS 
parses conf/dialplan/default.xml then each of the files in 
conf/dialplan/default/, but never sees any of the other files in 
conf/dialplan such as public.xml and the files in public/.

Thanks,
Jim.


From jason at jasonjgw.net  Tue Apr  6 04:15:41 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 6 Apr 2010 21:15:41 +1000
Subject: [Freeswitch-users] What is the 99999_enum.xml file?
In-Reply-To: <4BBB137C.7010604@k4gvo.com>
References: <4BBB137C.7010604@k4gvo.com>
Message-ID: <20100406111541.GA27336@jdc.jasonjgw.net>

Jim <jim at k4gvo.com> wrote:
> It seems to be preventing FS from parsing any files beyond it.  

No, it doesn't - have a look at log/freeswitch.xml.fsxml.

This file is a default extension to perform Enum look-ups. You can delete it
or rename it if you wish. If you leave it there, you have to make sure that
your extensions appear earlier in the fsxml file than this Enum extension
does. The order is determined by the expansion of the *.xml glob expression.




From jcasale at activenetwerx.com  Tue Apr  6 04:16:24 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 6 Apr 2010 11:16:24 +0000
Subject: [Freeswitch-users] What is the 99999_enum.xml file?
In-Reply-To: <4BBB137C.7010604@k4gvo.com>
References: <4BBB137C.7010604@k4gvo.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E99238F4@prato.activenetwerx.local>

>It seems to be preventing FS from parsing any files beyond it.  FS 
>parses conf/dialplan/default.xml then each of the files in 
>conf/dialplan/default/, but never sees any of the other files in 
>conf/dialplan such as public.xml and the files in public/.

Enum should be the last dialplan you get to if nothing matches, prolly
if you hit it, you aren't writing your dialplan properly or you overlooked
something.

http://en.wikipedia.org/wiki/Telephone_Number_Mapping

You are looking at the 'Default' context which is for auth'ed ua's.
The dialplan only encounters the 'Public' context for unauth'ed ua's,
which get placed in there first then hopefully route somewhere else,
the default context out of the box at least will never look into the
'Public' context.


From nazim.aghabayov at gmail.com  Tue Apr  6 04:39:54 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Tue, 06 Apr 2010 16:39:54 +0500
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>	<4BBA4F99.102@gmail.com>
	<s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>
Message-ID: <4BBB1D8A.6090603@gmail.com>

Agree, load balancer is better. It seems that libcurl doesn't respect
DNS RR.

On 04/06/2010 05:47 AM, Rob Forman wrote:
> Sounds like something more appropriate for a load balancer, such as
> lvs (http://www.linuxvirtualserver.org/).
>
> DNS round robin also can't detect failure and stopping routing
> requests to the node, while a tool like lvs could.
>
>
> On Mon, Apr 5, 2010 at 4:01 PM, Nazim Aghabayov
> <nazim.aghabayov at gmail.com <mailto:nazim.aghabayov at gmail.com>> wrote:
>
>     Most probably RR is not working for CNAME. You should really try:
>
>     cdr.mydomain.com <http://cdr.mydomain.com>.    IN  A    10.10.10.1
>     cdr.mydomain.com <http://cdr.mydomain.com>.    IN  A    10.10.10.2
>
>
>
>     On 04/06/2010 12:47 AM, Justin B Newman wrote:
>     > I have mod_xml_cdr set with a url to POST entries to.
>     >
>     > The DNS record is something along the lines of:
>     >
>     > cdr.mydomain.com <http://cdr.mydomain.com>.             600    
>     IN      CNAME   mydomain.com <http://mydomain.com>.
>     > mydomain.com <http://mydomain.com>.         600     IN      A  
>         10.10.10.1
>     > mydomain.com <http://mydomain.com>.         600     IN      A  
>         10.10.10.2
>     >
>     > Only one of the two web servers is receiving any POSTs.
>     >
>     > I tried adding "curl_easy_setopt(curl_handle,
>     > CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail.
>     >
>     > Any ideas?
>     >
>     > -jbn
>     >
>     > _______________________________________________
>     > 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 mrene_lists at avgs.ca  Tue Apr  6 04:54:54 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 6 Apr 2010 07:54:54 -0400
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100406041228.0aeb7038@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
	<20100406035209.6e5444d0@anubis.defcon1>
	<700C46B8-F74A-429D-AEA0-90112076817C@freeswitch.org>
	<20100406041228.0aeb7038@anubis.defcon1>
Message-ID: <05D1C7D3-226A-4DCC-AC1B-C256BC058F0F@avgs.ca>

Hi,

ldconfig -v will list you all libraries that are in ld's path, start  
by confirming the correct version of the lib appears in there. You may  
also chec: ldd /path/to/mod_cepstral.so, if any of those are "not  
found".

Also note that the linker may cache the path to some libraries, make  
sure you re-run ldconfig if you delete some symlinks or it'll still  
try to look at where they previously were.

If the problem persist it might be worth adding -rpath (adds an extra  
library path to the .so file) to mod_cepstral's Makefile, it already  
references -L/opt/swift/lib directly anyways.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-05, at 10:12 PM, Jean-Yves F. Barbier wrote:

> Le Mon, 5 Apr 2010 20:59:53 -0500,
> Brian West <brian at freeswitch.org> a ?crit :
>
>> That would be a wrong assumption.  Do you have the correct versions?
>
> What versions of what are you talking about?
>
> if it is about libs, FS was compiled on this computer.
>
>
> -- 
> A woman takes off her claim to respect along with her garments.
> 		-- Herodotus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Tue Apr  6 05:00:56 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 6 Apr 2010 08:00:56 -0400
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <y2jdebff5a81004060307pf81f5092z73406446c3cbf7f1@mail.gmail.com>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
	<40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>
	<y2jdebff5a81004060307pf81f5092z73406446c3cbf7f1@mail.gmail.com>
Message-ID: <37987E90-8D6F-4E67-B3EC-91F6E20752C1@avgs.ca>

SVN is a few days back, if you want to be on the greatest-and-lasted,  
try the GIT repo. See the wiki page for installation details.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-06, at 6:07 AM, afshin afzali wrote:

> No, I don't have any issue with it, I just saw the revision does not  
> change for some days just like previous ones :)
>
> Regards,
>
>
> On Tue, Apr 6, 2010 at 10:12 AM, Brian West <brian at freeswitch.org>  
> wrote:
> well 17188 is the current rev.. what are you having issues with...
>
> /b
>
> On Apr 6, 2010, at 12:28 AM, afshin afzali wrote:
>
> > Hi,
> >  After some issues about svn update in openzap and compile, my  
> FreeSWITCH version stuck at revision 17188 ! I'm in trouble?
> >
> > -- afshin
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Tue Apr  6 04:59:40 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 6 Apr 2010 07:59:40 -0400
Subject: [Freeswitch-users] Help to call a VXML SIP number
In-Reply-To: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
References: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
Message-ID: <539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>

NO_ANSWER means the call was successful, but noone picked it up. Try  
setting debug on (console loglevel debug) and watch for the call  
progress on the console.

Additionally you may enable sip-tracing so you can see all the packets  
on the console: sofia profile external siptrace on

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-06, at 5:44 AM, velusamy Krishnan wrote:

> Dear All,
>      I have received the XXX at sip.voxeo.net from my VXML application.  
> I have written the following dial plan to call that number.
>   <extension name="VXML">
>         <condition field="destination_number" expression="^(999)$">
>                 <action application="bridge" data="sofia/external/9996079973 at sip.voxeo.net 
> "/>
>                 <action application="hangup"/>
>         </condition>
>   </extension>
>
> I have registered the FreeSWITCH and called the 999 extension. But  
> the call was just trying. Finally I have got the NO_ANSWER  
> information in console.
>   2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate  
> Failed.  Cause: NO_ANSWER
>
> Is there configuration I need to do to call this number?? 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 a.afzali2003 at gmail.com  Tue Apr  6 05:24:00 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 6 Apr 2010 16:54:00 +0430
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <37987E90-8D6F-4E67-B3EC-91F6E20752C1@avgs.ca>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
	<40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>
	<y2jdebff5a81004060307pf81f5092z73406446c3cbf7f1@mail.gmail.com>
	<37987E90-8D6F-4E67-B3EC-91F6E20752C1@avgs.ca>
Message-ID: <y2ldebff5a81004060524t69245701z418b31cea1c8bda2@mail.gmail.com>

The fact is I've made some modifications to my source files and yet did not
submit my patches (with hope to be merge) , so I don't know if it is
possible to change my source control from SVN to GIT automatically ?

-- afshin

On Tue, Apr 6, 2010 at 4:30 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> SVN is a few days back, if you want to be on the greatest-and-lasted, try
> the GIT repo. See the wiki page for installation details.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-06, at 6:07 AM, afshin afzali wrote:
>
> No, I don't have any issue with it, I just saw the revision does not change
> for some days just like previous ones :)
>
> Regards,
>
>
> On Tue, Apr 6, 2010 at 10:12 AM, Brian West <brian at freeswitch.org> wrote:
>
>> well 17188 is the current rev.. what are you having issues with...
>>
>> /b
>>
>> On Apr 6, 2010, at 12:28 AM, afshin afzali wrote:
>>
>> > Hi,
>> >  After some issues about svn update in openzap and compile, my
>> FreeSWITCH version stuck at revision 17188 ! I'm in trouble?
>> >
>> > -- afshin
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Tue Apr  6 05:29:09 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 6 Apr 2010 08:29:09 -0400
Subject: [Freeswitch-users] Stuck at revision 17188
In-Reply-To: <y2ldebff5a81004060524t69245701z418b31cea1c8bda2@mail.gmail.com>
References: <k2kdebff5a81004052228nf34ade0et5b0a51383e517f6a@mail.gmail.com>
	<40AF6524-0B30-48BB-9DEC-6E7C7CD020A5@freeswitch.org>
	<y2jdebff5a81004060307pf81f5092z73406446c3cbf7f1@mail.gmail.com>
	<37987E90-8D6F-4E67-B3EC-91F6E20752C1@avgs.ca>
	<y2ldebff5a81004060524t69245701z418b31cea1c8bda2@mail.gmail.com>
Message-ID: <DF9A9CEA-DF23-4C7F-90E3-8B17003F9278@avgs.ca>

You can still make a diff out of them. Its even possible to create a  
branch in the past so it would be easier to resolve if there were to  
be merge conflicts.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-06, at 8:24 AM, afshin afzali wrote:

> The fact is I've made some modifications to my source files and yet  
> did not submit my patches (with hope to be merge) , so I don't know  
> if it is possible to change my source control from SVN to GIT  
> automatically ?
>
> -- afshin 
>
> On Tue, Apr 6, 2010 at 4:30 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> SVN is a few days back, if you want to be on the greatest-and- 
> lasted, try the GIT repo. See the wiki page for installation details.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-06, at 6:07 AM, afshin afzali wrote:
>
>> No, I don't have any issue with it, I just saw the revision does  
>> not change for some days just like previous ones :)
>>
>> Regards,
>>
>>
>> On Tue, Apr 6, 2010 at 10:12 AM, Brian West <brian at freeswitch.org>  
>> wrote:
>> well 17188 is the current rev.. what are you having issues with...
>>
>> /b
>>
>> On Apr 6, 2010, at 12:28 AM, afshin afzali wrote:
>>
>> > Hi,
>> >  After some issues about svn update in openzap and compile, my  
>> FreeSWITCH version stuck at revision 17188 ! I'm in trouble?
>> >
>> > -- afshin
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From 12ukwn at gmail.com  Tue Apr  6 06:31:24 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 6 Apr 2010 15:31:24 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <05D1C7D3-226A-4DCC-AC1B-C256BC058F0F@avgs.ca>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
	<20100406035209.6e5444d0@anubis.defcon1>
	<700C46B8-F74A-429D-AEA0-90112076817C@freeswitch.org>
	<20100406041228.0aeb7038@anubis.defcon1>
	<05D1C7D3-226A-4DCC-AC1B-C256BC058F0F@avgs.ca>
Message-ID: <20100406153124.38060434@anubis.defcon1>

Le Tue, 6 Apr 2010 07:54:54 -0400,
Mathieu Rene <mrene_lists at avgs.ca> a ?crit :

> ldconfig -v will list you all libraries that are in ld's path, start  
> by confirming the correct version of the lib appears in there. You may  
> also chec: ldd /path/to/mod_cepstral.so, if any of those are "not  
> found".

No everything's there, with the right version.

mod_cepstral is link to libswift.so.5, but can't find the specific
languages libs if they're not symlinked into /usr/local/freeswitch/lib

I tried every possible combination (ldconfig after & before recompiling the
module, etc) but it still need these symlinks, may be it is what you say
below

> Also note that the linker may cache the path to some libraries, make  
> sure you re-run ldconfig if you delete some symlinks or it'll still  
> try to look at where they previously were.


> If the problem persist it might be worth adding -rpath (adds an extra  
> library path to the .so file) to mod_cepstral's Makefile, it already  
> references -L/opt/swift/lib directly anyways.

Yeah, I greped the source, and that's why I don't understand.

But I think this issue is a very low priority as the solution's easy :)

(BTW, I tested some other TTS (unfortunately proprietary), and these voices
blew my mind:
http://www.loquendo.com/en/demos/demo_tts.htm
On some, you really can't tell if it is TTS of real.)


-- 
The number of arguments is unimportant unless some of them are correct.
		-- Ralph Hartley


From justin at ejtown.org  Tue Apr  6 06:49:02 2010
From: justin at ejtown.org (Justin B Newman)
Date: Tue, 6 Apr 2010 09:49:02 -0400
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
	<4BBA4F99.102@gmail.com>
	<s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>
Message-ID: <t2j27b0d5bc1004060649od896d083x3671ad9548286772@mail.gmail.com>

On Mon, Apr 5, 2010 at 8:47 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> Sounds like something more appropriate for a load balancer, such as lvs
> (http://www.linuxvirtualserver.org/).
>
> DNS round robin also can't detect failure and stopping routing requests to
> the node, while a tool like lvs could.

On Tue, Apr 6, 2010 at 7:39 AM, Nazim Aghabayov
<nazim.aghabayov at gmail.com> wrote:
> Agree, load balancer is better. It seems that libcurl doesn't respect DNS
> RR.
>

In this case, I'm primarily interested in failover.  Yes, a load
balancer might be more appropriate in the grand scheme of the world,
but because mod_xml_cdr retries a given URL upon failure, much of the
same result could be achieved with significantly less complexity if
round robin worked.

For my purposes, I've gone ahead and defined "url" multiple times;
mod_xml_cdr will try them one after the other.  I lose randomness, but
I can of course define that on a server-by-server basis.  Ugly, but
adding a load balancer to the mix seems like an unnecessary point of
failure for this use case.

Thanks for all your thoughts,

Yours,

-jbn


From frank at carmickle.com  Tue Apr  6 08:04:19 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 6 Apr 2010 11:04:19 -0400
Subject: [Freeswitch-users] mod_xml_cdr DNS
In-Reply-To: <t2j27b0d5bc1004060649od896d083x3671ad9548286772@mail.gmail.com>
References: <x2m27b0d5bc1004051247v3e052058t7e9f41603fd9c5bf@mail.gmail.com>
	<4BBA4F99.102@gmail.com>
	<s2oe15d328c1004051747v75bf5616r618a927f28bf9e@mail.gmail.com>
	<t2j27b0d5bc1004060649od896d083x3671ad9548286772@mail.gmail.com>
Message-ID: <20100406150419.GA18770@base.carmickle.com>

On Tue, Apr 06, Justin B Newman wrote:
Snip...
> For my purposes, I've gone ahead and defined "url" multiple times;
> mod_xml_cdr will try them one after the other.  I lose randomness, but
> I can of course define that on a server-by-server basis.  Ugly, but
> adding a load balancer to the mix seems like an unnecessary point of
> failure for this use case.

You can setup the load balancer to failover.  It seems like you've figured out a solution though.

--FC



From jim at k4gvo.com  Tue Apr  6 08:38:49 2010
From: jim at k4gvo.com (Jim)
Date: Tue, 06 Apr 2010 11:38:49 -0400
Subject: [Freeswitch-users] What is the 99999_enum.xml file?
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E99238F4@prato.activenetwerx.local>
References: <4BBB137C.7010604@k4gvo.com>
	<CA5A491E9DEFBE4CB777DE97E21575E99238F4@prato.activenetwerx.local>
Message-ID: <4BBB5589.8040100@k4gvo.com>

Joseph L. Casale wrote:
>> It seems to be preventing FS from parsing any files beyond it.  FS 
>> parses conf/dialplan/default.xml then each of the files in 
>> conf/dialplan/default/, but never sees any of the other files in 
>> conf/dialplan such as public.xml and the files in public/.
>>     
>
> Enum should be the last dialplan you get to if nothing matches, prolly
> if you hit it, you aren't writing your dialplan properly or you overlooked
> something.
>
> http://en.wikipedia.org/wiki/Telephone_Number_Mapping
>
> You are looking at the 'Default' context which is for auth'ed ua's.
> The dialplan only encounters the 'Public' context for unauth'ed ua's,
> which get placed in there first then hopefully route somewhere else,
> the default context out of the box at least will never look into the
> 'Public' context.
>   
Thanks.
Jim.


From msc at freeswitch.org  Tue Apr  6 11:52:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Apr 2010 11:52:01 -0700
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
Message-ID: <u2w87f2f3b91004061152wc0f8cacpda89b91be2e84e02@mail.gmail.com>

You might want to update to the boost PRI stack and try again:
http://wiki.sangoma.com/wanpipe-api-freetdm

-MC


On Mon, Apr 5, 2010 at 9:24 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> In my office we have a Hard PBX, with some 4 extensions.
> We also have sangoma A102 card.
> From the Hard PBX, if 0 is pressed, it is setup in a way that it will go to
> outside world.
> I've connected that line to span2 of the card.
> The span2 in the A102 card, is configured as PRI_NET.
>
> wanrouter status shows connected for the span2.
>
> But if I dial from the extension, I got the following in the sangoma_dchan
> log.
> 2010-04-03 12:49:49
> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 33
> 39 31 31 34 36 30 30 7d 02 91 81 ]
>   Call Ref:0164
>   Type:Setup (0x5)
>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
> Kbit/s(16) L1Prot:G.711 A-Law(3)
>   Calling Party Number:4439114600(l:10) plan:isdn(1)
> type:unknown(0)scr:user, passed(1) pres:allowed(0)
>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>
> 2010-04-03 12:49:49
> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>   Call Ref:0164
>   Type:Release Compl (0x5a)
>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
> Circuit/Channel Available(34)
>
> The call in not reaching freeswitch ( I enabled debug log. But nothing is
> printing in it ).
> Can someone suggest how to make this work.
>
> Please ask me if you need more information?, since I don't know what to
> give 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 phunk0000 at hotmail.com  Tue Apr  6 12:51:16 2010
From: phunk0000 at hotmail.com (Todd)
Date: Tue, 6 Apr 2010 15:51:16 -0400
Subject: [Freeswitch-users] mod_limit
Message-ID: <BLU110-DS43990ED0B4B6ABAB07D69A7180@phx.gbl>

Hey list, I have a question:  Can mod_limit be used to limit the number of
incoming call from a specific caller ID in a certain time interval?

 

Limit the calls per second by source ip + destination number: 

<action application="limit_hash" data="${sip_received_ip}
${destination_number} ${calls_per_second}/1" />

 

I found this in the wiki and I was wondering how you would change the action
to limit an incoming caller id to 5 calls per minute.  Still pretty noo, so
any help in the right direction would be great.  thx

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From mrene_lists at avgs.ca  Tue Apr  6 12:55:20 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 6 Apr 2010 15:55:20 -0400
Subject: [Freeswitch-users] mod_limit
In-Reply-To: <BLU110-DS43990ED0B4B6ABAB07D69A7180@phx.gbl>
References: <BLU110-DS43990ED0B4B6ABAB07D69A7180@phx.gbl>
Message-ID: <24699E3A-0777-4D3C-8BF7-8822030C3999@avgs.ca>

> <action application="limit_hash" data="cidlimit ${caller_id_number} $ 
> {calls_per_second}/1" />

5 calls per min would be 5/60.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-06, at 3:51 PM, Todd wrote:

> Hey list, I have a question:  Can mod_limit be used to limit the  
> number of incoming call from a specific caller ID in a certain time  
> interval?
>
> Limit the calls per second by source ip + destination number:
> <action application="limit_hash" data="${sip_received_ip} $ 
> {destination_number} ${calls_per_second}/1" />
>
> I found this in the wiki and I was wondering how you would change  
> the action to limit an incoming caller id to 5 calls per minute.   
> Still pretty noo, so any help in the right direction would be  
> great.  thx
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From uzoechi at yahoo.com  Tue Apr  6 12:55:30 2010
From: uzoechi at yahoo.com (Uzo Uzo)
Date: Tue, 6 Apr 2010 12:55:30 -0700 (PDT)
Subject: [Freeswitch-users] how do I get my DID number from lua
	script	with session:getvariable
In-Reply-To: <s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
	<s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
Message-ID: <471791.15451.qm@web32806.mail.mud.yahoo.com>

I placed the script in the public context and tried that, but it shows nothing.  In the default context, it will show 1002.  But in public, it shows nothing. 



________________________________
From: David Ponzone <david.ponzone at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, April 5, 2010 12:58:03 PM
Subject: Re: [Freeswitch-users] how do I get my DID number from lua scriptwith session:getvariable

try rdnis

But you could put your lua script in your public context, where the  
calls are handled first.
I dont really see the point to do that in default.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com 


      
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From lloyd.aloysius at gmail.com  Tue Apr  6 13:18:44 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 6 Apr 2010 16:18:44 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <5DA9DF5C-6F60-4CA5-A94D-8DF291BDF751@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
	<A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
	<m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>
	<5DA9DF5C-6F60-4CA5-A94D-8DF291BDF751@freeswitch.org>
Message-ID: <i2g8a19bf2e1004061318sc4901605i67bc546be1b3e2db@mail.gmail.com>

I try the 17186 same issue.

Then I pull the latest source from Git. Everything back to normal.

Thanks
Lloyd


On Mon, Apr 5, 2010 at 6:20 PM, Brian West <brian at freeswitch.org> wrote:

> Considering that their was only exactly ONE patch between 17155 to 17188
> that involved sofia and didn't touch any of that code in sofia related to
> nat (add killgw _all_ to delete all gws 17167)  the rest were mod_skinny,
> mod_sangoma_codec, swigall, packaging and git related changes.
>
> 17187 was making sofia work on OpenBSD you could try to use 17186, And
> report...
>
> But looking over the changes between those two I don't see anything that
> obvious related to sofia and nat that could have broken.
>
> /b
>
>
> On Apr 5, 2010, at 5:08 PM, Aloysius Lloyd wrote:
>
> > Brian,
> >
> > Further to my previous off list email.
> >
> > I can confirm now 100% something went wrong between 17155 to 17188.
> >
> > The same setup working on the following version
> >
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.trunk (17155)
> >
> > But not working in the following version
> >
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.head (svn-17188)
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  6 13:22:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 15:22:10 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <i2g8a19bf2e1004061318sc4901605i67bc546be1b3e2db@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
	<A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
	<m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>
	<5DA9DF5C-6F60-4CA5-A94D-8DF291BDF751@freeswitch.org>
	<i2g8a19bf2e1004061318sc4901605i67bc546be1b3e2db@mail.gmail.com>
Message-ID: <371566A8-720E-402E-A9A5-A46A91719D26@freeswitch.org>

No source changed between 17186 and 17187 other then the build system for OpenBSD.

/b

On Apr 6, 2010, at 3:18 PM, Aloysius Lloyd wrote:

> I try the 17186 same issue.
> 
> Then I pull the latest source from Git. Everything back to normal.
> 
> Thanks
> Lloyd



From brian at freeswitch.org  Tue Apr  6 13:22:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 15:22:33 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <i2g8a19bf2e1004061318sc4901605i67bc546be1b3e2db@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
	<A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>
	<p2o8a19bf2e1004051131u9bc7b320q8f4558972545338c@mail.gmail.com>
	<A49C9699-FB2C-46E5-8D24-D46E84FC828F@freeswitch.org>
	<m2r8a19bf2e1004051508t8a3c55abz4161d46017d61984@mail.gmail.com>
	<5DA9DF5C-6F60-4CA5-A94D-8DF291BDF751@freeswitch.org>
	<i2g8a19bf2e1004061318sc4901605i67bc546be1b3e2db@mail.gmail.com>
Message-ID: <A34A5D04-76AC-4DAE-8C52-3A3C611428BE@freeswitch.org>

Can you try 17187 then 17188

/b

On Apr 6, 2010, at 3:18 PM, Aloysius Lloyd wrote:

> I try the 17186 same issue.
> 
> Then I pull the latest source from Git. Everything back to normal.
> 
> Thanks
> Lloyd
> 
> 
> On Mon, Apr 5, 2010 at 6:20 PM, Brian West <brian at freeswitch.org> wrote:
> Considering that their was only exactly ONE patch between 17155 to 17188 that involved sofia and didn't touch any of that code in sofia related to nat (add killgw _all_ to delete all gws 17167)  the rest were mod_skinny, mod_sangoma_codec, swigall, packaging and git related changes.
> 
> 17187 was making sofia work on OpenBSD you could try to use 17186, And report...
> 
> But looking over the changes between those two I don't see anything that obvious related to sofia and nat that could have broken.
> 
> /b

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From david.yatsin at gmail.com  Tue Apr  6 13:47:39 2010
From: david.yatsin at gmail.com (David Yat Sin)
Date: Tue, 06 Apr 2010 16:47:39 -0400
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
Message-ID: <4BBB9DEB.2040702@gmail.com>

Hi Lakshmanan,
If you do not have anything printing in /var/log/sangoma_mgd.log, and 
you have:
verbose=4           //(or higher)

in /etc/wanpipe/smg_pri.conf, check that you have these lines in 
/etc/syslog.conf (or /etc/rsyslog.conf):

local2.*                /var/log/sangoma_mgd.log

and restart your syslog.


my first guess is that you do not have openzap loaded so sangoma_prid is 
rejecting all incoming calls, but I would need logs in 
/var/log/sangoma_mgd.log to confirm.

If openzap is not loaded, you can type:
load mod_openzap

from the freeswitch CLI to load it.

-- 

*David Yat Sin, **BEng, **Software Developer**
*Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON 
L3R 9T3 Canada
t. 1 905 474 1990 x 119 | e. _dyatsin at sangoma.com 
<mailto:dyatsin at sangoma.com>_



On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
> Hi all,
> In my office we have a Hard PBX, with some 4 extensions.
> We also have sangoma A102 card.
> From the Hard PBX, if 0 is pressed, it is setup in a way that it will 
> go to outside world.
> I've connected that line to span2 of the card.
> The span2 in the A102 card, is configured as PRI_NET.
>
> wanrouter status shows connected for the span2.
>
> But if I dial from the extension, I got the following in the 
> sangoma_dchan log.
> 2010-04-03 12:49:49
> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 
> 33 39 31 31 34 36 30 30 7d 02 91 81 ]
>   Call Ref:0164
>   Type:Setup (0x5)
>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) 
> TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3)
>   Calling Party Number:4439114600(l:10) plan:isdn(1) 
> type:unknown(0)scr:user, passed(1) pres:allowed(0)
>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>
> 2010-04-03 12:49:49
> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>   Call Ref:0164
>   Type:Release Compl (0x5a)
>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No 
> Circuit/Channel Available(34)
>
> The call in not reaching freeswitch ( I enabled debug log. But nothing 
> is printing in it ).
> Can someone suggest how to make this work.
>
> Please ask me if you need more information?, since I don't know what 
> to give 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  Tue Apr  6 13:55:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Apr 2010 13:55:04 -0700
Subject: [Freeswitch-users] how do I get my DID number from lua script
	with session:getvariable
In-Reply-To: <471791.15451.qm@web32806.mail.mud.yahoo.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
	<s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
	<471791.15451.qm@web32806.mail.mud.yahoo.com>
Message-ID: <t2m87f2f3b91004061355r287e7be4v89e184be91fa6c64@mail.gmail.com>

To make sure that the data you are looking for is even available I recommend
adding an info app before calling the Lua script:
<action application="info"/>

Watch the console and it will spit out a bunch of information. Find the data
field that contains the information you seek. Check out this page for more
information on the different channel variables:
http://wiki.freeswitch.org/wiki/Channel_Variables

Once you know which channel var has the data you're looking for then it's
easy to get it with session:getVariable("var_name")

-MC

On Tue, Apr 6, 2010 at 12:55 PM, Uzo Uzo <uzoechi at yahoo.com> wrote:

> I placed the script in the public context and tried that, but it shows
> nothing.  In the default context, it will show 1002.  But in public, it
> shows nothing.
>
> ------------------------------
> *From:* David Ponzone <david.ponzone at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Mon, April 5, 2010 12:58:03 PM
> *Subject:* Re: [Freeswitch-users] how do I get my DID number from lua
> script with session:getvariable
>
> try rdnis
>
> But you could put your lua script in your public context, where the
> calls are handled first.
> I dont really see the point to do that in default.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26www.ipeva.fr  -   www.ipeva-studio.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 david.ponzone at gmail.com  Tue Apr  6 16:27:17 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 7 Apr 2010 01:27:17 +0200
Subject: [Freeswitch-users] how do I get my DID number from lua script
	with session:getvariable
In-Reply-To: <471791.15451.qm@web32806.mail.mud.yahoo.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
	<188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>
	<s2o914fc92a1004051454v52cf53dz202a249aa06daacc@mail.gmail.com>
	<471791.15451.qm@web32806.mail.mud.yahoo.com>
Message-ID: <468C25B1-AEB1-446A-B2DC-389BF9864B78@gmail.com>

In the public context, your extension must now look like:

<extension name="mylua">
  <condition field="destination_number"  
"expression="^regexp_matching_your_DIDs$">
   <action application="lua" data="/usr/local/freeswitch/scripts/ 
foo.lua"/>
  </condition>
</extension>

Inside the script, you should be able to get the DID using the  
variable destination_number.
Another way, simpler I would say, is to do:

<extension name="mylua">
  <condition field="destination_number"  
"expression="^(regexp_matching_your_DIDs)$">
   <action application="lua" data="/usr/local/freeswitch/scripts/ 
foo.lua $1"/>
  </condition>
</extension>

This way, you get the DID as argv[1] in your LUA script.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 06/04/2010 ? 21:55, Uzo Uzo a ?crit :

> I placed the script in the public context and tried that, but it  
> shows nothing.  In the default context, it will show 1002.  But in  
> public, it shows nothing.
>
> From: David Ponzone <david.ponzone at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, April 5, 2010 12:58:03 PM
> Subject: Re: [Freeswitch-users] how do I get my DID number from lua  
> script	with session:getvariable
>
> try rdnis
>
> But you could put your lua script in your public context, where the
> calls are handled first.
> I dont really see the point to do that in default.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.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 gliu at tecomtech.com  Tue Apr  6 18:09:37 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 09:09:37 +0800
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139 Gateway
	information missing)
Message-ID: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>

Hi,

 

I'm testing SLA/BLA feature with Freeswitch, but failed. Could you tell me
what's wrong ?

 

I see that both the server and the phone subscribe to each other, but the
last NOTIFY from the phone to the server gets rejected with "Call does not
exist".

 

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.

 

 

 

Thanks.

Bryan

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From gliu at tecomtech.com  Tue Apr  6 19:37:21 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 10:37:21 +0800
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139 Gateway
	information missing)
Message-ID: <5C722E76DAAB41C596BBEC8A61D0323A@sohobryan>

 

Hi,

 

I'm testing SLA/BLA feature with Freeswitch, but failed. Could you tell me
what's wrong ?

 

I see that both the server and the phone subscribe to each other, but the
last NOTIFY from the phone to the server gets rejected with "Call does not
exist".

 

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.

 

 

 

Thanks.

Bryan

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From brian at freeswitch.org  Tue Apr  6 19:50:43 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 21:50:43 -0500
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139 Gateway
	information missing)
In-Reply-To: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
Message-ID: <735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>

You can't do BLA/BLF from FS to FS with a register... what phones are you using?

/b

On Apr 6, 2010, at 8:09 PM, Bryan wrote:

>  
> I see that both the server and the phone subscribe to each other, but the last NOTIFY from the phone to the server gets rejected with "Call does not exist".
>  
> On the console I get the following error message: [ERR] sofia_presence.c:2135 Gateway information missing.
>  
>  

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From gliu at tecomtech.com  Tue Apr  6 20:03:59 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 11:03:59 +0800
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
	Gatewayinformation missing)
In-Reply-To: <735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
Message-ID: <684CB9E9F3714D7D89A453DB302079F0@sohobryan>

Dear Brian,

 

Thanks for your reply.

 

I?m test the feature with our phone, I follow the
draft-anil-sipping-bla-03.txt to to the test.

 

Could you  tell me what?s wrong with SIP flow.

 

Thanks a lot.

 

Best Regards,

Bryan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 10:51
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
Gatewayinformation missing)

 

You can't do BLA/BLF from FS to FS with a register... what phones are you
using?

 

/b

 

On Apr 6, 2010, at 8:09 PM, Bryan wrote:





 

I see that both the server and the phone subscribe to each other, but the
last NOTIFY from the phone to the server gets rejected with "Call does not
exist".

 

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.

 

 

 

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From brian at freeswitch.org  Tue Apr  6 20:07:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 22:07:31 -0500
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
	Gatewayinformation missing)
In-Reply-To: <684CB9E9F3714D7D89A453DB302079F0@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
	<684CB9E9F3714D7D89A453DB302079F0@sohobryan>
Message-ID: <803F8963-0FD5-4050-8524-27612FE85F8A@freeswitch.org>

our phones? what phones might those be?  sounds like you're using dialog info.

/b

On Apr 6, 2010, at 10:03 PM, Bryan wrote:

> Dear Brian,
>  
> Thanks for your reply.
>  
> I?m test the feature with our phone, I follow the draft-anil-sipping-bla-03.txt to to the test.
>  
> Could you  tell me what?s wrong with SIP flow.
>  
> Thanks a lot.
>  
> Best Regards,
> Bryan

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From gliu at tecomtech.com  Tue Apr  6 20:08:15 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 11:08:15 +0800
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
	Gatewayinformation missing)
In-Reply-To: <735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
Message-ID: <827F380A80A7450F92996F9D3A379B90@sohobryan>

Dear Brian,

 

172.16.1.115 --> Freeswitch server.

172.16.11.176-> My IP phone.

 

 

FYR.

 

Bryan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 10:51
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
Gatewayinformation missing)

 

You can't do BLA/BLF from FS to FS with a register... what phones are you
using?

 

/b

 

On Apr 6, 2010, at 8:09 PM, Bryan wrote:





 

I see that both the server and the phone subscribe to each other, but the
last NOTIFY from the phone to the server gets rejected with "Call does not
exist".

 

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.

 

 

 

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From brian at freeswitch.org  Tue Apr  6 20:12:51 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 22:12:51 -0500
Subject: [Freeswitch-users] SLA/BLA issue(sofia_presence.c:2139
	Gatewayinformation missing)
In-Reply-To: <827F380A80A7450F92996F9D3A379B90@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
	<827F380A80A7450F92996F9D3A379B90@sohobryan>
Message-ID: <1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>

Ok I'm not sure how I can ask this where I can get thru to your brain... WHAT BRAND OF PHONE IS IT IT?  Can you elaborate on the phone in question?

/b

On Apr 6, 2010, at 10:08 PM, Bryan wrote:

> Dear Brian,
>  
> 172.16.1.115 ? Freeswitch server.
> 172.16.11.176-> My IP phone.
>  
>  
> FYR.
>  
> Bryan

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From gliu at tecomtech.com  Tue Apr  6 20:26:47 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 11:26:47 +0800
Subject: [Freeswitch-users] SLA/BLA
	issue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan>
	<1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
Message-ID: <964CAAB8FB9F4F37B8632B0489F10287@sohobryan>

Dear Brian,

 

Sorry, this phone is developed by  our RDs. 

We want to do IOT with FS.

 

BR/Bryan  

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 11:13
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLA/BLA
issue(sofia_presence.c:2139Gatewayinformation missing)

 

Ok I'm not sure how I can ask this where I can get thru to your brain...
WHAT BRAND OF PHONE IS IT IT?  Can you elaborate on the phone in question?

 

/b

 

On Apr 6, 2010, at 10:08 PM, Bryan wrote:





Dear Brian,

 

172.16.1.115 --> Freeswitch server.

172.16.11.176-> My IP phone.

 

 

FYR.

 

Bryan

 

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From gliu at tecomtech.com  Tue Apr  6 20:43:58 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 11:43:58 +0800
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <964CAAB8FB9F4F37B8632B0489F10287@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan><1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
	<964CAAB8FB9F4F37B8632B0489F10287@sohobryan>
Message-ID: <7DC4E1D54F6842D7B819159A13D184F5@sohobryan>

Dear Brian,

 

My issue is same as 

http://permalink.gmane.org/gmane.comp.telephony.freeswitch.user/23612

 

BR/Bryan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Sent: 2010?4?7? 11:27
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users]
SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)

 

Dear Brian,

 

Sorry, this phone is developed by  our RDs. 

We want to do IOT with FS.

 

BR/Bryan  

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 11:13
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLA/BLA
issue(sofia_presence.c:2139Gatewayinformation missing)

 

Ok I'm not sure how I can ask this where I can get thru to your brain...
WHAT BRAND OF PHONE IS IT IT?  Can you elaborate on the phone in question?

 

/b

 

On Apr 6, 2010, at 10:08 PM, Bryan wrote:

 

Dear Brian,

 

172.16.1.115 --> Freeswitch server.

172.16.11.176-> My IP phone.

 

 

FYR.

 

Bryan

 

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From brian at freeswitch.org  Tue Apr  6 20:47:16 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 22:47:16 -0500
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan><1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
	<964CAAB8FB9F4F37B8632B0489F10287@sohobryan>
	<7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
Message-ID: <A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>

No its not the same thing.  BroadSoft SCA is not the same as what you outlined.  You'er talking Dialog-info... totally different... 

/b

On Apr 6, 2010, at 10:43 PM, Bryan wrote:

> Dear Brian,
>  
> My issue is same as
> http://permalink.gmane.org/gmane.comp.telephony.freeswitch.user/23612
>  
> BR/Bryan
>  

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From gliu at tecomtech.com  Tue Apr  6 20:55:29 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 11:55:29 +0800
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan><1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org><964CAAB8FB9F4F37B8632B0489F10287@sohobryan><7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
	<A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>
Message-ID: <3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>

Do you mean FS just support Broadsoft?s SCA spec, no support for
draft-anil-sipping-bla-03.txt.

 

BTW, I use snom 360 phones, got the same result.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 11:47
To: freeswitch-users at lists.freeswitch.org
Subject: Re:
[Freeswitch-users]SLA/BLAissue(sofia_presence.c:2139Gatewayinformation
missing)

 

No its not the same thing.  BroadSoft SCA is not the same as what you
outlined.  You'er talking Dialog-info... totally different... 

 

/b

 

On Apr 6, 2010, at 10:43 PM, Bryan wrote:





Dear Brian,

 

My issue is same as

http://permalink.gmane.org/gmane.comp.telephony.freeswitch.user/23612

 

BR/Bryan

 

 

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From serget68 at msn.com  Tue Apr  6 20:59:50 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Tue, 6 Apr 2010 22:59:50 -0500
Subject: [Freeswitch-users] ZRTP protocol measurement
In-Reply-To: <20100401070915.GB13609@jdc.jasonjgw.net>
References: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>,
	<20100401070915.GB13609@jdc.jasonjgw.net>
Message-ID: <COL108-W61FE668BD21CF4D09E8BF6D9170@phx.gbl>


Hi Jason,

 

Thank you so much for your suggestions. You adviced to do the measurement in whatever way I normally would, comparing ZRTP sessions with sessions that do not involve ZRTP, and seeing if there are performance differences that affect my usage scenario.

 

In other world, comparing the situation where I have zfone to secure end points and the situation where I have no zfone. I still want to know the tools (security tools or network tools or any others) to use and how to come up with performance differences.

 

Thank you!

 

Serge.
 


> Date: Thu, 1 Apr 2010 18:09:15 +1100
> From: jason at jasonjgw.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ZRTP protocol measurement
> 
> SERGE TUMBA <serget68 at msn.com> wrote:
> > 
> > I would like to know how to measure the performence of the zrtp using
> > FreeSWITCH that connect two X-Lite softphones which use the zfone for
> > encrypting voice packets on both end phones.
> 
> What do you want to know? There are almost as many performance measurements as
> there are people doing the measuring. I would suggest measuring in whatever
> way you normally would, comparing ZRTP sessions with sessions that do not
> involve ZRTP, and seeing if there are performance differences that affect your
> usage scenario.
> > 
> > Also, can someone contrast and compare ZRTP to SRTP focusing on these two
> > protocol behaviors.
> 
> Have a look at http://www.zfone.com/ for a description of ZRTP. From an
> operational perspective, the main difference is that in configuring SRTP, you
> need to use TLS to secure the SIP signaling; otherwise, the cryptographic keys
> are transmitted in the clear, which completely eliminates the security.
> Setting up TLS securely requires a public-key infrastructure whereby each side
> verifies the identity of the other.
> 
> In ZRTP, the negotiation takes place entirely in the RTP stream; there are
> several protection mechanisms provided to prevent third-parties from
> masquerading as one of the end-points (namely, key finger-prints, displayed to
> the user as words that can be verified in the conversation, and
> the use of mathematically related keys in subsequent sessions between the same
> parties, but without diminishing security). No public-key infrastructure is
> needed, hence no X.509 certificates or TLS are required.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. 
http://www.windowslive.com/campaign/thenewbusy?tile=multicalendar&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5
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From brian at freeswitch.org  Tue Apr  6 21:01:21 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Apr 2010 23:01:21 -0500
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan><1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org><964CAAB8FB9F4F37B8632B0489F10287@sohobryan><7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
	<A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>
	<3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>
Message-ID: <4B224150-69C0-43D2-A6D5-58966F59E23E@freeswitch.org>

I know it works if you enable manage-presence and what not... and setup the lines to subscribe correctly.

/b

On Apr 6, 2010, at 10:55 PM, Bryan wrote:

> Do you mean FS just support Broadsoft?s SCA spec, no support for draft-anil-sipping-bla-03.txt.
>  
> BTW, I use snom 360 phones, got the same result.

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From serget68 at msn.com  Tue Apr  6 21:13:57 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Tue, 6 Apr 2010 23:13:57 -0500
Subject: [Freeswitch-users] Security mesurement
Message-ID: <COL108-W14D3257D1FBD8588EC58ED9170@phx.gbl>



When comparing ZRTP sessions with sessions that do not involve ZRTP, and seeing if there are performance differences that affect usage scenario.
 
In other world, comparing the situation where zfone encrypt two end points and the situation where no zfone is used. What tools (security tools or network tools or any others) can help to perform this task and how to come up with performance differences.
 
Thank you!
 
Serge.



 		 	   		  
_________________________________________________________________
The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail.
http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4
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From serget68 at msn.com  Tue Apr  6 21:15:33 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Tue, 6 Apr 2010 23:15:33 -0500
Subject: [Freeswitch-users] SRTP on FreeSWITCH
Message-ID: <COL108-W6549C5FDCFC9877AFBD931D9170@phx.gbl>


Dear All,
 
I know that FreeSWITCH supports ZRTP but I would like to know if someone already tried to deploy SRTP on FreeSWITCH? If so, can I have the HOWTO describing this setup if any have been done?
 
Thank you!
 
Serge.





 


 		 	   		  
_________________________________________________________________
The New Busy is not the old busy. Search, chat and e-mail from your inbox.
http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3
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From gliu at tecomtech.com  Tue Apr  6 21:23:48 2010
From: gliu at tecomtech.com (Bryan)
Date: Wed, 7 Apr 2010 12:23:48 +0800
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <4B224150-69C0-43D2-A6D5-58966F59E23E@freeswitch.org>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan><735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org><827F380A80A7450F92996F9D3A379B90@sohobryan><1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org><964CAAB8FB9F4F37B8632B0489F10287@sohobryan><7DC4E1D54F6842D7B819159A13D184F5@sohobryan><A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org><3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>
	<4B224150-69C0-43D2-A6D5-58966F59E23E@freeswitch.org>
Message-ID: <DA5E20C0FF09410798F9637759C7B8EE@sohobryan>

Dear Brian,

 

I have set

 

    <param name="manage-presence" value="true"/>

         <param name="manage-shared-appearance" value="sylantro"/>

         <param name="multiple-registrations" value="contact"/>

 

BR

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 2010?4?7? 12:01
To: freeswitch-users at lists.freeswitch.org
Subject: Re:
[Freeswitch-users]SLA/BLAissue(sofia_presence.c:2139Gatewayinformation
missing)

 

I know it works if you enable manage-presence and what not... and setup the
lines to subscribe correctly.

 

/b

 

On Apr 6, 2010, at 10:55 PM, Bryan wrote:





Do you mean FS just support Broadsoft?s SCA spec, no support for
draft-anil-sipping-bla-03.txt.

 

BTW, I use snom 360 phones, got the same result.

 

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From serget68 at msn.com  Tue Apr  6 21:27:49 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Tue, 6 Apr 2010 23:27:49 -0500
Subject: [Freeswitch-users] Protection with ZRTP
Message-ID: <COL108-W2FD2DC001F754B53C748BD9170@phx.gbl>


I know that in ZRTP, the negotiation of the cryptographic keys takes place entirely in the RTP stream. However, I do not understand:

 

1. How can this be proven when having setup FreeSWITCH and having deployed ZRTP, and having two softphones (such as X-LITE) connected in a test environment. Can a tool cush as Wireshark or any other be used in this case?

 

2. What are other protection mechanisms provided to prevent third-parties from masquerading as one of the end-points?

 

Thank you!

 

Serge.




 


 		 	   		  
_________________________________________________________________
The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. 
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From jason at jasonjgw.net  Tue Apr  6 22:03:32 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Apr 2010 15:03:32 +1000
Subject: [Freeswitch-users] Security mesurement
In-Reply-To: <COL108-W14D3257D1FBD8588EC58ED9170@phx.gbl>
References: <COL108-W14D3257D1FBD8588EC58ED9170@phx.gbl>
Message-ID: <20100407050332.GA13936@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> 
> When comparing ZRTP sessions with sessions that do not involve ZRTP, and seeing if there are performance differences that affect usage scenario.
>  
> In other world, comparing the situation where zfone encrypt two end points and the situation where no zfone is used. What tools (security tools or network tools or any others) can help to perform this task and how to come up with performance differences.

My guess is that the performance differences are negligible, but if you're
going to do load testing anyway, then you should try it with and without ZRTP
to make sure the overhead of the encryption isn't a problem for you.

What testing you decide to do is entirely up to you.



From jason at jasonjgw.net  Tue Apr  6 22:06:41 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Apr 2010 15:06:41 +1000
Subject: [Freeswitch-users] SRTP on FreeSWITCH
In-Reply-To: <COL108-W6549C5FDCFC9877AFBD931D9170@phx.gbl>
References: <COL108-W6549C5FDCFC9877AFBD931D9170@phx.gbl>
Message-ID: <20100407050641.GB13936@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> I know that FreeSWITCH supports ZRTP but I would like to know if someone already tried to deploy SRTP on FreeSWITCH? If so, can I have the HOWTO describing this setup if any have been done?

 I tried it, but it exposed bugs in OpenSSL. Soon thereafter, ZRTP became
 available so I switched to that.

 If you want SRTP, you have to turn on TLS, make sure that both ends are using
 it, and enable SRTP in the SIP profile, if I remember correctly. ZRTP is
 better in my opinion; the only reason why you would want to use SRTP is for
 interoperability with end-points that support it, but not ZRTP - unless you
 have a public-key infrastructure in place that you want to use for some
 reason.



From serget68 at msn.com  Tue Apr  6 23:00:51 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Wed, 7 Apr 2010 01:00:51 -0500
Subject: [Freeswitch-users] SRTP interoperability
In-Reply-To: <20100407050641.GB13936@jdc.jasonjgw.net>
References: <COL108-W6549C5FDCFC9877AFBD931D9170@phx.gbl>,
	<20100407050641.GB13936@jdc.jasonjgw.net>
Message-ID: <COL108-W40A66E3A7C9E32E7E7D5B3D9170@phx.gbl>


hello Jason,

 

Thank you for your prompt responses and you have been very helpfull. I also tried SRTP on Asterisk and I got virus when downloading libraries ... and I also found ZRTP is better. However, can you be specific about SRTP interoperability with end-points that support it? it looks like this is a benefit of SRTP over ZRTP? While ZRTP is based on SRTP right?

 

Thank you!

 

Serge.
 


> Date: Wed, 7 Apr 2010 15:06:41 +1000
> From: jason at jasonjgw.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SRTP on FreeSWITCH
> 
> SERGE TUMBA <serget68 at msn.com> wrote:
> > 
> > I know that FreeSWITCH supports ZRTP but I would like to know if someone already tried to deploy SRTP on FreeSWITCH? If so, can I have the HOWTO describing this setup if any have been done?
> 
> I tried it, but it exposed bugs in OpenSSL. Soon thereafter, ZRTP became
> available so I switched to that.
> 
> If you want SRTP, you have to turn on TLS, make sure that both ends are using
> it, and enable SRTP in the SIP profile, if I remember correctly. ZRTP is
> better in my opinion; the only reason why you would want to use SRTP is for
> interoperability with end-points that support it, but not ZRTP - unless you
> have a public-key infrastructure in place that you want to use for some
> reason.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  6 23:15:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Apr 2010 02:15:37 -0400
Subject: [Freeswitch-users] SRTP interoperability
In-Reply-To: <COL108-W40A66E3A7C9E32E7E7D5B3D9170@phx.gbl>
References: <COL108-W6549C5FDCFC9877AFBD931D9170@phx.gbl>,
	<20100407050641.GB13936@jdc.jasonjgw.net>
	<COL108-W40A66E3A7C9E32E7E7D5B3D9170@phx.gbl>
Message-ID: <547AF7AB-B0AE-4BEF-9042-9092F6CEF6C3@jerris.com>

I can see that you are looking for some more information on how strp and zrtp work. I have found quite a bit of information on this topic at the following url:

http://www.google.com/search?q=srtp+zrtp

Mike

On Apr 7, 2010, at 2:00 AM, SERGE TUMBA wrote:

> hello Jason,
>  
> Thank you for your prompt responses and you have been very helpfull. I also tried SRTP on Asterisk and I got virus when downloading libraries ... and I also found ZRTP is better. However, can you be specific about SRTP interoperability with end-points that support it? it looks like this is a benefit of SRTP over ZRTP? While ZRTP is based on SRTP right?
>  
> Thank you!
>  
> Serge.
>  
> 
> > Date: Wed, 7 Apr 2010 15:06:41 +1000
> > From: jason at jasonjgw.net
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] SRTP on FreeSWITCH
> > 
> > SERGE TUMBA <serget68 at msn.com> wrote:
> > > 
> > > I know that FreeSWITCH supports ZRTP but I would like to know if someone already tried to deploy SRTP on FreeSWITCH? If so, can I have the HOWTO describing this setup if any have been done?
> > 
> > I tried it, but it exposed bugs in OpenSSL. Soon thereafter, ZRTP became
> > available so I switched to that.
> > 
> > If you want SRTP, you have to turn on TLS, make sure that both ends are using
> > it, and enable SRTP in the SIP profile, if I remember correctly. ZRTP is
> > better in my opinion; the only reason why you would want to use SRTP is for
> > interoperability with end-points that support it, but not ZRTP - unless you
> > have a public-key infrastructure in place that you want to use for some
> > reason.
> > 
> > 
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From velu.technical at gmail.com  Tue Apr  6 23:33:12 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Wed, 7 Apr 2010 12:03:12 +0530
Subject: [Freeswitch-users] Help to call a VXML SIP number
In-Reply-To: <539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>
References: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
	<539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>
Message-ID: <y2g1452e2981004062333vc6da46b0s4a7249142cfe09b@mail.gmail.com>

Hi,
   I have enabled the siptrace in console. I have found that only SIP INVITE
packets came in the console. The response packet didn't come.
   Could you anyone help me to know what is the problem??


On Tue, Apr 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> NO_ANSWER means the call was successful, but noone picked it up. Try
> setting debug on (console loglevel debug) and watch for the call progress on
> the console.
>
> Additionally you may enable sip-tracing so you can see all the packets on
> the console: sofia profile external siptrace on
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-06, at 5:44 AM, velusamy Krishnan wrote:
>
> Dear All,
>      I have received the XXX at sip.voxeo.net from my VXML application. I
> have written the following dial plan to call that number.
>   <extension name="VXML">
>         <condition field="destination_number" expression="^(999)$">
>                 <action application="bridge" data="sofia/external/
> 9996079973 at sip.voxeo.net"/>
>                 <action application="hangup"/>
>         </condition>
>   </extension>
>
> I have registered the FreeSWITCH and called the 999 extension. But the call
> was just trying. Finally I have got the NO_ANSWER information in console.
>   2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate Failed.
> Cause: NO_ANSWER
>
> Is there configuration I need to do to call this number?? 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
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>
>
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From lakindia89 at gmail.com  Wed Apr  7 01:48:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 7 Apr 2010 14:18:29 +0530
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <4BBB9DEB.2040702@gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
	<4BBB9DEB.2040702@gmail.com>
Message-ID: <v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>

Hi,
I have set verbose=4 in smg_pri.conf.
I started the wanrouter.
Here is the sangoma_mgd.log
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: ================System
restart=============
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Version:
1.63                           =
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Date: Feb 26
2010                       =
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
15607                =
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid:
===========================================
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Number of spans:2
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Verbosity set to:4
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Debug disabled (local:2)
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Boost disabled (local:6)
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:1 pri_cpe euroisdn
dChan:16
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:2 pri_net euroisdn
dChan:16
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Down
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(P)
Local:127.0.0.66:53001Remote:
127.0.0.65:53001
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(N)
Local:127.0.0.66:53000Remote:
127.0.0.65:53000
Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:Version:103
Apr  7 13:58:27 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Up
Apr  7 13:58:31 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Down
Apr  7 13:58:32 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Up
Apr  7 13:58:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:37 last:13:58:26 grace:0)
Apr  7 13:58:43 FMS-FreeSwitch sangoma_prid: Opening
/var/log/sangoma_pri/dchan_1.log
Apr  7 13:58:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:47 last:13:58:37 grace:0)
Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:57 last:13:58:47 grace:0)
Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
Apr  7 13:59:07 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:07 last:13:58:57 grace:0)
Apr  7 13:59:17 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:17 last:13:59:07 grace:0)
Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:27 last:13:59:17 grace:0)
Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: Assuming application is dead
Apr  7 13:59:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:37 last:13:59:27 grace:0)
Apr  7 13:59:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:47 last:13:59:37 grace:0)
Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:57 last:13:59:47 grace:0)
Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead

>From freeswitch cli, if I say
originate openzap/1/a/39114603 at g2 &park(), I got the following in the
sangoma_mgd.log.

Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Outgoing call (Smg-ID:2)
Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2:Outgoing call ChanRq:1
Called-Nb[39114603] Calling-Nb[Unknown] (Smg-ID:2)
Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c1:Remote
released-Unallocated (unassigned) number(1)
Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c0:Call was already cleared
(TSOFT-ID:4)
Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Call cleared (SMG-ID:2)

By seeing this, I confirmed that span2 is working. Leave out the Unallocated
number, that's my internal problem.

>From the extension, if I dial 0, nothing is printing in sangoma_mgd.log
But in dhcan2.log I got the No/Circuit or channel available as I mentioned
earlier.

In freeswitch mod_openzap is loaded properly.
Any help!!!


On Wed, Apr 7, 2010 at 2:17 AM, David Yat Sin <david.yatsin at gmail.com>wrote:

>  Hi Lakshmanan,
> If you do not have anything printing in /var/log/sangoma_mgd.log, and you
> have:
> verbose=4           //(or higher)
>
> in /etc/wanpipe/smg_pri.conf, check that you have these lines in
> /etc/syslog.conf (or /etc/rsyslog.conf):
>
> local2.*                /var/log/sangoma_mgd.log
>
> and restart your syslog.
>
>
> my first guess is that you do not have openzap loaded so sangoma_prid is
> rejecting all incoming calls, but I would need logs in
> /var/log/sangoma_mgd.log to confirm.
>
> If openzap is not loaded, you can type:
> load mod_openzap
>
> from the freeswitch CLI to load it.
>
> --
>
> *David Yat Sin, **BEng, **Software Developer**
> *Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 119 | e. *dyatsin at sangoma.com*
>
>
>
>
> On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
>
> Hi all,
> In my office we have a Hard PBX, with some 4 extensions.
> We also have sangoma A102 card.
> >From the Hard PBX, if 0 is pressed, it is setup in a way that it will go
> to outside world.
> I've connected that line to span2 of the card.
> The span2 in the A102 card, is configured as PRI_NET.
>
> wanrouter status shows connected for the span2.
>
> But if I dial from the extension, I got the following in the sangoma_dchan
> log.
> 2010-04-03 12:49:49
> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 33
> 39 31 31 34 36 30 30 7d 02 91 81 ]
>   Call Ref:0164
>   Type:Setup (0x5)
>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
> Kbit/s(16) L1Prot:G.711 A-Law(3)
>   Calling Party Number:4439114600(l:10) plan:isdn(1)
> type:unknown(0)scr:user, passed(1) pres:allowed(0)
>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>
> 2010-04-03 12:49:49
> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>   Call Ref:0164
>   Type:Release Compl (0x5a)
>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
> Circuit/Channel Available(34)
>
> The call in not reaching freeswitch ( I enabled debug log. But nothing is
> printing in it ).
> Can someone suggest how to make this work.
>
> Please ask me if you need more information?, since I don't know what to
> give now.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-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 anthony.minessale at gmail.com  Wed Apr  7 01:59:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Apr 2010 03:59:37 -0500
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <DA5E20C0FF09410798F9637759C7B8EE@sohobryan>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<735BB860-0DBE-440A-B924-0840D75B4112@freeswitch.org>
	<827F380A80A7450F92996F9D3A379B90@sohobryan>
	<1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
	<964CAAB8FB9F4F37B8632B0489F10287@sohobryan>
	<7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
	<A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>
	<3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>
	<4B224150-69C0-43D2-A6D5-58966F59E23E@freeswitch.org>
	<DA5E20C0FF09410798F9637759C7B8EE@sohobryan>
Message-ID: <n2q191c3a031004070159z2b03092dgf3d79a25ff8910eb@mail.gmail.com>

that mode you describe only works with polycom.

We are deprecating that mode in favor of broadsoft SCA


2010/4/6 Bryan <gliu at tecomtech.com>

>  Dear Brian,
>
>
>
> I have set
>
>
>
>     <param name="manage-presence" value="true"/>
>
>          <param name="manage-shared-appearance" value="sylantro"/>
>
>          <param name="multiple-registrations" value="contact"/>
>
>
>
> BR
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* 2010?4?7? 12:01
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re:
> [Freeswitch-users]SLA/BLAissue(sofia_presence.c:2139Gatewayinformation
> missing)
>
>
>
> I know it works if you enable manage-presence and what not... and setup the
> lines to subscribe correctly.
>
>
>
> /b
>
>
>
> On Apr 6, 2010, at 10:55 PM, Bryan wrote:
>
>
>
>  Do you mean FS just support Broadsoft?s SCA spec, no support for
> draft-anil-sipping-bla-03.txt.
>
>
>
> BTW, I use snom 360 phones, got the same result.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Wed Apr  7 02:19:55 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 7 Apr 2010 11:19:55 +0200
Subject: [Freeswitch-users]
	SLA/BLAissue(sofia_presence.c:2139Gatewayinformation missing)
In-Reply-To: <n2q191c3a031004070159z2b03092dgf3d79a25ff8910eb@mail.gmail.com>
References: <2E57E6EAFA8B48B998391BB90673B16A@sohobryan>
	<827F380A80A7450F92996F9D3A379B90@sohobryan>
	<1CA7836D-9499-4E26-83BD-2C4A083E1878@freeswitch.org>
	<964CAAB8FB9F4F37B8632B0489F10287@sohobryan>
	<7DC4E1D54F6842D7B819159A13D184F5@sohobryan>
	<A35E4D3D-DBFB-448B-8991-EA0884FC630D@freeswitch.org>
	<3CA0AADD762442B5A1F2B17DC25A44C6@sohobryan>
	<4B224150-69C0-43D2-A6D5-58966F59E23E@freeswitch.org>
	<DA5E20C0FF09410798F9637759C7B8EE@sohobryan>
	<n2q191c3a031004070159z2b03092dgf3d79a25ff8910eb@mail.gmail.com>
Message-ID: <r2k4468a6771004070219g2d139b5as5e7528cd42e5f6cc@mail.gmail.com>

There's sylantro mode in Linksys SPA 9xx too. I'll test it.



2010/4/7 Anthony Minessale <anthony.minessale at gmail.com>

> that mode you describe only works with polycom.
>
> We are deprecating that mode in favor of broadsoft SCA
>
>
> 2010/4/6 Bryan <gliu at tecomtech.com>
>
>>  Dear Brian,
>>
>>
>>
>> I have set
>>
>>
>>
>>     <param name="manage-presence" value="true"/>
>>
>>          <param name="manage-shared-appearance" value="sylantro"/>
>>
>>          <param name="multiple-registrations" value="contact"/>
>>
>>
>>
>> BR
>>
>>
>>  ------------------------------
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
>> *Sent:* 2010?4?7? 12:01
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re:
>> [Freeswitch-users]SLA/BLAissue(sofia_presence.c:2139Gatewayinformation
>> missing)
>>
>>
>>
>> I know it works if you enable manage-presence and what not... and setup
>> the lines to subscribe correctly.
>>
>>
>>
>> /b
>>
>>
>>
>> On Apr 6, 2010, at 10:55 PM, Bryan wrote:
>>
>>
>>
>>  Do you mean FS just support Broadsoft?s SCA spec, no support for
>> draft-anil-sipping-bla-03.txt.
>>
>>
>>
>> BTW, I use snom 360 phones, got the same result.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jayesh.voip at gmail.com  Wed Apr  7 05:17:43 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Wed, 7 Apr 2010 17:47:43 +0530
Subject: [Freeswitch-users] extract domain from SIP URI
Message-ID: <o2v4fe8b8271004070517r2b40ea28w25b84b0fb3e01b64@mail.gmail.com>

Hi,
Is it possible to identify the domain part in an incoming R-URI by using
some built-in variable in Freeswitch?
For eg. if a call comes into freeswitch as vmail at domainA.com,  can i use
something like "sip_h" to extract the domain part of the R-URI?

Thanks in advance,

w/regards,
Jayesh.
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From msc at freeswitch.org  Wed Apr  7 08:22:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Apr 2010 08:22:27 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call - Moises Silva From
	Sangoma Talks FreeTDM!
Message-ID: <x2y87f2f3b91004070822l898a4f5cqaf593693ee932050@mail.gmail.com>

Hello everyone!

Moises (moy) is going to be speaking today about FreeTDM. Please join the
conference call at noon Eastern, 9am Pacific time:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_07

Moises will start his presentation at about 12:15/9:15. Be sure to bring
your questions about TDM and Sangoma!

-Michael
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From david.yatsin at gmail.com  Wed Apr  7 10:14:27 2010
From: david.yatsin at gmail.com (David Yat Sin)
Date: Wed, 07 Apr 2010 13:14:27 -0400
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>	<4BBB9DEB.2040702@gmail.com>
	<v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
Message-ID: <4BBCBD73.7020206@gmail.com>

Can you provide me with SSH access to that box and a phone number I can 
use to trigger an incoming call on span 2 to that box?

You can email me at: dyatsin at sangoma.com

David

On 4/7/2010 4:48 AM, lakshmanan ganapathy wrote:
> Hi,
> I have set verbose=4 in smg_pri.conf.
> I started the wanrouter.
> Here is the sangoma_mgd.log
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: ================System 
> restart=============
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol 
> Stack Daemon       =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Version: 
> 1.63                           =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Date: Feb 26 
> 2010                       =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Wanpipe Release: 
> wanpipe-3.5.8.6       =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 
> 15607                =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: 
> ===========================================
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Number of spans:2
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Verbosity set to:4
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Debug disabled (local:2)
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Boost disabled (local:6)
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:1 pri_cpe 
> euroisdn dChan:16
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:2 pri_net 
> euroisdn dChan:16
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Down
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(P) 
> Local:127.0.0.66:53001 <http://127.0.0.66:53001> 
> Remote:127.0.0.65:53001 <http://127.0.0.65:53001>
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(N) 
> Local:127.0.0.66:53000 <http://127.0.0.66:53000> 
> Remote:127.0.0.65:53000 <http://127.0.0.65:53000>
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:Version:103
> Apr  7 13:58:27 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Up
> Apr  7 13:58:31 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Down
> Apr  7 13:58:32 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Up
> Apr  7 13:58:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:58:37 last:13:58:26 grace:0)
> Apr  7 13:58:43 FMS-FreeSwitch sangoma_prid: Opening 
> /var/log/sangoma_pri/dchan_1.log
> Apr  7 13:58:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:58:47 last:13:58:37 grace:0)
> Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:58:57 last:13:58:47 grace:0)
> Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> Apr  7 13:59:07 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:07 last:13:58:57 grace:0)
> Apr  7 13:59:17 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:17 last:13:59:07 grace:0)
> Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:27 last:13:59:17 grace:0)
> Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> Apr  7 13:59:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:37 last:13:59:27 grace:0)
> Apr  7 13:59:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:47 last:13:59:37 grace:0)
> Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout 
> (current:13:59:57 last:13:59:47 grace:0)
> Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
>
> From freeswitch cli, if I say
> originate openzap/1/a/39114603 at g2 &park(), I got the following in the 
> sangoma_mgd.log.
>
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Outgoing call (Smg-ID:2)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2:Outgoing call ChanRq:1 
> Called-Nb[39114603] Calling-Nb[Unknown] (Smg-ID:2)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c1:Remote 
> released-Unallocated (unassigned) number(1)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c0:Call was already 
> cleared (TSOFT-ID:4)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Call cleared (SMG-ID:2)
>
> By seeing this, I confirmed that span2 is working. Leave out the 
> Unallocated number, that's my internal problem.
>
> From the extension, if I dial 0, nothing is printing in sangoma_mgd.log
> But in dhcan2.log I got the No/Circuit or channel available as I 
> mentioned earlier.
>
> In freeswitch mod_openzap is loaded properly.
> Any help!!!
>
>
> On Wed, Apr 7, 2010 at 2:17 AM, David Yat Sin <david.yatsin at gmail.com 
> <mailto:david.yatsin at gmail.com>> wrote:
>
>     Hi Lakshmanan,
>     If you do not have anything printing in /var/log/sangoma_mgd.log,
>     and you have:
>     verbose=4           //(or higher)
>
>     in /etc/wanpipe/smg_pri.conf, check that you have these lines in
>     /etc/syslog.conf (or /etc/rsyslog.conf):
>
>     local2.*                /var/log/sangoma_mgd.log
>
>     and restart your syslog.
>
>
>     my first guess is that you do not have openzap loaded so
>     sangoma_prid is rejecting all incoming calls, but I would need
>     logs in /var/log/sangoma_mgd.log to confirm.
>
>     If openzap is not loaded, you can type:
>     load mod_openzap
>
>     from the freeswitch CLI to load it.
>
>     -- 
>
>     *David Yat Sin, **BEng, **Software Developer**
>     *Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham
>     ON L3R 9T3 Canada
>     t. 1 905 474 1990 x 119 | e. _dyatsin at sangoma.com
>     <mailto:dyatsin at sangoma.com>_
>
>
>
>     On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
>>     Hi all,
>>     In my office we have a Hard PBX, with some 4 extensions.
>>     We also have sangoma A102 card.
>>     >From the Hard PBX, if 0 is pressed, it is setup in a way that it
>>     will go to outside world.
>>     I've connected that line to span2 of the card.
>>     The span2 in the A102 card, is configured as PRI_NET.
>>
>>     wanrouter status shows connected for the span2.
>>
>>     But if I dial from the extension, I got the following in the
>>     sangoma_dchan log.
>>     2010-04-03 12:49:49
>>     INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81
>>     34 34 33 39 31 31 34 36 30 30 7d 02 91 81 ]
>>       Call Ref:0164
>>       Type:Setup (0x5)
>>       Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0)
>>     TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3)
>>       Calling Party Number:4439114600(l:10) plan:isdn(1)
>>     type:unknown(0)scr:user, passed(1) pres:allowed(0)
>>       High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>>
>>     2010-04-03 12:49:49
>>     OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>>       Call Ref:0164
>>       Type:Release Compl (0x5a)
>>       Cause:coding:ITU-T(0) location:Public network, local user(2)
>>     val:No Circuit/Channel Available(34)
>>
>>     The call in not reaching freeswitch ( I enabled debug log. But
>>     nothing is printing in it ).
>>     Can someone suggest how to make this work.
>>
>>     Please ask me if you need more information?, since I don't know
>>     what to give now.
>>
>>
>>     _______________________________________________
>>     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 oseslija at gmail.com  Wed Apr  7 13:15:21 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 7 Apr 2010 22:15:21 +0200
Subject: [Freeswitch-users] extract domain from SIP URI
In-Reply-To: <o2v4fe8b8271004070517r2b40ea28w25b84b0fb3e01b64@mail.gmail.com>
References: <o2v4fe8b8271004070517r2b40ea28w25b84b0fb3e01b64@mail.gmail.com>
Message-ID: <s2s4468a6771004071315t17ce80afq533cb71883bc4e45@mail.gmail.com>

Yes.
 I suggest adding  <action application="info"/> line in the dialplan and
check its output in console. You'll find many sip_* vars.


On Wed, Apr 7, 2010 at 2:17 PM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi,
> Is it possible to identify the domain part in an incoming R-URI by using
> some built-in variable in Freeswitch?
> For eg. if a call comes into freeswitch as vmail at domainA.com,  can i use
> something like "sip_h" to extract the domain part of the R-URI?
>
> Thanks in advance,
>
> w/regards,
> Jayesh.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jasonchu at avaya.com  Wed Apr  7 11:03:16 2010
From: jasonchu at avaya.com (CHU, XINGJUN (XINGJUN))
Date: Wed, 7 Apr 2010 14:03:16 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
Message-ID: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>

Hi,

I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.

Now I got a problem when A calls B then attended transfer B to freeswitch,
The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.

Any suggestions are greatly appreciated.

Thanks
Xingjun


From gianluca.varisco at privatewave.com  Wed Apr  7 09:06:39 2010
From: gianluca.varisco at privatewave.com (Gianluca Varisco)
Date: Wed, 07 Apr 2010 18:06:39 +0200
Subject: [Freeswitch-users] Join XML dialplan configuration (local/XML_CURL)
Message-ID: <4BBCAD8F.8020701@privatewave.com>

Hello guys,

I tried to set up a brand new installation of FS 1.0.6 so that it reads 
dialplan configs from XML_CURL module and from local files. No troubles 
so far with it.

I neeed to know how to set my freeswitch configuration in a way that my 
dialplan takes a common part from the local files and a user specific 
part from an external URI (via XML_CURL). At the moment I've a working 
configuration with both local and XML_CURL correct responses.

Any suggestion on ways to join the two parts is appreciated.

Many thanks.

Gianluca


From c.hiller at baigtel.com  Wed Apr  7 02:53:02 2010
From: c.hiller at baigtel.com (Christian Hiller - Baig Tel LTD)
Date: Wed, 07 Apr 2010 11:53:02 +0200
Subject: [Freeswitch-users] mod_dahdi_codec compile
Message-ID: <4BBC55FE.8020208@baigtel.com>

Hello,

i would like to use mod_dahdi_codec, but when compiling from trunk it 
doesnt show up in /usr/local/freeswitch/mod
What options are nesesarry?

chris


From msc at freeswitch.org  Wed Apr  7 13:25:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Apr 2010 13:25:37 -0700
Subject: [Freeswitch-users] Help to call a VXML SIP number
In-Reply-To: <y2g1452e2981004062333vc6da46b0s4a7249142cfe09b@mail.gmail.com>
References: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
	<539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>
	<y2g1452e2981004062333vc6da46b0s4a7249142cfe09b@mail.gmail.com>
Message-ID: <m2y87f2f3b91004071325ma133f4d4x2ad9c9ce1c18ceae@mail.gmail.com>

So you sent INVITEs out and nothing came back? Smells like NAT or a SIP ALG
in the firewall...
-MC

On Tue, Apr 6, 2010 at 11:33 PM, velusamy Krishnan <velu.technical at gmail.com
> wrote:

> Hi,
>    I have enabled the siptrace in console. I have found that only SIP
> INVITE packets came in the console. The response packet didn't come.
>    Could you anyone help me to know what is the problem??
>
>
> On Tue, Apr 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> NO_ANSWER means the call was successful, but noone picked it up. Try
>> setting debug on (console loglevel debug) and watch for the call progress on
>> the console.
>>
>> Additionally you may enable sip-tracing so you can see all the packets on
>> the console: sofia profile external siptrace on
>>
>>  Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 2010-04-06, at 5:44 AM, velusamy Krishnan wrote:
>>
>> Dear All,
>>      I have received the XXX at sip.voxeo.net from my VXML application. I
>> have written the following dial plan to call that number.
>>   <extension name="VXML">
>>         <condition field="destination_number" expression="^(999)$">
>>                 <action application="bridge" data="sofia/external/
>> 9996079973 at sip.voxeo.net"/>
>>                 <action application="hangup"/>
>>         </condition>
>>   </extension>
>>
>> I have registered the FreeSWITCH and called the 999 extension. But the
>> call was just trying. Finally I have got the NO_ANSWER information in
>> console.
>>   2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate Failed.
>> Cause: NO_ANSWER
>>
>> Is there configuration I need to do to call this number?? 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Wed Apr  7 13:30:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 7 Apr 2010 13:30:06 -0700
Subject: [Freeswitch-users] Setting "accountcode" Dynamically
Message-ID: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>


Regarding the "accountcode" channel variable in FS:
http://wiki.freeswitch.org/wiki/Variable_accountcode.  Is there any example
or method of dynamically setting this accountcode for a the call has already
been established?  This would apply for a case where a call is received and
the callee would like to set the accountcode during the conversation.

Thanks,
Jerry



From phunk0000 at hotmail.com  Wed Apr  7 13:30:58 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 7 Apr 2010 16:30:58 -0400
Subject: [Freeswitch-users] Call Screener
Message-ID: <BLU110-DS1767C1740CBF59AC9C2346A7170@phx.gbl>

Hey List-  I have a fresh FS install on Centos 5.  I am trying to use the
limit_hash application to screen caller ID numbers before transferring calls
to another call server on my  internal network.

 

<action application="limit_hash" data="cidlimit ${caller_id_number} 50/1" />

<action application="transfer" data="sip/sofia/192.168.15.2" />

 

Is this the correct/best way to transfer all calls to another SIP server
that will be managing all calls? Where is the best place to put this
application so that it will run on every incoming call from several SIP
providers?  I have installed all the sample data, and have several phones
registered to extensions that can make internal calls.  I am just looking
for the general outline of where these applications should go after I have
setup the several SIP gateways for my external providers.  I am still noo
and would greatly appreciate any advice as to the best way to configure FS
to simply receive all calls from the outside, run the limit_hash, then
transfer to another SIP server.  Thanks List!
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From mrene_lists at avgs.ca  Wed Apr  7 13:32:33 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 7 Apr 2010 16:32:33 -0400
Subject: [Freeswitch-users] mod_dahdi_codec compile
In-Reply-To: <4BBC55FE.8020208@baigtel.com>
References: <4BBC55FE.8020208@baigtel.com>
Message-ID: <89018CB0-7492-4096-ABFC-9230DAE7154E@avgs.ca>

Add it to modules.conf in your build directory

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-07, at 5:53 AM, Christian Hiller - Baig Tel LTD wrote:

> Hello,
>
> i would like to use mod_dahdi_codec, but when compiling from trunk it
> doesnt show up in /usr/local/freeswitch/mod
> What options are nesesarry?
>
> 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



From msc at freeswitch.org  Wed Apr  7 13:35:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Apr 2010 13:35:39 -0700
Subject: [Freeswitch-users] mod_dahdi_codec compile
In-Reply-To: <4BBC55FE.8020208@baigtel.com>
References: <4BBC55FE.8020208@baigtel.com>
Message-ID: <h2j87f2f3b91004071335uf73fc294la9f34b400e85f679@mail.gmail.com>

I just did this on a CentOS machine 2 minutes ago and it worked:
Edit modules.conf and enable "codecs/mod_dahdi_codec"
Build it: make mod_dahdi_codec-install

-MC

On Wed, Apr 7, 2010 at 2:53 AM, Christian Hiller - Baig Tel LTD <
c.hiller at baigtel.com> wrote:

> Hello,
>
> i would like to use mod_dahdi_codec, but when compiling from trunk it
> doesnt show up in /usr/local/freeswitch/mod
> What options are nesesarry?
>
> 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 brian at freeswitch.org  Wed Apr  7 13:36:09 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Apr 2010 15:36:09 -0500
Subject: [Freeswitch-users] Setting "accountcode" Dynamically
In-Reply-To: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
References: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
Message-ID: <1B39B1F0-8320-4A7D-B5D9-FE1AA0FA7EE4@freeswitch.org>

account code is arbitrary .. you can set any variable you want...

/b

On Apr 7, 2010, at 3:30 PM, Jerry Richards wrote:

> 
> Regarding the "accountcode" channel variable in FS:
> http://wiki.freeswitch.org/wiki/Variable_accountcode.  Is there any example
> or method of dynamically setting this accountcode for a the call has already
> been established?  This would apply for a case where a call is received and
> the callee would like to set the accountcode during the conversation.
> 
> Thanks,
> Jerry

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From msc at freeswitch.org  Wed Apr  7 13:39:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Apr 2010 13:39:09 -0700
Subject: [Freeswitch-users] Setting "accountcode" Dynamically
In-Reply-To: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
References: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
Message-ID: <t2o87f2f3b91004071339v274e0f6g6698749295c7b3da@mail.gmail.com>

You could do bind_meta_app to execute_extension and have the extension ask
the user for an account code and set it there. I'm curious to see how well
that would work. You could copy the basic structure of the Local_Extension
in default.xml and the corresponding entries in features.xml and then write
your own extension that does the dirty work of prompting the user, accepting
the digits, and then changing the accountcode chan var.

-MC

On Wed, Apr 7, 2010 at 1:30 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Regarding the "accountcode" channel variable in FS:
> http://wiki.freeswitch.org/wiki/Variable_accountcode.  Is there any
> example
> or method of dynamically setting this accountcode for a the call has
> already
> been established?  This would apply for a case where a call is received and
> the callee would like to set the accountcode during the conversation.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From oseslija at gmail.com  Wed Apr  7 13:45:13 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 7 Apr 2010 22:45:13 +0200
Subject: [Freeswitch-users] Setting "accountcode" Dynamically
In-Reply-To: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
References: <C14EE890E0CA4AC3ACC3F975DD8DA175@greyhawk.tonecommander.com>
Message-ID: <l2s4468a6771004071345n49354bbdve2b3f5fc7e283a9f@mail.gmail.com>

There's CMC soft key on Snom phones to do that. FS supports it
out-of-the-box.


On Wed, Apr 7, 2010 at 10:30 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Regarding the "accountcode" channel variable in FS:
> http://wiki.freeswitch.org/wiki/Variable_accountcode.  Is there any
> example
> or method of dynamically setting this accountcode for a the call has
> already
> been established?  This would apply for a case where a call is received and
> the callee would like to set the accountcode during the conversation.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 carmickle.com  Wed Apr  7 14:03:14 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Wed, 7 Apr 2010 17:03:14 -0400
Subject: [Freeswitch-users] Call Screener
In-Reply-To: <BLU110-DS1767C1740CBF59AC9C2346A7170@phx.gbl>
References: <BLU110-DS1767C1740CBF59AC9C2346A7170@phx.gbl>
Message-ID: <20100407210314.GC18770@base.carmickle.com>

On Wed, Apr 07, Todd wrote:
> Hey List-  I have a fresh FS install on Centos 5.  I am trying to use the
> limit_hash application to screen caller ID numbers before transferring calls
> to another call server on my  internal network.

Just screen calls?  Limit and limit_hash are for tracking how many calls / how many calls per second.  Use the dialplan if you just want to allow certain numbers through.

Tell us the functionality you want and we can make recommendations.
--FC


> 
>  
> 
> <action application="limit_hash" data="cidlimit ${caller_id_number} 50/1" />
> 
> <action application="transfer" data="sip/sofia/192.168.15.2" />
> 
>  
> 
> Is this the correct/best way to transfer all calls to another SIP server
> that will be managing all calls? Where is the best place to put this
> application so that it will run on every incoming call from several SIP
> providers?  I have installed all the sample data, and have several phones
> registered to extensions that can make internal calls.  I am just looking
> for the general outline of where these applications should go after I have
> setup the several SIP gateways for my external providers.  I am still noo
> and would greatly appreciate any advice as to the best way to configure FS
> to simply receive all calls from the outside, run the limit_hash, then
> transfer to another SIP server.  Thanks List!

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



From vetali100 at gmail.com  Wed Apr  7 14:17:37 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Thu, 8 Apr 2010 00:17:37 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and then
	answer a call
Message-ID: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>

Hi dear community,

I am using a Lua script that is being executed when a call reaches a
particular extension, say 1001.
It works ok, but it answers immediately when call reaches the system.

How can I make it to wait 5-10 seconds (so the caller will hear several
ringtones) and only after that the Lua script should answer and start the
processing?

Thank you,
Vitalie
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From phunk0000 at hotmail.com  Wed Apr  7 14:27:09 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 7 Apr 2010 17:27:09 -0400
Subject: [Freeswitch-users] Call Screener
In-Reply-To: <20100407210314.GC18770@base.carmickle.com>
References: <BLU110-DS1767C1740CBF59AC9C2346A7170@phx.gbl>
	<20100407210314.GC18770@base.carmickle.com>
Message-ID: <BLU110-DS11DED2B1E88CAD896432A6A7170@phx.gbl>

What I need it to do is this:  I have apprx. 100,000 numbers from across the
country provided by eTollFree, PacWest, XO, Global Pop, and Vitelity.  In
order to prevent call bomb DoS attacks I need FS to screen the CID of every
incoming call and BlackList or Block any CID's that violate a threshold of
20 call per minute or day even.  I was thinking that limit_hash 

<action application="limit_hash" data="cidlimit ${caller_id_number} 20/60"
/>

Would block any calls from the same CID above 20 in a one minute period.  If
there is a better way to setup FS to examine every incoming call and
BlackList CID's that violate threshold parameters that you could tell me
that would be great.  Thanks!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank
Carmickle
Sent: Wednesday, April 07, 2010 5:03 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call Screener

On Wed, Apr 07, Todd wrote:
> Hey List-  I have a fresh FS install on Centos 5.  I am trying to use the
> limit_hash application to screen caller ID numbers before transferring
calls
> to another call server on my  internal network.

Just screen calls?  Limit and limit_hash are for tracking how many calls /
how many calls per second.  Use the dialplan if you just want to allow
certain numbers through.

Tell us the functionality you want and we can make recommendations.
--FC


> 
>  
> 
> <action application="limit_hash" data="cidlimit ${caller_id_number} 50/1"
/>
> 
> <action application="transfer" data="sip/sofia/192.168.15.2" />
> 
>  
> 
> Is this the correct/best way to transfer all calls to another SIP server
> that will be managing all calls? Where is the best place to put this
> application so that it will run on every incoming call from several SIP
> providers?  I have installed all the sample data, and have several phones
> registered to extensions that can make internal calls.  I am just looking
> for the general outline of where these applications should go after I have
> setup the several SIP gateways for my external providers.  I am still noo
> and would greatly appreciate any advice as to the best way to configure FS
> to simply receive all calls from the outside, run the limit_hash, then
> transfer to another SIP server.  Thanks List!

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


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

No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.800 / Virus Database: 271.1.1/2779 - Release Date: 04/07/10
02:32:00



From nandy1925 at gmail.com  Wed Apr  7 15:37:08 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 8 Apr 2010 06:37:08 +0800
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
Message-ID: <o2m7d0bfd8c1004071537p198d3d32l2306ff5401c3bcbd@mail.gmail.com>

hi vitalii,

just add this line into your Lua script:
session:execute("sleep","5000")

i hope this is what you need.
-nandy


On Thu, Apr 8, 2010 at 5:17 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Wed Apr  7 21:00:35 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 8 Apr 2010 07:00:35 +0300
Subject: [Freeswitch-users] Causing a Polycom phone to dial by a remote
	instruction
Message-ID: <w2sd361ca7b1004072100q2ee64eb0r772500dd5170ec99@mail.gmail.com>

Hello,

  I would like to offer our users a WEB interface to manage their calls and
initiate a call through this user interface. I can create two call legs via
FreeSwitch and bridge them together; however, the call would show on the
phone as an incoming call and not outgoing call.

  Is there a way to command a Polycom from remote to dial a number?

                                   Thanks! __Yehavi:
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From david.ponzone at gmail.com  Wed Apr  7 22:17:24 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 8 Apr 2010 07:17:24 +0200
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
Message-ID: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>

Perhaps:
pre_	answer
then
sleep ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :

> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a  
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear  
> several ringtones) and only after that the Lua script should answer  
> and start the processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From velu.technical at gmail.com  Wed Apr  7 22:28:33 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Thu, 8 Apr 2010 10:58:33 +0530
Subject: [Freeswitch-users] Help to call a VXML SIP number
In-Reply-To: <m2y87f2f3b91004071325ma133f4d4x2ad9c9ce1c18ceae@mail.gmail.com>
References: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
	<539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>
	<y2g1452e2981004062333vc6da46b0s4a7249142cfe09b@mail.gmail.com>
	<m2y87f2f3b91004071325ma133f4d4x2ad9c9ce1c18ceae@mail.gmail.com>
Message-ID: <u2y1452e2981004072228haf2a8658wbd99fc7714f2a5b2@mail.gmail.com>

Hi,
     I have found that  NAT has been enabled in my Firewall. But I don't
know how to solve this problem. Could you help me please..

On Thu, Apr 8, 2010 at 1:55 AM, Michael Collins <msc at freeswitch.org> wrote:

> So you sent INVITEs out and nothing came back? Smells like NAT or a SIP ALG
> in the firewall...
> -MC
>
>
> On Tue, Apr 6, 2010 at 11:33 PM, velusamy Krishnan <
> velu.technical at gmail.com> wrote:
>
>> Hi,
>>    I have enabled the siptrace in console. I have found that only SIP
>> INVITE packets came in the console. The response packet didn't come.
>>    Could you anyone help me to know what is the problem??
>>
>>
>> On Tue, Apr 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>>> NO_ANSWER means the call was successful, but noone picked it up. Try
>>> setting debug on (console loglevel debug) and watch for the call progress on
>>> the console.
>>>
>>> Additionally you may enable sip-tracing so you can see all the packets on
>>> the console: sofia profile external siptrace on
>>>
>>>  Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 2010-04-06, at 5:44 AM, velusamy Krishnan wrote:
>>>
>>> Dear All,
>>>      I have received the XXX at sip.voxeo.net from my VXML application. I
>>> have written the following dial plan to call that number.
>>>   <extension name="VXML">
>>>         <condition field="destination_number" expression="^(999)$">
>>>                 <action application="bridge" data="sofia/external/
>>> 9996079973 at sip.voxeo.net"/>
>>>                 <action application="hangup"/>
>>>         </condition>
>>>   </extension>
>>>
>>> I have registered the FreeSWITCH and called the 999 extension. But the
>>> call was just trying. Finally I have got the NO_ANSWER information in
>>> console.
>>>   2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate Failed.
>>> Cause: NO_ANSWER
>>>
>>> Is there configuration I need to do to call this number?? 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Apr  7 23:54:27 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 8 Apr 2010 16:54:27 +1000
Subject: [Freeswitch-users] version number: git checkout
Message-ID: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>

Hi!

I just used git for the first time ever to checkout FreeSwitch as
described on the wiki at
http://wiki.freeswitch.org/wiki/Installation_Guide

Now my version number says:
FreeSWITCH Version 1.0.head (git-)

Is there a mistake in my procedure or the building of FS when using
GIT?  Hard to know the build number of FS with a tag like that!


From wasim at convergence.pk  Thu Apr  8 00:01:57 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Thu, 8 Apr 2010 12:01:57 +0500
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
Message-ID: <p2rb8ad2a5b1004080001r7bff6d0et7c47fbbff9c2fe12@mail.gmail.com>

On Thu, Apr 8, 2010 at 11:54, Mark Campbell-Smith
<mcampbellsmith at gmail.com>wrote:

> Hi!
>
> I just used git for the first time ever to checkout FreeSwitch as
> described on the wiki at
> http://wiki.freeswitch.org/wiki/Installation_Guide
>
> Now my version number says:
> FreeSWITCH Version 1.0.head (git-)
>
> Is there a mistake in my procedure or the building of FS when using
> GIT?  Hard to know the build number of FS with a tag like that!
>

freeswitch at de> version

FreeSWITCH Version 1.0.head (git-25b3b4d 2010-04-08 00:23:06 -0300)

i just did a fresh checkout

-wasim


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



-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From djbinter at yahoo.com  Thu Apr  8 00:14:48 2010
From: djbinter at yahoo.com (DJB)
Date: Thu, 8 Apr 2010 00:14:48 -0700 (PDT)
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
Message-ID: <738759.37967.qm@web37505.mail.mud.yahoo.com>

Upgrade your git, then it will show it correctly.
http://wiki.freeswitch.org/wiki/Git_Install

djbinter



________________________________
From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, April 7, 2010 11:54:27 PM
Subject: [Freeswitch-users] version number: git checkout

Hi!

I just used git for the first time ever to checkout FreeSwitch as
described on the wiki at
http://wiki.freeswitch.org/wiki/Installation_Guide

Now my version number says:
FreeSWITCH Version 1.0.head (git-)

Is there a mistake in my procedure or the building of FS when using
GIT?  Hard to know the build number of FS with a tag like that!

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



      
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From b_ball_henry at hotmail.com  Thu Apr  8 01:15:32 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Thu, 8 Apr 2010 16:15:32 +0800
Subject: [Freeswitch-users] xml_odbc module
Message-ID: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>

I just downloaded the 1.0.6 version and some previous trunk versions, but I
can't seem to be able to locate xml_odbc source code anymore.

Our SIP registration is utilizing it for realtime user adding.... Can anyone
tell me what happen to that module? or it's just been forgotten to put on?

thanks

-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From mayamatakeshi at gmail.com  Thu Apr  8 01:53:11 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 8 Apr 2010 17:53:11 +0900
Subject: [Freeswitch-users] xml_odbc module
In-Reply-To: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
References: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
Message-ID: <q2q15b9404e1004080153sb30d397cs27ff569926e158e0@mail.gmail.com>

On Thu, Apr 8, 2010 at 5:15 PM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> I just downloaded the 1.0.6 version and some previous trunk versions, but I
> can't seem to be able to locate xml_odbc source code anymore.
>
> Our SIP registration is utilizing it for realtime user adding.... Can
> anyone tell me what happen to that module? or it's just been forgotten to
> put on?
>

It will be eventually available again:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055679.html

Have you tried getting it with git? Maybe it is already available from there
as someone told svn repo is some revisions behind (it stalled on revision
17188 some days ago).
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From klejch+freeswitch at netbox.cz  Thu Apr  8 02:03:36 2010
From: klejch+freeswitch at netbox.cz (Vladimir Klejch)
Date: Thu, 8 Apr 2010 11:03:36 +0200 (CEST)
Subject: [Freeswitch-users] xml_odbc module
In-Reply-To: <q2q15b9404e1004080153sb30d397cs27ff569926e158e0@mail.gmail.com>
References: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
	<q2q15b9404e1004080153sb30d397cs27ff569926e158e0@mail.gmail.com>
Message-ID: <alpine.DEB.2.00.1004081102170.5009@crathva.argobk-va.pm>

Hi

its already in git in contrib repo

git clone git://git.freeswitch.org/freeswitch-contrib.git

  in mod/xml_int/mod_xml_odbc



 				Kleo

On Thu, 8 Apr 2010, mayamatakeshi wrote:

> On Thu, Apr 8, 2010 at 5:15 PM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> I just downloaded the 1.0.6 version and some previous trunk versions, but I
>> can't seem to be able to locate xml_odbc source code anymore.
>>
>> Our SIP registration is utilizing it for realtime user adding.... Can
>> anyone tell me what happen to that module? or it's just been forgotten to
>> put on?
>>
>
> It will be eventually available again:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055679.html
>
> Have you tried getting it with git? Maybe it is already available from there
> as someone told svn repo is some revisions behind (it stalled on revision
> 17188 some days ago).
>

-- 
 	klejch+freeswitch at netbox.cz
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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

From leon at scarlet-internet.nl  Thu Apr  8 02:07:03 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 8 Apr 2010 11:07:03 +0200
Subject: [Freeswitch-users] xml_odbc module
In-Reply-To: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
References: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
Message-ID: <703C2011-872D-4B06-AB04-945815516C89@scarlet-internet.nl>

Hi,

It has moved to a seperate repository: freeswitch-contrib.. I did the  
following to get the same tree as before on svn:

cd /usr/src
git clone git://git.freeswitch.org/freeswitch.git
cd freeswitch
git clone git://git.freeswitch.org/freeswitch-contrib.git contrib

(don't know if this is the best way, but it works)

Kind regards,

Leon


On Apr 8, 2010, at 10:15 AM, Henry Huang wrote:

> I just downloaded the 1.0.6 version and some previous trunk  
> versions, but I can't seem to be able to locate xml_odbc source code  
> anymore.
>
> Our SIP registration is utilizing it for realtime user adding....  
> Can anyone tell me what happen to that module? or it's just been  
> forgotten to put on?
>
> thanks
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From red.rain.seven at gmail.com  Thu Apr  8 02:23:02 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 8 Apr 2010 17:23:02 +0800
Subject: [Freeswitch-users] xml_odbc module
In-Reply-To: <703C2011-872D-4B06-AB04-945815516C89@scarlet-internet.nl>
References: <u2t59ad9ca11004080115ga1f081e7iae528f37135e9b92@mail.gmail.com>
	<703C2011-872D-4B06-AB04-945815516C89@scarlet-internet.nl>
Message-ID: <k2r59ad9ca11004080223ue4676693y605a358f236f6661@mail.gmail.com>

Thanks

It does work, and it save us a round trip of setting up a web server to work
with xml_curl_odbc

On Thu, Apr 8, 2010 at 5:07 PM, Leon de Rooij <leon at scarlet-internet.nl>wrote:

> Hi,
>
> It has moved to a seperate repository: freeswitch-contrib.. I did the
> following to get the same tree as before on svn:
>
> cd /usr/src
> git clone git://git.freeswitch.org/freeswitch.git
> cd freeswitch
> git clone git://git.freeswitch.org/freeswitch-contrib.git contrib
>
> (don't know if this is the best way, but it works)
>
> Kind regards,
>
> Leon
>
>
> On Apr 8, 2010, at 10:15 AM, Henry Huang wrote:
>
> > I just downloaded the 1.0.6 version and some previous trunk
> > versions, but I can't seem to be able to locate xml_odbc source code
> > anymore.
> >
> > Our SIP registration is utilizing it for realtime user adding....
> > Can anyone tell me what happen to that module? or it's just been
> > forgotten to put on?
> >
> > thanks
> >
> > --
> > Henry Huang
> > UniC Solution - Communication Unified
> > VoIP & Open Source software Consultant
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From mcampbellsmith at gmail.com  Thu Apr  8 02:52:29 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 8 Apr 2010 19:52:29 +1000
Subject: [Freeswitch-users] fatal: unrecognized argument: --format=%h %ci
Message-ID: <h2i33c87fa31004080252v1dc9b0b2nd53ddabe8e0080a6@mail.gmail.com>

Hi!

I just performed a 'git pull %% make install' and noticed the following:

Compiling src/switch_apr.c ...
Compiling src/switch_buffer.c ...
Compiling src/switch_caller.c ...
Compiling src/switch_channel.c ...
fatal: unrecognized argument: --format=%h %ci
Compiling src/switch_console.c ...
Compiling src/switch_mprintf.c ...
Compiling src/switch_core_media_bug.c ...

is the fatal argument anything to worry about?

Thanks


From david.ponzone at gmail.com  Thu Apr  8 03:04:00 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 8 Apr 2010 12:04:00 +0200
Subject: [Freeswitch-users] fatal: unrecognized argument: --format=%h %ci
In-Reply-To: <h2i33c87fa31004080252v1dc9b0b2nd53ddabe8e0080a6@mail.gmail.com>
References: <h2i33c87fa31004080252v1dc9b0b2nd53ddabe8e0080a6@mail.gmail.com>
Message-ID: <4D25B974-B331-41BE-B7B0-4A7AC8E4AA19@gmail.com>

As far as I know, nothing to worry about.
I think it's related to git.
I suspect you have an old version (coming from a repo package) ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 08/04/2010 ? 11:52, Mark Campbell-Smith a ?crit :

> Hi!
>
> I just performed a 'git pull %% make install' and noticed the  
> following:
>
> Compiling src/switch_apr.c ...
> Compiling src/switch_buffer.c ...
> Compiling src/switch_caller.c ...
> Compiling src/switch_channel.c ...
> fatal: unrecognized argument: --format=%h %ci
> Compiling src/switch_console.c ...
> Compiling src/switch_mprintf.c ...
> Compiling src/switch_core_media_bug.c ...
>
> is the fatal argument anything to worry about?
>
> 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 mcampbellsmith at gmail.com  Thu Apr  8 06:01:48 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 8 Apr 2010 23:01:48 +1000
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <738759.37967.qm@web37505.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
Message-ID: <s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>

Git was installed as described on the wiki.  I am using Debian Lenny
and Git version 1.5.6.5

I just did a git pull and had the same issue... FS still shows
FreeSWITCH Version 1.0.head (git-)



On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> Upgrade your git, then it will show it correctly.
> http://wiki.freeswitch.org/wiki/Git_Install
> djbinter
> ________________________________
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, April 7, 2010 11:54:27 PM
> Subject: [Freeswitch-users] version number: git checkout
>
> Hi!
>
> I just used git for the first time ever to checkout FreeSwitch as
> described on the wiki at
> http://wiki.freeswitch.org/wiki/Installation_Guide
>
> Now my version number says:
> FreeSWITCH Version 1.0.head (git-)
>
> Is there a mistake in my procedure or the building of FS when using
> GIT?? Hard to know the build number of FS with a tag like that!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mayamatakeshi at gmail.com  Thu Apr  8 06:33:47 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 8 Apr 2010 22:33:47 +0900
Subject: [Freeswitch-users] git clone via http
Message-ID: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>

Is there any chance of getting freeswitch using git thru http?
I need to go to internet thru a proxy so I cannot access it (unless I
install some sort of git_proxy).
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From testeador01 at gmail.com  Thu Apr  8 06:51:13 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 8 Apr 2010 09:51:13 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
Message-ID: <w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>

Can anything be done in the freeswitch code so when "git pull" is executed,
the "--pretty" argument is also set where "--format" is set to make it
compatible with both older and newer versions of git? or it is all up to
what git does and nothing to do on fs?




PS: Mark, the issue you're facing is because of your version of git, the CLI
shows the freeswitch version properly with git 1.7.0.4, the "format"
argument isn't recognized by your version of git.



2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>

> Git was installed as described on the wiki.  I am using Debian Lenny
> and Git version 1.5.6.5
>
> I just did a git pull and had the same issue... FS still shows
> FreeSWITCH Version 1.0.head (git-)
>
>
>
> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> > Upgrade your git, then it will show it correctly.
> > http://wiki.freeswitch.org/wiki/Git_Install
> > djbinter
> > ________________________________
> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> > To: freeswitch-users at lists.freeswitch.org
> > Sent: Wed, April 7, 2010 11:54:27 PM
> > Subject: [Freeswitch-users] version number: git checkout
> >
> > Hi!
> >
> > I just used git for the first time ever to checkout FreeSwitch as
> > described on the wiki at
> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >
> > Now my version number says:
> > FreeSWITCH Version 1.0.head (git-)
> >
> > Is there a mistake in my procedure or the building of FS when using
> > GIT?  Hard to know the build number of FS with a tag like that!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From testeador01 at gmail.com  Thu Apr  8 06:56:43 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 8 Apr 2010 09:56:43 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
Message-ID: <h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>

hi,

put an <action application="sleep" data="2000"/> before the instruction to
play something


2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>

> Hi,
>
> I have Freeswitch and other two SIP sets all registered with a SIP
> proxy/registry, and I am writing a control software to  control the
> freeswitch via event socket. Baisically the control software tells
> freeswitch when someone calls it, what action to take, for example, acts as
> Auto attendant and play prompt. Etc.
>
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the
> trailing part of the prompt played to A when A called freeswitch in the
> beginning of the transfer.  I understand that's because nobody tell
> freeswitch to stop and start from the beginning.  I am looking for what
> event should the control software monitor for the replaced session (the
> transfer is done via "invite replaces" ) and how to cancel the current
> prompt and start from beginning.
>
> Any suggestions are greatly appreciated.
>
> Thanks
> Xingjun
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Thu Apr  8 08:59:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 8 Apr 2010 11:59:34 -0400
Subject: [Freeswitch-users] Reload Configurations
Message-ID: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>

Hi All,

I add some SIP trunks to FreeSWITCH.

reloadxml

reload mod_sofia  - Not allow to load the configuration when there are
active calls.

How to load the configurations when there are active calls in the system.

Thanks
Lloyd
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From brian at freeswitch.org  Thu Apr  8 09:05:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Apr 2010 11:05:09 -0500
Subject: [Freeswitch-users] Reload Configurations
In-Reply-To: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
Message-ID: <D3C25AC7-06D9-443C-8E1E-31BECC4515E1@freeswitch.org>

.

sofia profile xxx rescan


/b

On Apr 8, 2010, at 10:59 AM, Aloysius Lloyd wrote:

> Hi All,
> 
> I add some SIP trunks to FreeSWITCH.
> 
> reloadxml
> 
> reload mod_sofia  - Not allow to load the configuration when there are active calls. 
> 
> How to load the configurations when there are active calls in the system.
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Thu Apr  8 09:24:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 8 Apr 2010 12:24:34 -0400
Subject: [Freeswitch-users] Reload Configurations
In-Reply-To: <D3C25AC7-06D9-443C-8E1E-31BECC4515E1@freeswitch.org>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
	<D3C25AC7-06D9-443C-8E1E-31BECC4515E1@freeswitch.org>
Message-ID: <w2h8a19bf2e1004080924je77f989pec8b88edf9588b96@mail.gmail.com>

Thanks
Lloyd


On Thu, Apr 8, 2010 at 12:05 PM, Brian West <brian at freeswitch.org> wrote:

> .
>
> sofia profile xxx rescan
>
>
> /b
>
> On Apr 8, 2010, at 10:59 AM, Aloysius Lloyd wrote:
>
> > Hi All,
> >
> > I add some SIP trunks to FreeSWITCH.
> >
> > reloadxml
> >
> > reload mod_sofia  - Not allow to load the configuration when there are
> active calls.
> >
> > How to load the configurations when there are active calls in the system.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From janvb at live.com  Thu Apr  8 09:38:11 2010
From: janvb at live.com (Jan Berger)
Date: Thu, 8 Apr 2010 18:38:11 +0200
Subject: [Freeswitch-users] Reload Configurations
In-Reply-To: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
Message-ID: <SNT135-w242877E7D97206DB3A5B28D2160@phx.gbl>


reload -f mod_sofia 

 

that forces the reload I detected

 

Jan
 


Date: Thu, 8 Apr 2010 11:59:34 -0400
From: lloyd.aloysius at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Reload Configurations

Hi All, 


I add some SIP trunks to FreeSWITCH.


reloadxml


reload mod_sofia  - Not allow to load the configuration when there are active calls. 


How to load the configurations when there are active calls in the system.


Thanks
Lloyd 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Thu Apr  8 09:43:11 2010
From: janvb at live.com (Jan Berger)
Date: Thu, 8 Apr 2010 18:43:11 +0200
Subject: [Freeswitch-users] mod-sofia stops
In-Reply-To: <w2h8a19bf2e1004080924je77f989pec8b88edf9588b96@mail.gmail.com>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>,
	<D3C25AC7-06D9-443C-8E1E-31BECC4515E1@freeswitch.org>,
	<w2h8a19bf2e1004080924je77f989pec8b88edf9588b96@mail.gmail.com>
Message-ID: <SNT135-w87ECB8E34B0CCE421A836D2160@phx.gbl>


hi,

 

One of the issues I have is that mod-sofia stops up if the network hicks up. The profiles drop out...

 

Is there any setting i can use to force sofia to come back into seervice automatically???

 

Jan
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brian at freeswitch.org  Thu Apr  8 09:46:50 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Apr 2010 11:46:50 -0500
Subject: [Freeswitch-users] Reload Configurations
In-Reply-To: <SNT135-w242877E7D97206DB3A5B28D2160@phx.gbl>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
	<SNT135-w242877E7D97206DB3A5B28D2160@phx.gbl>
Message-ID: <CF98CA81-1BB6-42FD-A67C-594E0C18F57A@freeswitch.org>

OH sure force it... how about you play with fire and dynamite at the same time.  NOT RECOMMENDED.

that -f flag can cause really evil shit to happen if you don't watch it.  Rescan will do what you want.

/b

On Apr 8, 2010, at 11:38 AM, Jan Berger wrote:

> reload -f mod_sofia 
>  
> that forces the reload I detected
>  
> Jan
>  

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From anthony.minessale at gmail.com  Thu Apr  8 09:51:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Apr 2010 11:51:57 -0500
Subject: [Freeswitch-users] mod-sofia stops
In-Reply-To: <SNT135-w87ECB8E34B0CCE421A836D2160@phx.gbl>
References: <x2s8a19bf2e1004080859g2c5a1991n96473e65d88224cf@mail.gmail.com>
	<D3C25AC7-06D9-443C-8E1E-31BECC4515E1@freeswitch.org>
	<w2h8a19bf2e1004080924je77f989pec8b88edf9588b96@mail.gmail.com>
	<SNT135-w87ECB8E34B0CCE421A836D2160@phx.gbl>
Message-ID: <i2o191c3a031004080951k2a9db24ahb8758cd3d70c5101@mail.gmail.com>

sometimes it gets false positives and switches to a bad ip
you can disable it in the autoload_configs/sofia.conf.xml

<param name="auto-restart" value="false"/>



On Thu, Apr 8, 2010 at 11:43 AM, Jan Berger <janvb at live.com> wrote:

>  hi,
>
> One of the issues I have is that mod-sofia stops up if the network hicks
> up. The profiles drop out...
>
> Is there any setting i can use to force sofia to come back into seervice
> automatically???
>
> Jan
>
> ------------------------------
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From devel at thom.fr.eu.org  Thu Apr  8 10:03:57 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Thu, 8 Apr 2010 19:03:57 +0200
Subject: [Freeswitch-users] Problem sending DTMF using an FXS channel
Message-ID: <000c01cad73d$7f324be0$7d96e3a0$@fr.eu.org>

Hello,

 

I?m having trouble with calls to remote IVR using DTMF, when the A leg is an
FXS port.

 

What happens is when the key is pressed on the phone, the DTMF is sent
inband to the callee party as voice, but also detected by freeswitch and so
resent by freeswitch to the callee party. This results in unusability of
called IVR.

 

Is there any setting that could be used to prevent freeswitch from detecting
DTMF and/or prevent freeswitch from resending the DTMF.

 

Thanks

 

Fran?ois

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From msc at freeswitch.org  Thu Apr  8 10:44:20 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 10:44:20 -0700
Subject: [Freeswitch-users] Help to call a VXML SIP number
In-Reply-To: <u2y1452e2981004072228haf2a8658wbd99fc7714f2a5b2@mail.gmail.com>
References: <o2k1452e2981004060244lcd21bf79j5d3a9ebcd76eaa9d@mail.gmail.com>
	<539EE8C5-A37F-432B-B1D0-6DA048E36EE0@avgs.ca>
	<y2g1452e2981004062333vc6da46b0s4a7249142cfe09b@mail.gmail.com>
	<m2y87f2f3b91004071325ma133f4d4x2ad9c9ce1c18ceae@mail.gmail.com>
	<u2y1452e2981004072228haf2a8658wbd99fc7714f2a5b2@mail.gmail.com>
Message-ID: <s2o87f2f3b91004081044j74b920apb122788cf80383c4@mail.gmail.com>

On Wed, Apr 7, 2010 at 10:28 PM, velusamy Krishnan <velu.technical at gmail.com
> wrote:

> Hi,
>      I have found that  NAT has been enabled in my Firewall. But I don't
> know how to solve this problem. Could you help me please..

What is the make and model of your firewall?
-MC
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From zhakrin at gmail.com  Thu Apr  8 04:04:57 2010
From: zhakrin at gmail.com (Chang Zhao)
Date: Thu, 8 Apr 2010 19:04:57 +0800
Subject: [Freeswitch-users] Unable to call out on FXO interface -- can
	receive calls
Message-ID: <001001cad70b$56d74100$0485c300$@com>

Hi,

I have a TDM400 card with 4 fxo interfaces on it, I can receive calls just
fine, but when trying to make an outbound call, the following shows up:

2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on
1:1
2010-04-08 17:23:46.843723 [DEBUG] mod_openzap.c:1530 got FXO sig 1:1
[TONE_DETECTED]
2010-04-08 17:23:46.843723 [WARNING] mod_openzap.c:1577 Unhandled msg type 9
for channel 1:1
2010-04-08 17:23:46.843723 [ERR] ozmod_analog.c:671 No Digits to send!
2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:672 Changing state on 1:1
from DIALING to BUSY

So it appears that it's detecting a dialtone, but then what's msg type 9?
Does anyone have any insights?  Using latest GIT head version and CentOS 5.4
64bit.


Thanks and regards,

CZ



From c.hiller at baigtel.com  Thu Apr  8 03:50:55 2010
From: c.hiller at baigtel.com (Christian Hiller - Baig Tel LTD)
Date: Thu, 08 Apr 2010 12:50:55 +0200
Subject: [Freeswitch-users] transcoding
Message-ID: <4BBDB50F.9070600@baigtel.com>

Hello,

i am using a TC400B transcoding PCI card and the module mod_dahdi_codec
seems to be loaded correctly, even i am getting

freeswitch at internal> dahdi_transcode
Using 0 encoders of a total of 92 available.
Using 0 decoders of a total of 92 available.

twice the message that i have encoders available.
Now how can i use them? I want G723 calls transcoded to PCMA.

I have set these variables. But when it comes to a call, it negoiates 
the codecs and fails.

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729"/>

Any help apreciated.
Its my first contact with freeswitch and i find it to be a pretty 
straightforward software. good job. thx

chris


From xianchun.bi at quanshi.com  Thu Apr  8 07:02:14 2010
From: xianchun.bi at quanshi.com (Bi,Xianchun)
Date: Thu, 8 Apr 2010 22:02:14 +0800
Subject: [Freeswitch-users] FS cannot produce any log file
Message-ID: <00a201cad724$18332540$48996fc0$@bi@quanshi.com>

Hi,

 

I have installed FreeSWITCH in my test server, the OS is Ubuntu 8.04 Server

 

In module configuration file conf/autoload_configs/modules.conf.xml,
log_file module is set active.

 

<load module="mod_logfile"/>

 

And in the log_file configuration file
conf/autoload_configs/logfile.conf.xml, the settings are as following:

 

<configuration name="logfile.conf" description="File Logging">

  <settings>

   <!-- true to auto rotate on HUP, false to open/close -->

  <param name="rotate-on-hup" value="true"/>

  </settings>

  <profiles>

    <profile name="default">

      <settings>

        <!-- File to log to -->

        <param name="logfile" value="/var/log/freeswitch.log"/>

        <!-- At this length in bytes rotate the log file (0 for never) -->

        <param name="rollover" value="10485760"/>

    <!-- Uncomment to prefix all log lines by the session's uuid  -->

    <!-- <param name="uuid" value="true" /> -->

      </settings>

      <mappings>

  <!--

       name can be a file name, function name or 'all'

       value is one or more of debug,info,notice,warning,err,crit,alert,all

       Please see comments in console.conf.xml for more information

  -->

  <map name="all" value="debug,info,notice,warning,err,crit,alert"/>

      </mappings>

    </profile>

  </profiles>

</configuration>

 

 

But the logfile never appers. I also try using touch to create a dummy file,
but the file never grow.

Restarting FreeSWITCH has no effect.

What wrong with my settings?

Anyone could give any suggestion?

Thanks in advance.

 

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From brian at freeswitch.org  Thu Apr  8 10:58:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Apr 2010 12:58:16 -0500
Subject: [Freeswitch-users] transcoding
In-Reply-To: <4BBDB50F.9070600@baigtel.com>
References: <4BBDB50F.9070600@baigtel.com>
Message-ID: <4FF7CCE3-ABFC-4BF2-A999-7276BD814547@freeswitch.org>

make sure mod_g723_1 isn't loaded.

/b

On Apr 8, 2010, at 5:50 AM, Christian Hiller - Baig Tel LTD wrote:

> Hello,
> 
> i am using a TC400B transcoding PCI card and the module mod_dahdi_codec
> seems to be loaded correctly, even i am getting
> 
> freeswitch at internal> dahdi_transcode
> Using 0 encoders of a total of 92 available.
> Using 0 decoders of a total of 92 available.
> 
> twice the message that i have encoders available.
> Now how can i use them? I want G723 calls transcoded to PCMA.
> 
> I have set these variables. But when it comes to a call, it negoiates 
> the codecs and fails.
> 
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729"/>
> 
> Any help apreciated.
> Its my first contact with freeswitch and i find it to be a pretty 
> straightforward software. good job. thx
> 
> chris



From msc at freeswitch.org  Thu Apr  8 10:59:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 10:59:10 -0700
Subject: [Freeswitch-users] Problem sending DTMF using an FXS channel
In-Reply-To: <000c01cad73d$7f324be0$7d96e3a0$@fr.eu.org>
References: <000c01cad73d$7f324be0$7d96e3a0$@fr.eu.org>
Message-ID: <y2s87f2f3b91004081059mceec7650r50ef6b8f1e4d464@mail.gmail.com>

Pastebin your dialplan config that handles this call as well as a debug
trace from the console.
-MC

On Thu, Apr 8, 2010 at 10:03 AM, <devel at thom.fr.eu.org> wrote:

>  Hello,
>
>
>
> I?m having trouble with calls to remote IVR using DTMF, when the A leg is
> an FXS port.
>
>
>
> What happens is when the key is pressed on the phone, the DTMF is sent
> inband to the callee party as voice, but also detected by freeswitch and so
> resent by freeswitch to the callee party. This results in unusability of
> called IVR.
>
>
>
> Is there any setting that could be used to prevent freeswitch from
> detecting DTMF and/or prevent freeswitch from resending the DTMF.
>
>
>
> Thanks
>
>
>
> Fran?ois
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 11:01:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:01:12 -0700
Subject: [Freeswitch-users] Unable to call out on FXO interface -- can
	receive calls
In-Reply-To: <001001cad70b$56d74100$0485c300$@com>
References: <001001cad70b$56d74100$0485c300$@com>
Message-ID: <v2q87f2f3b91004081101ubed164d1k888d243bb6fe03bf@mail.gmail.com>

Pastebin the dialplan config that handles this call as well as the whole
debug trace from start to finish of the call. The clues will be there.
Unfortunately these few debug lines don't paint the whole picture.
-MC

On Thu, Apr 8, 2010 at 4:04 AM, Chang Zhao <zhakrin at gmail.com> wrote:

> Hi,
>
> I have a TDM400 card with 4 fxo interfaces on it, I can receive calls just
> fine, but when trying to make an outbound call, the following shows up:
>
> 2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on
> 1:1
> 2010-04-08 17:23:46.843723 [DEBUG] mod_openzap.c:1530 got FXO sig 1:1
> [TONE_DETECTED]
> 2010-04-08 17:23:46.843723 [WARNING] mod_openzap.c:1577 Unhandled msg type
> 9
> for channel 1:1
> 2010-04-08 17:23:46.843723 [ERR] ozmod_analog.c:671 No Digits to send!
> 2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:672 Changing state on 1:1
> from DIALING to BUSY
>
> So it appears that it's detecting a dialtone, but then what's msg type 9?
> Does anyone have any insights?  Using latest GIT head version and CentOS
> 5.4
> 64bit.
>
>
> Thanks and regards,
>
> CZ
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 11:02:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:02:44 -0700
Subject: [Freeswitch-users] FS cannot produce any log file
In-Reply-To: <8315592515048549950@unknownmsgid>
References: <8315592515048549950@unknownmsgid>
Message-ID: <j2k87f2f3b91004081102y49087905jc4d0e7dec88ae9a0@mail.gmail.com>

Make absolutely certain that there isn't a permissions issue with FreeSWITCH
writing to the /var/log/ directory.
-MC

On Thu, Apr 8, 2010 at 7:02 AM, Bi,Xianchun <xianchun.bi at quanshi.com> wrote:

>  Hi,
>
>
>
> I have installed FreeSWITCH in my test server, the OS is Ubuntu 8.04 Server
>
>
>
> In module configuration file conf/autoload_configs/modules.conf.xml,
> log_file module is set active.
>
>
>
> <load module="mod_logfile"/>
>
>
>
> And in the log_file configuration file
> conf/autoload_configs/logfile.conf.xml, the settings are as following:
>
>
>
> <configuration name="logfile.conf" description="File Logging">
>
>   <settings>
>
>    <!-- true to auto rotate on HUP, false to open/close -->
>
>   <param name="rotate-on-hup" value="true"/>
>
>   </settings>
>
>   <profiles>
>
>     <profile name="default">
>
>       <settings>
>
>         <!-- File to log to -->
>
>         <param name="logfile" value="/var/log/freeswitch.log"/>
>
>         <!-- At this length in bytes rotate the log file (0 for never) -->
>
>         <param name="rollover" value="10485760"/>
>
>     <!-- Uncomment to prefix all log lines by the session's uuid  -->
>
>     <!-- <param name="uuid" value="true" /> -->
>
>       </settings>
>
>       <mappings>
>
>   <!--
>
>        name can be a file name, function name or 'all'
>
>        value is one or more of debug,info,notice,warning,err,crit,alert,all
>
>        Please see comments in console.conf.xml for more information
>
>   -->
>
>   <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
>       </mappings>
>
>     </profile>
>
>   </profiles>
>
> </configuration>
>
>
>
>
>
> But the logfile never appers. I also try using touch to create a dummy
> file, but the file never grow.
>
> Restarting FreeSWITCH has no effect.
>
> What wrong with my settings?
>
> Anyone could give any suggestion?
>
> Thanks in advance.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 11:06:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:06:26 -0700
Subject: [Freeswitch-users] Another IRC day in #freeswitch!
Message-ID: <g2s87f2f3b91004081106q88161761w198af77b76ace035@mail.gmail.com>

Hey everyone, come join us in #freeswitch on irc.freenode.net! We are going
for 250 users. (No sockpuppets, please.) If you don't have an IRC client
then visit http://www.freeswitch.org and click the Join Us On IRC link to
use the Web IRC applet.

See you in channel!
-MC (IRC: mercutioviz)
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From msc at freeswitch.org  Thu Apr  8 11:19:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:19:37 -0700
Subject: [Freeswitch-users] G.729 Info On Wiki
Message-ID: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>

Hi folks,

Just a quick FYI, there is g.729 license info now on the wiki:
http://wiki.freeswitch.org/wiki/Mod_com_g729

Thanks to Steve Ayre for starting this page. If you have g.729 knowledge to
add to the page please do so. Also, the INSTALL.txt file has been updated
with a bit more detail:
http://files.freeswitch.org/g729/INSTALL.txt

Happy transcoding!

-Michael
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From aep.lists at it46.se  Thu Apr  8 12:04:18 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Thu, 8 Apr 2010 21:04:18 +0200
Subject: [Freeswitch-users] Parsing XML files from Spidermonkey
Message-ID: <48fade7f8a99c447d71334d2d3e589bd.squirrel@correo.nodo50.org>

Hi,

After one year using FS i am starting to like XML so i am trying to get
a Javascript script to read local XML files.

I am using the XML method and getting Syntax errors from spidermonkey

While something like this works:
xmldata = new XML("<tag>foo</tag>");

I have not been able to read and parse XML local files, using File or FileIO
methods

A simple example like this returns Syntax error.
var foo = apiExecute ("show", "channels as xml");
xmldata = new XML(foo);

Has anyone managed to use the XML method from spidermonkey to read a XML
stored file?

There are some E4X bugs around and i wonder if those are the cause of the
"Syntax Error" feedback even reading a very basic XML file
-- 
Stopping junk mailers is good for the environment





From moises.silva at gmail.com  Thu Apr  8 12:47:27 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 8 Apr 2010 15:47:27 -0400
Subject: [Freeswitch-users] transcoding
In-Reply-To: <4FF7CCE3-ABFC-4BF2-A999-7276BD814547@freeswitch.org>
References: <4BBDB50F.9070600@baigtel.com>
	<4FF7CCE3-ABFC-4BF2-A999-7276BD814547@freeswitch.org>
Message-ID: <x2xc4d05cbe1004081247qc8644eeax1f2afa4667c958e9@mail.gmail.com>

and I think mod_g729 (pass-thru codec) should not be loaded either.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, Apr 8, 2010 at 1:58 PM, Brian West <brian at freeswitch.org> wrote:

> make sure mod_g723_1 isn't loaded.
>
> /b
>
> On Apr 8, 2010, at 5:50 AM, Christian Hiller - Baig Tel LTD wrote:
>
> > Hello,
> >
> > i am using a TC400B transcoding PCI card and the module mod_dahdi_codec
> > seems to be loaded correctly, even i am getting
> >
> > freeswitch at internal> dahdi_transcode
> > Using 0 encoders of a total of 92 available.
> > Using 0 decoders of a total of 92 available.
> >
> > twice the message that i have encoders available.
> > Now how can i use them? I want G723 calls transcoded to PCMA.
> >
> > I have set these variables. But when it comes to a call, it negoiates
> > the codecs and fails.
> >
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729"/>
> >
> > Any help apreciated.
> > Its my first contact with freeswitch and i find it to be a pretty
> > straightforward software. good job. thx
> >
> > 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 kenfulmer at icstechnologysolutions.com  Thu Apr  8 12:54:19 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 8 Apr 2010 14:54:19 -0500
Subject: [Freeswitch-users] Two Major Problems
Message-ID: <00da01cad755$474b9c50$d5e2d4f0$@com>

I'm using the external sip profile to route calls from a voice gateway
router to an internal PBX.

 

In separate xml files in the sofia/external directory, I have the following:

 

<include>

    <gateway name="Server-1">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value=" X.X.X.X "/>

      <param name="proxy" value=" X.X.X.X "/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="Server-2">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="Voice Gateway">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

In the dial plan/public directory, I have the following (different xml
files):

 

<extension name="PBX">

   <condition field="destination_number" expression="^(Directory Numbers)$">

       <action application="bridge" data="sofia/gateway/Server-1/$1"/>

       <action application="bridge" data="sofia/gateway/Server-2/$1"/>

   </condition>

</extension>

 

<extension name="PSTN">

   <condition field="destination_number" expression="^(\d{10})$">

       <action application="bridge" data="sofia/gateway/Voice Gateway/$1"/>

   </condition>

</extension>

 

In the sip profiles/eternal.xml file, I have the following major settings:

 

  <settings>

    <param name="inbound-proxy-media" value="false"/>

    <param name="disable-transcoding" value="false"/>

    <param name="inbound-late-negotiation" value="true"/>

    <param name="user-agent-string" value="My Company Name"/>

    <param name="disable-register" value="true"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="inbound-codec-prefs" value="G729,PCMU"

    <param name="outbound-codec-prefs" value="G729,PCMU"/>

    <param name="media-option" value="resume-media-on-hold"/>

    <param name="stun-enabled" value="false"/>

    <param name="debug" value="0"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="rtp-ip" value="X.X.X.X"/>

    <param name="sip-ip" value="X.X.X.X"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="2000"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

  </settings>

</profile>

 

Currently, I'm experiencing two issues.

 

1.       Inbound calls work as long as I'm using "inbound-proxy-media". When
I disable that setting and force transcoding, the phone rings but
immediately hangs up when I pick up the receiver. I see an "Incompatible
Destination" message on the console.

2.       Outbound calls don't work at all. On the console, I see "407 Proxy
Authentication Required". If I'm using the external profile, how could it
require authentication? 

 

Any ideas? 

 

Thanks,

 

Ken Fulmer

 

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From devel at thom.fr.eu.org  Thu Apr  8 12:55:35 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Thu, 8 Apr 2010 21:55:35 +0200
Subject: [Freeswitch-users] Problem sending DTMF using an FXS channel
Message-ID: <004d01cad755$75426d00$5fc74700$@fr.eu.org>

Log is in http://pastebin.freeswitch.org/12665

Dialplan is following (included in default freeswitch) :

 

    <extension name="Outgoing FXO Free">

      <condition field="destination_number" expression="^(0[1234589].*)$">

                  <!--<action application="ring_ready"/>-->

          <action application="set" data="effective_caller_id_name=Francois
Legal"/>

          <action application="set"
data="effective_caller_id_number=0950640430"/>

          <action application="set" data="ringback=${hold_music}"/>

          <action application="set" data="transfer_ringback=${hold_music}"/>

          <action application="bridge"
data="openzap/2/2/$1|sofia/gateway/freephonie.net/$1|openzap/2/1/$1"/>

      </condition>

    </extension>

 

I forgot to mention that I?m using sangoma A400 with HW DTMF detection.

 

Fran?ois

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael
Collins
Envoy? : jeudi 8 avril 2010 19:59
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] Problem sending DTMF using an FXS channel

 

Pastebin your dialplan config that handles this call as well as a debug
trace from the console.
-MC

On Thu, Apr 8, 2010 at 10:03 AM, <devel at thom.fr.eu.org> wrote:

Hello,

 

I?m having trouble with calls to remote IVR using DTMF, when the A leg is an
FXS port.

 

What happens is when the key is pressed on the phone, the DTMF is sent
inband to the callee party as voice, but also detected by freeswitch and so
resent by freeswitch to the callee party. This results in unusability of
called IVR.

 

Is there any setting that could be used to prevent freeswitch from detecting
DTMF and/or prevent freeswitch from resending the DTMF.

 

Thanks

 

Fran?ois


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Thu Apr  8 12:59:36 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Apr 2010 14:59:36 -0500
Subject: [Freeswitch-users] transcoding
In-Reply-To: <x2xc4d05cbe1004081247qc8644eeax1f2afa4667c958e9@mail.gmail.com>
References: <4BBDB50F.9070600@baigtel.com>
	<4FF7CCE3-ABFC-4BF2-A999-7276BD814547@freeswitch.org>
	<x2xc4d05cbe1004081247qc8644eeax1f2afa4667c958e9@mail.gmail.com>
Message-ID: <528D6591-3E12-469E-8DFA-9AF01F368630@freeswitch.org>

You know we should deny load if something is already registered maybe?

Cuz I can see this being a problem.

/b

On Apr 8, 2010, at 2:47 PM, Moises Silva wrote:

> and I think mod_g729 (pass-thru codec) should not be loaded either.
> 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> 
> 
> On Thu, Apr 8, 2010 at 1:58 PM, Brian West <brian at freeswitch.org> wrote:
> make sure mod_g723_1 isn't loaded.
> 
> /b

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From lloyd.aloysius at gmail.com  Thu Apr  8 14:00:35 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 8 Apr 2010 17:00:35 -0400
Subject: [Freeswitch-users] NAT Problems
Message-ID: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>

Hi All,

I am having lots of problem with NAT and FreeSWITCH.

FreeeSWITCH running Public IP.

Phones Connecting to FreeeSWITCH  through NAT. Cable Provider Router/Modem :
SMC Gateway.

The Phone Losing the Registrations. The Environment working for Asterisk
without any problem.

Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
FreeSWITCH work.

How to solve this problem.

Thanks,
Lloyd
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From msc at freeswitch.org  Thu Apr  8 14:07:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 14:07:26 -0700
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
Message-ID: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>

Greetings all,

The FreeSWITCH team would like to let everyone know that we do indeed sell
g.729 licenses for $10 each. Use this link to initiate a purchase:
http://www.freeswitch.org/node/235

Note: licenses are available only for Linux-based systems at this time.
Please stay tuned for updates.

The INSTALL.txt file has very detailed instructions:
http://files.freeswitch.org/g729/INSTALL.txt

Keep in mind that a single license includes one encoder and one decoder,
that is, it can transcode both directions of a single phone call. If you
have any other questions please email us here or join us in #freeswitch on
irc.freenode.net.
-Michael
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From krzysztofdrewicz at gmail.com  Thu Apr  8 13:29:28 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Thu, 8 Apr 2010 22:29:28 +0200
Subject: [Freeswitch-users] move on from asterisk,
	the Random or rand function
Message-ID: <u2l9372a8331004081329yba87f64n3b09e96663fe575c@mail.gmail.com>

Hi,

i'm slowly, but constantly moving from asterisk to fs.

Stuck on translating this into a freeswitch:

exten => s,1,Random(90:s,3)
exten => s,2,Noop(his is hit 10% times)
exten => s,3,Dial(SIP/8000)

What i need to do is put a bucket on some calling_number + destination numer
with some random percent to a voiceguide and hangup.

i've found that memcache is very convinent way to put data into
dialplan logic in runtime, so i plan to do:

<extension name="bucket_part_01" continue="true">
        <condition field="${do_bucket}" expression=".+" break="on-true"/>
        <condition field="caller_id_number" expression="^123456789$">
        <action data="do_bucket=${memcache(get
b_${caller_id_number})}" application="set"/>
        <action data="bucket_part_02" application="transfer"/>
        <action application="hangup"/>
        </condition>
</extension>

So above i check for caller_id_number in the memchache b_123456789 with option 1
Then i transfer to anoter extension:

?<extension name="bucket_part_02">
        <condition field="destination_number" expression="bucket_part_02"/>
        <condition field="${czy_robic}" expression="^1$">
                <action data="bypass_media=true" application="set"/>
                <action data="hits=${memcache(increment
hit_bucket_test)}" application="set"/>

AND HERE I do need some logic that does if Random() > 50%

                <action application="answer"/>
                <action
data="/usr/local/freeswitch/sounds/pc/weRbusyRightNow.wav"
application="playback"/>
                <action application="hangup"/>
        </condition>
</extension>


any cleaner approach or someone has done it better?
note: don't want to use any lua, api or sth, just plain xml dialplan logic.


From jeremy at seadragons.us  Thu Apr  8 12:10:16 2010
From: jeremy at seadragons.us (Jeremy Shaffner)
Date: Thu, 8 Apr 2010 15:10:16 -0400
Subject: [Freeswitch-users] mod_conference with H.323 Video
Message-ID: <3CA91892-B327-4575-BF7A-9B5795BF2532@seadragons.us>

Hello,

Is this known to work at all?

-Jeremy



From brian at freeswitch.org  Thu Apr  8 14:11:44 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Apr 2010 16:11:44 -0500
Subject: [Freeswitch-users] mod_conference with H.323 Video
In-Reply-To: <3CA91892-B327-4575-BF7A-9B5795BF2532@seadragons.us>
References: <3CA91892-B327-4575-BF7A-9B5795BF2532@seadragons.us>
Message-ID: <4840E9FA-C9CE-4DBE-BE1A-53539B15F560@freeswitch.org>

Nope.

/b

On Apr 8, 2010, at 2:10 PM, Jeremy Shaffner wrote:

> Hello,
> 
> Is this known to work at all?
> 
> -Jeremy



From msc at freeswitch.org  Thu Apr  8 14:25:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 14:25:11 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <00da01cad755$474b9c50$d5e2d4f0$@com>
References: <AcrXVUbBfxNrQGSATK6Slv2XsFvqkg==>
	<00da01cad755$474b9c50$d5e2d4f0$@com>
Message-ID: <y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>

Did you purchase g.729 licenses for this system? If not then you won't be
able to transcode g.729 and the incompatible destination error will keep
popping up if you're not in proxy media mode.

Where is the 407 coming from? If it's coming from your provider then they
want you to auth calls, which means you need to make sure your gateway has
the correct username and password information.

-MC

On Thu, Apr 8, 2010 at 12:54 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  I?m using the external sip profile to route calls from a voice gateway
> router to an internal PBX.
>
>
>
> In separate xml files in the sofia/external directory, I have the
> following:
>
>
>
> <include>
>
>     <gateway name="Server-1">
>
>       <param name="username" value="gateway"/>
>
>       <param name="password" value="gateway"/>
>
>       <param name="realm" value=" X.X.X.X "/>
>
>       <param name="proxy" value=" X.X.X.X "/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> <include>
>
>     <gateway name="Server-2">
>
>       <param name="username" value="gateway"/>
>
>       <param name="password" value="gateway"/>
>
>       <param name="realm" value="X.X.X.X"/>
>
>       <param name="proxy" value="X.X.X.X"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> <include>
>
>     <gateway name="Voice Gateway">
>
>       <param name="username" value="gateway"/>
>
>       <param name="password" value="gateway"/>
>
>       <param name="realm" value="X.X.X.X"/>
>
>       <param name="proxy" value="X.X.X.X"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> In the dial plan/public directory, I have the following (different xml
> files):
>
>
>
> <extension name="PBX">
>
>    <condition field="destination_number" expression="^(Directory
> Numbers)$">
>
>        <action application="bridge" data="sofia/gateway/Server-1/$1"/>
>
>        <action application="bridge" data="sofia/gateway/Server-2/$1"/>
>
>    </condition>
>
> </extension>
>
>
>
> <extension name="PSTN">
>
>    <condition field="destination_number" expression="^(\d{10})$">
>
>        <action application="bridge" data="sofia/gateway/Voice Gateway/$1"/>
>
>    </condition>
>
> </extension>
>
>
>
> In the sip profiles/eternal.xml file, I have the following major settings:
>
>
>
>   <settings>
>
>     <param name="inbound-proxy-media" value="false"/>
>
>     <param name="disable-transcoding" value="false"/>
>
>     <param name="inbound-late-negotiation" value="true"/>
>
>     <param name="user-agent-string" value="My Company Name"/>
>
>     <param name="disable-register" value="true"/>
>
>     <param name="inbound-codec-negotiation" value="generous"/>
>
>     <param name="inbound-codec-prefs" value="G729,PCMU"
>
>     <param name="outbound-codec-prefs" value="G729,PCMU"/>
>
>     <param name="media-option" value="resume-media-on-hold"/>
>
>     <param name="stun-enabled" value="false"/>
>
>     <param name="debug" value="0"/>
>
>     <param name="sip-trace" value="no"/>
>
>     <param name="rfc2833-pt" value="101"/>
>
>     <param name="sip-port" value="5080"/>
>
>     <param name="rtp-ip" value="X.X.X.X"/>
>
>     <param name="sip-ip" value="X.X.X.X"/>
>
>     <param name="dialplan" value="XML"/>
>
>     <param name="context" value="public"/>
>
>     <param name="dtmf-duration" value="2000"/>
>
>     <param name="rtp-timer-name" value="soft"/>
>
>     <param name="rtp-timeout-sec" value="300"/>
>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>
>   </settings>
>
> </profile>
>
>
>
> Currently, I?m experiencing two issues.
>
>
>
> 1.       Inbound calls work as long as I?m using ?inbound-proxy-media?.
> When I disable that setting and force transcoding, the phone rings but
> immediately hangs up when I pick up the receiver. I see an ?Incompatible
> Destination? message on the console.
>
> 2.       Outbound calls don?t work at all. On the console, I see ?407
> Proxy Authentication Required?. If I?m using the external profile, how could
> it require authentication?
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 14:27:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 14:27:41 -0700
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
Message-ID: <w2t87f2f3b91004081427l545cf84dq485b143156af5eab@mail.gmail.com>

Did you disable the SIP ALG in the SMC Gateway?
-MC

On Thu, Apr 8, 2010 at 2:00 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I am having lots of problem with NAT and FreeSWITCH.
>
> FreeeSWITCH running Public IP.
>
> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider Router/Modem
> : SMC Gateway.
>
> The Phone Losing the Registrations. The Environment working for Asterisk
> without any problem.
>
> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
> FreeSWITCH work.
>
> How to solve this problem.
>
> Thanks,
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at miriamtech.com  Thu Apr  8 14:42:24 2010
From: ken at miriamtech.com (Ken Treis)
Date: Thu, 8 Apr 2010 14:42:24 -0700
Subject: [Freeswitch-users] Setting channel variables for FIFO-originated
	calls
Message-ID: <A2710AD0-1C7A-40A0-BDB5-1FA5F99B9CA1@miriamtech.com>

I'm still pretty new to FreeSWITCH, but learning my way around.

I've been trying to figure out how to use a different filename for each call in an on-hook agent fifo setup. I've been setting fifo_record_template in the originate string of the member definition, e.g.

>  {member_wait=nowait,fifo_record_template=$${base_dir}/recordings/fifo.wav}sofia/gateway/sip.jnctn.net/18005551212

This works, but as soon as I put a (non-preprocessor) variable in there, I get a complaint that my variable contains a variable: "Invalid data (${fifo_record_template} contains a variable)". But the code in mod_fifo.c seems to expect to find variables in there:

>                 if (record_template) {
>                     expanded = switch_channel_expand_variables(other_channel, record_template);
>                     switch_ivr_record_session(session, expanded, 0, NULL);
>                 }

So my only conclusions at this point are that either:

1. The validation rule that gives me the "contains a variable" is being overly paranoid, or
2. There must be some other way to set this variable.

I've chased #2 for a little bit, but apparently since FIFO-originated calls are created without any linked session (first arg to switch_ivr_originate is NULL), I can't export the variable from the original channel either. Is there some other way to set channel variables on the calls the FIFO originates?

--
Ken Treis
Miriam Technologies, Inc.



From kenfulmer at icstechnologysolutions.com  Thu Apr  8 15:18:40 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 8 Apr 2010 17:18:40 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
References: <AcrXVUbBfxNrQGSATK6Slv2XsFvqkg==>	<00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
Message-ID: <011701cad769$714e8670$53eb9350$@com>

Actually, I did purchase a license and installed it today. One call
establishes at 729. When I hang up the phone and try again, it's 711. 

 

The Proxy Authentication Required is being sent by FreeSwitch to the
internal PBX. I have registration disabled on the FreeSwitch gateway and the
internal server. 

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 08, 2010 4:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

Did you purchase g.729 licenses for this system? If not then you won't be
able to transcode g.729 and the incompatible destination error will keep
popping up if you're not in proxy media mode.

Where is the 407 coming from? If it's coming from your provider then they
want you to auth calls, which means you need to make sure your gateway has
the correct username and password information.

-MC

On Thu, Apr 8, 2010 at 12:54 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

I'm using the external sip profile to route calls from a voice gateway
router to an internal PBX.

 

In separate xml files in the sofia/external directory, I have the following:

 

<include>

    <gateway name="Server-1">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value=" X.X.X.X "/>

      <param name="proxy" value=" X.X.X.X "/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="Server-2">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="Voice Gateway">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

In the dial plan/public directory, I have the following (different xml
files):

 

<extension name="PBX">

   <condition field="destination_number" expression="^(Directory Numbers)$">

       <action application="bridge" data="sofia/gateway/Server-1/$1"/>

       <action application="bridge" data="sofia/gateway/Server-2/$1"/>

   </condition>

</extension>

 

<extension name="PSTN">

   <condition field="destination_number" expression="^(\d{10})$">

       <action application="bridge" data="sofia/gateway/Voice Gateway/$1"/>

   </condition>

</extension>

 

In the sip profiles/eternal.xml file, I have the following major settings:

 

  <settings>

    <param name="inbound-proxy-media" value="false"/>

    <param name="disable-transcoding" value="false"/>

    <param name="inbound-late-negotiation" value="true"/>

    <param name="user-agent-string" value="My Company Name"/>

    <param name="disable-register" value="true"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="inbound-codec-prefs" value="G729,PCMU"

    <param name="outbound-codec-prefs" value="G729,PCMU"/>

    <param name="media-option" value="resume-media-on-hold"/>

    <param name="stun-enabled" value="false"/>

    <param name="debug" value="0"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="rtp-ip" value="X.X.X.X"/>

    <param name="sip-ip" value="X.X.X.X"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="2000"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

  </settings>

</profile>

 

Currently, I'm experiencing two issues.

 

1.       Inbound calls work as long as I'm using "inbound-proxy-media". When
I disable that setting and force transcoding, the phone rings but
immediately hangs up when I pick up the receiver. I see an "Incompatible
Destination" message on the console.

2.       Outbound calls don't work at all. On the console, I see "407 Proxy
Authentication Required". If I'm using the external profile, how could it
require authentication? 

 

Any ideas? 

 

Thanks,

 

Ken Fulmer

 


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

 

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From ken at miriamtech.com  Thu Apr  8 15:46:43 2010
From: ken at miriamtech.com (Ken Treis)
Date: Thu, 8 Apr 2010 15:46:43 -0700
Subject: [Freeswitch-users] Setting channel variables for
	FIFO-originated calls
In-Reply-To: <A2710AD0-1C7A-40A0-BDB5-1FA5F99B9CA1@miriamtech.com>
References: <A2710AD0-1C7A-40A0-BDB5-1FA5F99B9CA1@miriamtech.com>
Message-ID: <5F59D6E4-2E17-4786-8261-E0B2C9E152A1@miriamtech.com>

For what it's worth, I've worked around this by pulling the fifo_record_template variable from the originating channel instead by using the attached patch. It means you can't set the fifo_record_template in your member definition anymore, but now the following variable expansion works:

	<action application="set" data='fifo_record_template=/mnt/recordings/${strftime(%Y%m%d_%H%M%S)}_${uuid:-4}_${ani}.wav' />
	<action application="fifo" data="..."/>

In my case, I don't really care about any of the details of the outbound call, so this worked nicely.

--
Ken Treis
Miriam Technologies, Inc.
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From david.ponzone at gmail.com  Thu Apr  8 16:23:33 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 9 Apr 2010 01:23:33 +0200
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
Message-ID: <23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>

On most low-end routers, the NAT table will expire UDP translations  
after 60 sec.
Did you configure your phones to send a NAT keep-alive every X  
seconds, with X < 60 ?
You can also use sip-force-expires on the FS side.

In your Asterisk config, do you use qualify=yes ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :

> Hi All,
>
> I am having lots of problem with NAT and FreeSWITCH.
>
> FreeeSWITCH running Public IP.
>
> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider Router/ 
> Modem : SMC Gateway.
>
> The Phone Losing the Registrations. The Environment working for  
> Asterisk without any problem.
>
> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying  
> make FreeSWITCH work.
>
> How to solve this problem.
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 16:24:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 16:24:43 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <011701cad769$714e8670$53eb9350$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
Message-ID: <y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>

On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
Make sure that the encoder/decoder isn't still in use prior to trying the
second call. After you hang up, do a "show channels" and see if the call is
still "up" or not. Also, do "g729_status" to see if the encoder or decoder
is in use. Keep doing "g729_status" until the 'coders are not in use. If
there is a long delay then open up a JIRA ticket on jira.freeswitch.org.

>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
By default the SIP profile will challenge if the IP address of the caller is
not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
"domains" node. Add your PBX's IP address. You'll see an example in the
comments. Once you're done editing, save the file and then go to the fs_cli
and do:
reloadacl reloadxml

Then make a call from PBX to FS and it should go through.
-MC
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From msc at freeswitch.org  Thu Apr  8 16:29:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 16:29:23 -0700
Subject: [Freeswitch-users] move on from asterisk,
	the Random or rand 	function
In-Reply-To: <u2l9372a8331004081329yba87f64n3b09e96663fe575c@mail.gmail.com>
References: <u2l9372a8331004081329yba87f64n3b09e96663fe575c@mail.gmail.com>
Message-ID: <y2j87f2f3b91004081629ncbe6b11eo8d8886db17c1331f@mail.gmail.com>

Good news, bad news:

The good news is that the XML dialplan is a very efficient way of handling
call routing. The bad news is that the XML dialplan is not a programming
language. There is no "rand" function, so you'll need to do something
different. One option is to use mod_distributor, however it is not "random"
so that may not work for you. The other option is to call Lua or system or
something and snag a random number.

-MC

On Thu, Apr 8, 2010 at 1:29 PM, Krzysztof Drewicz <
krzysztofdrewicz at gmail.com> wrote:

> Hi,
>
> i'm slowly, but constantly moving from asterisk to fs.
>
> Stuck on translating this into a freeswitch:
>
> exten => s,1,Random(90:s,3)
> exten => s,2,Noop(his is hit 10% times)
> exten => s,3,Dial(SIP/8000)
>
> What i need to do is put a bucket on some calling_number + destination
> numer
> with some random percent to a voiceguide and hangup.
>
> i've found that memcache is very convinent way to put data into
> dialplan logic in runtime, so i plan to do:
>
> <extension name="bucket_part_01" continue="true">
>        <condition field="${do_bucket}" expression=".+" break="on-true"/>
>        <condition field="caller_id_number" expression="^123456789$">
>        <action data="do_bucket=${memcache(get
> b_${caller_id_number})}" application="set"/>
>        <action data="bucket_part_02" application="transfer"/>
>        <action application="hangup"/>
>        </condition>
> </extension>
>
> So above i check for caller_id_number in the memchache b_123456789 with
> option 1
> Then i transfer to anoter extension:
>
>  <extension name="bucket_part_02">
>        <condition field="destination_number" expression="bucket_part_02"/>
>        <condition field="${czy_robic}" expression="^1$">
>                <action data="bypass_media=true" application="set"/>
>                <action data="hits=${memcache(increment
> hit_bucket_test)}" application="set"/>
>
> AND HERE I do need some logic that does if Random() > 50%
>
>                <action application="answer"/>
>                <action
> data="/usr/local/freeswitch/sounds/pc/weRbusyRightNow.wav"
> application="playback"/>
>                <action application="hangup"/>
>        </condition>
> </extension>
>
>
> any cleaner approach or someone has done it better?
> note: don't want to use any lua, api or sth, just plain xml dialplan logic.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Thu Apr  8 18:29:45 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 8 Apr 2010 21:29:45 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
Message-ID: <k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>

Yes Asterisk config using the qualify=yes.

I will modify the Registration time out by 50sec and see how it is behaving.

Thanks
Lloyd




On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> On most low-end routers, the NAT table will expire UDP translations after
> 60 sec.
> Did you configure your phones to send a NAT keep-alive every X seconds,
> with X < 60 ?
> You can also use sip-force-expires on the FS side.
>
> In your Asterisk config, do you use qualify=yes ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>
> Hi All,
>
> I am having lots of problem with NAT and FreeSWITCH.
>
> FreeeSWITCH running Public IP.
>
> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider Router/Modem
> : SMC Gateway.
>
> The Phone Losing the Registrations. The Environment working for Asterisk
> without any problem.
>
> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
> FreeSWITCH work.
>
> How to solve this problem.
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Thu Apr  8 18:31:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 8 Apr 2010 21:31:00 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
Message-ID: <w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>

what is the lowest value I can use for sip-force-expires ?

Thanks
Lloyd


On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

>
> Yes Asterisk config using the qualify=yes.
>
> I will modify the Registration time out by 50sec and see how it is
> behaving.
>
> Thanks
> Lloyd
>
>
>
>
> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> On most low-end routers, the NAT table will expire UDP translations after
>> 60 sec.
>> Did you configure your phones to send a NAT keep-alive every X seconds,
>> with X < 60 ?
>> You can also use sip-force-expires on the FS side.
>>
>> In your Asterisk config, do you use qualify=yes ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>
>> Hi All,
>>
>> I am having lots of problem with NAT and FreeSWITCH.
>>
>> FreeeSWITCH running Public IP.
>>
>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider Router/Modem
>> : SMC Gateway.
>>
>> The Phone Losing the Registrations. The Environment working for Asterisk
>> without any problem.
>>
>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>> FreeSWITCH work.
>>
>> How to solve this problem.
>>
>> Thanks,
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mcampbellsmith at gmail.com  Thu Apr  8 20:48:17 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Apr 2010 13:48:17 +1000
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
Message-ID: <m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>

Thanks Milena... I upgraded git now - git version 1.7.0.4

I did a 'get pull && make install' and still the same problem.

Do I have to do a get clone or something?  Ideas?  Thanks

freeswitch:~# git --version
git version 1.7.0.4
freeswitch:~# fs_cli
            _____ ____     ____ _     ___
           |  ___/ ___|   / ___| |   |_ _|
           | |_  \___ \  | |   | |    | |
           |  _|  ___) | | |___| |___ | |
           |_|   |____/   \____|_____|___|

*******************************************************
* Anthony Minessale II, Ken Rice, Michael Jerris      *
* FreeSWITCH (http://www.freeswitch.org)              *
* Paypal Donations Appreciated: paypal at freeswitch.org *
* Brought to you by ClueCon http://www.cluecon.com/   *
*******************************************************

Type /help <enter> to see a list of commands


+OK log level  [7]
freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-)

On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
>
>
> Can anything be done in the freeswitch code so when "git pull" is executed,
> the "--pretty"?argument is?also set?where "--format" is set to make it
> compatible with both older and newer versions of git? or it is all up to
> what git does and nothing to do on fs?
>
>
>
> PS: Mark,?the issue you're facing is because of your version of git, the CLI
> shows the freeswitch version properly with git 1.7.0.4, the "format"
> argument isn't recognized by your version of git.
>
>
> 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
>>
>> Git was installed as described on the wiki. ?I am using Debian Lenny
>> and Git version 1.5.6.5
>>
>> I just did a git pull and had the same issue... FS still shows
>> FreeSWITCH Version 1.0.head (git-)
>>
>>
>>
>> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
>> > Upgrade your git, then it will show it correctly.
>> > http://wiki.freeswitch.org/wiki/Git_Install
>> > djbinter
>> > ________________________________
>> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
>> > To: freeswitch-users at lists.freeswitch.org
>> > Sent: Wed, April 7, 2010 11:54:27 PM
>> > Subject: [Freeswitch-users] version number: git checkout
>> >
>> > Hi!
>> >
>> > I just used git for the first time ever to checkout FreeSwitch as
>> > described on the wiki at
>> > http://wiki.freeswitch.org/wiki/Installation_Guide
>> >
>> > Now my version number says:
>> > FreeSWITCH Version 1.0.head (git-)
>> >
>> > Is there a mistake in my procedure or the building of FS when using
>> > GIT?? Hard to know the build number of FS with a tag like that!
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From william.suffill at gmail.com  Thu Apr  8 21:01:00 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 9 Apr 2010 00:01:00 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com> 
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com> 
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
Message-ID: <g2p6b65470d1004082101z30248c1an110072b83551131d@mail.gmail.com>

version is set during the build process so until you recompile freeswitch
the version reported by fs_cli won't change.
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From lakindia89 at gmail.com  Thu Apr  8 21:38:35 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 9 Apr 2010 10:08:35 +0530
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <4BBCBD73.7020206@gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
	<4BBB9DEB.2040702@gmail.com>
	<v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
	<4BBCBD73.7020206@gmail.com>
Message-ID: <x2n7d79b3931004082138y6573df06hcfd093103a3f944b@mail.gmail.com>

Right now I can't provide you the ssh access to the box, since the company
policy doesn't allow to do so. But I can explain you more clearly about
the setup and my need.

The following is the setup.
>From a Telephone Exchange, a line will be connected to the FS-BOX,  in
span1.
We have an internal Hard PBX with some 4 extensions.
We have connected the Hard PBX to FS-BOX in span2.

I've configured span1 as PRI_CPE and span2 as PRI_NET.
span1 don't have any problems. Both incoming and outgoing works fine.
In span2, I was able to make incoming call. It rings the extension.
But when I make outgoing call from those extension, I got NO CIRCUIT OR
CHANNEL AVAILABLE.

Right now in sangoma_mgd log, I got the following when I make outgoing call.


Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Opening
/var/log/sangoma_pri/dchan_2.log
Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
3:134716232 id:65535]
Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
4:134716308 id:65535]

Moreover I also have an Asterisk Box with Digim card. It works fine there.
When i dial from extension, it reaches asterisk and it responds with
SETUP_ACK.

please guide me.


On Wed, Apr 7, 2010 at 10:44 PM, David Yat Sin <david.yatsin at gmail.com>
wrote:
>
> Can you provide me with SSH access to that box and a phone number I can
use to trigger an incoming call on span 2 to that box?
>
> You can email me at: dyatsin at sangoma.com
>
> David
>
> On 4/7/2010 4:48 AM, lakshmanan ganapathy wrote:
>
> Hi,
> I have set verbose=4 in smg_pri.conf.
> I started the wanrouter.
> Here is the sangoma_mgd.log
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: ================System
restart=============
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Version: 1.63
            =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Date: Feb 26 2010
            =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 15607
             =
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid:
===========================================
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Number of spans:2
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Verbosity set to:4
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Debug disabled (local:2)
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Boost disabled (local:6)
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:1 pri_cpe euroisdn
dChan:16
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:2 pri_net euroisdn
dChan:16
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Down
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(P) Local:
127.0.0.66:53001 Remote:127.0.0.65:53001
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(N) Local:
127.0.0.66:53000 Remote:127.0.0.65:53000
> Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:Version:103
> Apr  7 13:58:27 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Up
> Apr  7 13:58:31 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Down
> Apr  7 13:58:32 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Up
> Apr  7 13:58:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:37 last:13:58:26 grace:0)
> Apr  7 13:58:43 FMS-FreeSwitch sangoma_prid: Opening
/var/log/sangoma_pri/dchan_1.log
> Apr  7 13:58:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:47 last:13:58:37 grace:0)
> Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:58:57 last:13:58:47 grace:0)
> Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> Apr  7 13:59:07 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:07 last:13:58:57 grace:0)
> Apr  7 13:59:17 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:17 last:13:59:07 grace:0)
> Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:27 last:13:59:17 grace:0)
> Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> Apr  7 13:59:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:37 last:13:59:27 grace:0)
> Apr  7 13:59:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:47 last:13:59:37 grace:0)
> Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
(current:13:59:57 last:13:59:47 grace:0)
> Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
>
> >From freeswitch cli, if I say
> originate openzap/1/a/39114603 at g2 &park(), I got the following in the
sangoma_mgd.log.
>
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Outgoing call (Smg-ID:2)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2:Outgoing call ChanRq:1
Called-Nb[39114603] Calling-Nb[Unknown] (Smg-ID:2)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c1:Remote
released-Unallocated (unassigned) number(1)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c0:Call was already cleared
(TSOFT-ID:4)
> Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Call cleared (SMG-ID:2)
>
> By seeing this, I confirmed that span2 is working. Leave out the
Unallocated number, that's my internal problem.
>
> >From the extension, if I dial 0, nothing is printing in sangoma_mgd.log
> But in dhcan2.log I got the No/Circuit or channel available as I mentioned
earlier.
>
> In freeswitch mod_openzap is loaded properly.
> Any help!!!
>
>
> On Wed, Apr 7, 2010 at 2:17 AM, David Yat Sin <david.yatsin at gmail.com>
wrote:
>>
>> Hi Lakshmanan,
>> If you do not have anything printing in /var/log/sangoma_mgd.log, and you
have:
>> verbose=4           //(or higher)
>>
>> in /etc/wanpipe/smg_pri.conf, check that you have these lines in
/etc/syslog.conf (or /etc/rsyslog.conf):
>>
>> local2.*                /var/log/sangoma_mgd.log
>>
>> and restart your syslog.
>>
>>
>> my first guess is that you do not have openzap loaded so sangoma_prid is
rejecting all incoming calls, but I would need logs in
/var/log/sangoma_mgd.log to confirm.
>>
>> If openzap is not loaded, you can type:
>> load mod_openzap
>>
>> from the freeswitch CLI to load it.
>>
>> --
>>
>> David Yat Sin, BEng, Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
9T3 Canada
>> t. 1 905 474 1990 x 119 | e. dyatsin at sangoma.com
>>
>>
>>
>> On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
>>
>> Hi all,
>> In my office we have a Hard PBX, with some 4 extensions.
>> We also have sangoma A102 card.
>> >From the Hard PBX, if 0 is pressed, it is setup in a way that it will go
to outside world.
>> I've connected that line to span2 of the card.
>> The span2 in the A102 card, is configured as PRI_NET.
>>
>> wanrouter status shows connected for the span2.
>>
>> But if I dial from the extension, I got the following in the
sangoma_dchan log.
>> 2010-04-03 12:49:49
>> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 33
39 31 31 34 36 30 30 7d 02 91 81 ]
>>   Call Ref:0164
>>   Type:Setup (0x5)
>>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
Kbit/s(16) L1Prot:G.711 A-Law(3)
>>   Calling Party Number:4439114600(l:10) plan:isdn(1)
type:unknown(0)scr:user, passed(1) pres:allowed(0)
>>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>>
>> 2010-04-03 12:49:49
>> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>>   Call Ref:0164
>>   Type:Release Compl (0x5a)
>>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
Circuit/Channel Available(34)
>>
>> The call in not reaching freeswitch ( I enabled debug log. But nothing is
printing in it ).
>> Can someone suggest how to make this work.
>>
>> Please ask me if you need more information?, since I don't know what to
give 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mustafa.pk at gmail.com  Thu Apr  8 22:13:12 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 9 Apr 2010 10:13:12 +0500
Subject: [Freeswitch-users] xml_curl loading/unloading user specific
	gateways.
Message-ID: <s2n8213d6071004082213v5ed23c42l39d0845f9a7ef99c@mail.gmail.com>

Hi,

is there a way to load a user specific gateway on sofia::register
event, and killgw/unregister gateway on sofia::unregister event, i
know this can be accomplished in many ways. but i am looking for a
cleaner and proper way, anyone?

thanks and best regards,

-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From david.ponzone at gmail.com  Thu Apr  8 22:37:30 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 9 Apr 2010 07:37:30 +0200
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
Message-ID: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>

I dont think there is a lowest value, but 30 seconds is reasonable in  
most cases.
You can also add a ping parameter with value 30, in the user config  
with the variables.
The result is a SIP OPTIONS sent to the phone every X sec.
That's quite the equivalent of qualify=yes in Asterisk.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :

> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
>
> Yes Asterisk config using the qualify=yes.
>
> I will modify the Registration time out by 50sec and see how it is  
> behaving.
>
> Thanks
> Lloyd
>
>
>
>
> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone  
> <david.ponzone at gmail.com> wrote:
> On most low-end routers, the NAT table will expire UDP translations  
> after 60 sec.
> Did you configure your phones to send a NAT keep-alive every X  
> seconds, with X < 60 ?
> You can also use sip-force-expires on the FS side.
>
> In your Asterisk config, do you use qualify=yes ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>
>> Hi All,
>>
>> I am having lots of problem with NAT and FreeSWITCH.
>>
>> FreeeSWITCH running Public IP.
>>
>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider  
>> Router/Modem : SMC Gateway.
>>
>> The Phone Losing the Registrations. The Environment working for  
>> Asterisk without any problem.
>>
>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying  
>> make FreeSWITCH work.
>>
>> How to solve this problem.
>>
>> Thanks,
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr  8 22:41:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 22:41:08 -0700
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <x2n7d79b3931004082138y6573df06hcfd093103a3f944b@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
	<4BBB9DEB.2040702@gmail.com>
	<v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
	<4BBCBD73.7020206@gmail.com>
	<x2n7d79b3931004082138y6573df06hcfd093103a3f944b@mail.gmail.com>
Message-ID: <n2t87f2f3b91004082241m1262877fsb2b596671c3d5f2@mail.gmail.com>

On Thu, Apr 8, 2010 at 9:38 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Right now I can't provide you the ssh access to the box, since the company
> policy doesn't allow to do so.


You might want to ask the company if they can lift that policy for the sake
of saving everyone's time and energy. I know we have some professionals who
would be willing to sign a reasonable NDA or whatnot to ensure security.
-MC


> But I can explain you more clearly about
> the setup and my need.
>
> The following is the setup.
> >From a Telephone Exchange, a line will be connected to the FS-BOX,  in
> span1.
> We have an internal Hard PBX with some 4 extensions.
> We have connected the Hard PBX to FS-BOX in span2.
>
> I've configured span1 as PRI_CPE and span2 as PRI_NET.
> span1 don't have any problems. Both incoming and outgoing works fine.
> In span2, I was able to make incoming call. It rings the extension.
> But when I make outgoing call from those extension, I got NO CIRCUIT OR
> CHANNEL AVAILABLE.
>
> Right now in sangoma_mgd log, I got the following when I make outgoing
> call.
>
> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Opening
> /var/log/sangoma_pri/dchan_2.log
> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
> 3:134716232 id:65535]
> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
> 4:134716308 id:65535]
>
> Moreover I also have an Asterisk Box with Digim card. It works fine there.
> When i dial from extension, it reaches asterisk and it responds with
> SETUP_ACK.
>
> please guide me.
>
>
> On Wed, Apr 7, 2010 at 10:44 PM, David Yat Sin <david.yatsin at gmail.com>
> wrote:
> >
> > Can you provide me with SSH access to that box and a phone number I can
> use to trigger an incoming call on span 2 to that box?
> >
> > You can email me at: dyatsin at sangoma.com
> >
> > David
> >
> > On 4/7/2010 4:48 AM, lakshmanan ganapathy wrote:
> >
> > Hi,
> > I have set verbose=4 in smg_pri.conf.
> > I started the wanrouter.
> > Here is the sangoma_mgd.log
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: ================System
> restart=============
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
> Daemon       =
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Version: 1.63
>               =
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Date: Feb 26 2010
>               =
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
> wanpipe-3.5.8.6       =
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 15607
>              =
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid:
> ===========================================
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Number of spans:2
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Verbosity set to:4
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Debug disabled (local:2)
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Boost disabled (local:6)
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:1 pri_cpe
> euroisdn dChan:16
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:2 pri_net
> euroisdn dChan:16
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Down
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(P) Local:
> 127.0.0.66:53001 Remote:127.0.0.65:53001
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(N) Local:
> 127.0.0.66:53000 Remote:127.0.0.65:53000
> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:Version:103
> > Apr  7 13:58:27 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Up
> > Apr  7 13:58:31 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Down
> > Apr  7 13:58:32 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Up
> > Apr  7 13:58:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:58:37 last:13:58:26 grace:0)
> > Apr  7 13:58:43 FMS-FreeSwitch sangoma_prid: Opening
> /var/log/sangoma_pri/dchan_1.log
> > Apr  7 13:58:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:58:47 last:13:58:37 grace:0)
> > Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:58:57 last:13:58:47 grace:0)
> > Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> > Apr  7 13:59:07 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:07 last:13:58:57 grace:0)
> > Apr  7 13:59:17 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:17 last:13:59:07 grace:0)
> > Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:27 last:13:59:17 grace:0)
> > Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> > Apr  7 13:59:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:37 last:13:59:27 grace:0)
> > Apr  7 13:59:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:47 last:13:59:37 grace:0)
> > Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
> (current:13:59:57 last:13:59:47 grace:0)
> > Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: Assuming application is dead
> >
> > >From freeswitch cli, if I say
> > originate openzap/1/a/39114603 at g2 &park(), I got the following in the
> sangoma_mgd.log.
> >
> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Outgoing call (Smg-ID:2)
> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2:Outgoing call ChanRq:1
> Called-Nb[39114603] Calling-Nb[Unknown] (Smg-ID:2)
> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c1:Remote
> released-Unallocated (unassigned) number(1)
> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c0:Call was already
> cleared (TSOFT-ID:4)
> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Call cleared (SMG-ID:2)
> >
> > By seeing this, I confirmed that span2 is working. Leave out the
> Unallocated number, that's my internal problem.
> >
> > >>From the extension, if I dial 0, nothing is printing in sangoma_mgd.log
> > But in dhcan2.log I got the No/Circuit or channel available as I
> mentioned earlier.
> >
> > In freeswitch mod_openzap is loaded properly.
> > Any help!!!
> >
> >
> > On Wed, Apr 7, 2010 at 2:17 AM, David Yat Sin <david.yatsin at gmail.com>
> wrote:
> >>
> >> Hi Lakshmanan,
> >> If you do not have anything printing in /var/log/sangoma_mgd.log, and
> you have:
> >> verbose=4           //(or higher)
> >>
> >> in /etc/wanpipe/smg_pri.conf, check that you have these lines in
> /etc/syslog.conf (or /etc/rsyslog.conf):
> >>
> >> local2.*                /var/log/sangoma_mgd.log
> >>
> >> and restart your syslog.
> >>
> >>
> >> my first guess is that you do not have openzap loaded so sangoma_prid is
> rejecting all incoming calls, but I would need logs in
> /var/log/sangoma_mgd.log to confirm.
> >>
> >> If openzap is not loaded, you can type:
> >> load mod_openzap
> >>
> >> from the freeswitch CLI to load it.
> >>
> >> --
> >>
> >> David Yat Sin, BEng, Software Developer
> >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> >> t. 1 905 474 1990 x 119 | e. dyatsin at sangoma.com
> >>
> >>
> >>
> >> On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
> >>
> >> Hi all,
> >> In my office we have a Hard PBX, with some 4 extensions.
> >> We also have sangoma A102 card.
> >> >From the Hard PBX, if 0 is pressed, it is setup in a way that it will
> go to outside world.
> >> I've connected that line to span2 of the card.
> >> The span2 in the A102 card, is configured as PRI_NET.
> >>
> >> wanrouter status shows connected for the span2.
> >>
> >> But if I dial from the extension, I got the following in the
> sangoma_dchan log.
> >> 2010-04-03 12:49:49
> >> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34
> 33 39 31 31 34 36 30 30 7d 02 91 81 ]
> >>   Call Ref:0164
> >>   Type:Setup (0x5)
> >>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0)
> TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3)
> >>   Calling Party Number:4439114600(l:10) plan:isdn(1)
> type:unknown(0)scr:user, passed(1) pres:allowed(0)
> >>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
> >>
> >> 2010-04-03 12:49:49
> >> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
> >>   Call Ref:0164
> >>   Type:Release Compl (0x5a)
> >>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
> Circuit/Channel Available(34)
> >>
> >> The call in not reaching freeswitch ( I enabled debug log. But nothing
> is printing in it ).
> >> Can someone suggest how to make this work.
> >>
> >> Please ask me if you need more information?, since I don't know what to
> give 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
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kond at nstel.ru  Thu Apr  8 23:36:59 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 9 Apr 2010 10:36:59 +0400
Subject: [Freeswitch-users] call bargein (urgent call)
Message-ID: <20100409063659.6DBF711492@mail.nstel.ru>

Hi all,
 
i need a sort of  "call barge in" functianality, ar may be "urgent call" will be better name... 
I know about existing "eavesdrop" application, but it does not suite me because of two reasons: 
1. callee must be on active call, or eavesdrop will fail.
2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If 1003 hangs up, the call will terminate.
 
Existing "intercept" application also does not suite.
 
So i need the following: 
A dispatcher must be able to call any user urgently, that is if a user is idle, it just should be a call, and when a user A is talking to user B, and dispatcher makes urgent call to A, all three must occur in a conference.
I'm sure it's possible, but i'm rather new to FS and looks like i'm lost while searching how to achive it....
 
Can anybody please advise how to do it?
 
Thanks in advance,
Nikolay.
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From msc at freeswitch.org  Thu Apr  8 23:48:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 23:48:28 -0700
Subject: [Freeswitch-users] call bargein (urgent call)
In-Reply-To: <20100409063659.6DBF711492@mail.nstel.ru>
References: <20100409063659.6DBF711492@mail.nstel.ru>
Message-ID: <k2x87f2f3b91004082348j694a7bb5pff5bdb1f63654b9a@mail.gmail.com>

On Thu, Apr 8, 2010 at 11:36 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
>
> i need a sort of  "call barge in" functianality, ar may be "urgent call"
> will be better name...
> I know about existing "eavesdrop" application, but it does not suite me
> because of two reasons:
> 1. callee must be on active call, or eavesdrop will fail.
> 2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If
> 1003 hangs up, the call will terminate.
>
> Existing "intercept" application also does not suite.
>
> So i need the following:
> A dispatcher must be able to call any user urgently, that is if a user is
> idle, it just should be a call, and when a user A is talking to user B, and
> dispatcher makes urgent call to A, all three must occur in a conference.
> I'm sure it's possible, but i'm rather new to FS and looks like i'm lost
> while searching how to achive it....
>
> Can anybody please advise how to do it?
>

So in other words, if the dispatcher calls A, and A's phone is idle it just
rings normally.
If the dispatcher calls A while A is connected to B, then you want to throw
A, B, and the dispatcher all into a conference?
Just confirming that I understand what is happening.
-MC
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From maciej.aniserowicz at gmail.com  Thu Apr  8 23:58:50 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Apr 2010 22:58:50 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <4BB1A30A.4060701@gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
Message-ID: <1270796330100-4875677.post@n2.nabble.com>


Hi,
I'll refresh this subject a little.
Here is what I found in this matter and tried to use to make this error
disappear:

execute_on_answer
record_session
bridge_pre_execute_aleg_app
media_bug_answer_req=true

I put these in various configurations, in dialstrings and dialplans.
Unfortunately the error still appears, none of these solved my issue.

Is this the right direction? Ideally I'd like to control the recording by
sending commands via event socket, just like I'm doing it now after
bridging.

Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4875677.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kond at nstel.ru  Fri Apr  9 00:07:30 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 9 Apr 2010 11:07:30 +0400
Subject: [Freeswitch-users] call bargein (urgent call)
In-Reply-To: <k2x87f2f3b91004082348j694a7bb5pff5bdb1f63654b9a@mail.gmail.com>
Message-ID: <20100409070730.C3CD31151B@mail.nstel.ru>

Yes, that is exactly what i want.
Nikolay.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, April 09, 2010 10:48 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call bargein (urgent call)





On Thu, Apr 8, 2010 at 11:36 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:


Hi all,
 
i need a sort of  "call barge in" functianality, ar may be "urgent call"
will be better name... 
I know about existing "eavesdrop" application, but it does not suite me
because of two reasons: 
1. callee must be on active call, or eavesdrop will fail.
2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If
1003 hangs up, the call will terminate.
 
Existing "intercept" application also does not suite.
 
So i need the following: 
A dispatcher must be able to call any user urgently, that is if a user is
idle, it just should be a call, and when a user A is talking to user B, and
dispatcher makes urgent call to A, all three must occur in a conference.
I'm sure it's possible, but i'm rather new to FS and looks like i'm lost
while searching how to achive it....
 
Can anybody please advise how to do it?


So in other words, if the dispatcher calls A, and A's phone is idle it just
rings normally.
If the dispatcher calls A while A is connected to B, then you want to throw
A, B, and the dispatcher all into a conference?
Just confirming that I understand what is happening.
-MC



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From excelsio at gmx.net  Fri Apr  9 03:46:01 2010
From: excelsio at gmx.net (excelsio at gmx.net)
Date: Fri, 09 Apr 2010 12:46:01 +0200
Subject: [Freeswitch-users] loop within conference
Message-ID: <20100409104601.313340@gmx.net>

Hi,

I setup a conference following the hints on: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR

So far, I can dial in:

- I?m told to enter the conference room number => ok
- I?m told to enter the moderator pin for that conference room => ok

But then I?m again told to enter the conference number and again the key for it and again and again and again.


Let?s see output:

2010-04-09 14:30:15.768367 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/external/1333 at 188.168.2.1] has been answered
2010-04-09 14:30:15.768367 [INFO] conf-ivr.js:14 Prompting for Conference Room...
2010-04-09 14:30:19.621485 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#]
2010-04-09 14:30:19.621485 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888]
2010-04-09 14:30:20.441945 [INFO] conf-ivr.js:14 Prompting for Moderator Password...
2010-04-09 14:30:21.682133 [INFO] conf-ivr.js:14 Password Digits Collected = [777#]
2010-04-09 14:30:21.682133 [INFO] conf-ivr.js:14 This is the first person in the conference
2010-04-09 14:30:24.721511 [INFO] conf-ivr.js:14 Entering conference room [666] as a [MODERATOR]
2010-04-09 14:30:27.142215 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333 at 188.168.2.1 to XML[conf-room--666 at normal-conference+flags{moderator}@public]
2010-04-09 14:30:27.146007 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666 at normal-conference+flags{moderator} in context public
2010-04-09 14:30:27.162559 [INFO] conf-ivr.js:14 Prompting for Conference Room...
2010-04-09 14:30:32.382171 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#]
2010-04-09 14:30:32.382171 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888]
2010-04-09 14:30:33.202121 [INFO] conf-ivr.js:14 Prompting for Moderator Password...
2010-04-09 14:30:36.101702 [INFO] conf-ivr.js:14 Password Digits Collected = [777#]
2010-04-09 14:30:36.101702 [INFO] conf-ivr.js:14 This is the first person in the conference
2010-04-09 14:30:39.141451 [INFO] conf-ivr.js:14 Entering conference room [666] as a [MODERATOR]
2010-04-09 14:30:41.562049 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333 at 188.168.2.1 to XML[conf-room--666 at normal-conference+flags{moderator}@public]



=> So, there?s obviously a kind of loop. But where does this occur?


Also, I have this behavior, when I enter no password. In that case I?m a normal user:

2010-04-09 14:44:56.355377 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->3975 in context public
2010-04-09 14:44:56.377115 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/external/1333 at 188.168.2.1] has been answered
2010-04-09 14:44:56.377115 [INFO] conf-ivr.js:14 Prompting for Conference Room...
2010-04-09 14:45:00.322060 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#]
2010-04-09 14:45:00.322060 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888]
2010-04-09 14:45:01.141552 [INFO] conf-ivr.js:14 Prompting for Moderator Password...
2010-04-09 14:45:03.542187 [INFO] conf-ivr.js:14 Password Digits Collected = [#]
2010-04-09 14:45:03.542187 [INFO] conf-ivr.js:14 This is the first person in the conference
2010-04-09 14:45:06.581903 [INFO] conf-ivr.js:14 Entering conference room [666] as a [USER]
2010-04-09 14:45:06.581903 [INFO] conf-ivr.js:14 Moderator is NOT present in the room
2010-04-09 14:45:09.222209 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333 at 188.168.2.1 to XML[conf-room--666 at normal-conference@public]
Error in my_thread_global_end(): 1 threads didn't exit
2010-04-09 14:45:14.225453 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666 at normal-conference in context public
2010-04-09 14:45:14.242444 [INFO] conf-ivr.js:14 Prompting for Conference Room...
2010-04-09 14:45:18.341791 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#]
2010-04-09 14:45:18.341791 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888]
2010-04-09 14:45:19.162446 [INFO] conf-ivr.js:14 Prompting for Moderator Password...
2010-04-09 14:45:21.722027 [INFO] conf-ivr.js:14 Password Digits Collected = [#]
2010-04-09 14:45:21.722027 [INFO] conf-ivr.js:14 This is the first person in the conference
2010-04-09 14:45:24.762133 [INFO] conf-ivr.js:14 Entering conference room [666] as a [USER]
2010-04-09 14:45:24.762133 [INFO] conf-ivr.js:14 Moderator is NOT present in the room
2010-04-09 14:45:27.401371 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333 at 188.168.2.1 to XML[conf-room--666 at normal-conference@public]
Error in my_thread_global_end(): 1 threads didn't exit
2010-04-09 14:45:32.405295 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666 at normal-conference in context public
2010-04-09 14:45:32.423449 [INFO] conf-ivr.js:14 Prompting for Conference Room...

=> Again that loop


So, where?s the mistake I made?

Michael
-- 
GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01


From excelsio at gmx.net  Fri Apr  9 05:46:10 2010
From: excelsio at gmx.net (excelsio at gmx.net)
Date: Fri, 09 Apr 2010 14:46:10 +0200
Subject: [Freeswitch-users] loop within conference
In-Reply-To: <20100409104601.313340@gmx.net>
References: <20100409104601.313340@gmx.net>
Message-ID: <20100409124610.208680@gmx.net>

Found the mistake. I did have an

<condition field="destination_number" expression=".*">

Had to exchange the .* to the called number
-- 
GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01


From bcxml at hotmail.com  Fri Apr  9 07:33:19 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 06:33:19 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
Message-ID: <1270823599661-4877398.post@n2.nabble.com>



I tried what you mentioned but it did not seem to have any effect. I am
still losing part of the openning message

Here is my inbound dialplan..

<extension name="Inbound_1234567890"> 
  <condition field='destination_number' expression='1234567890'> 
    <action application="playback" data="silence_stream://1000"/> 
    <action application="bridge"
data="{bypass_media=true}sofia/internal/1234567890 at 127.0.0.1:5060;transport=tcp"/> 
  </condition> 
</extension>


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4877398.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From srinivas.ksvreddy at gmail.com  Fri Apr  9 08:04:31 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 9 Apr 2010 20:34:31 +0530
Subject: [Freeswitch-users] compile errors
Message-ID: <j2uf8af5741004090804n24e41535y4d15c8a684237e77@mail.gmail.com>

Hi,

i am compiling the freeswitch1.0.2 its giving compilation errors in
libsofia_sip_ua_static project. my enviroment is windows xp.
any idea

-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Fri Apr  9 08:12:05 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 9 Apr 2010 10:12:05 -0500
Subject: [Freeswitch-users] compile errors
In-Reply-To: <j2uf8af5741004090804n24e41535y4d15c8a684237e77@mail.gmail.com>
References: <j2uf8af5741004090804n24e41535y4d15c8a684237e77@mail.gmail.com>
Message-ID: <282AE6EE-F19F-41FE-8D08-79728F1570C4@freeswitch.org>

Please use FreeSWITCH 1.0.6

/b

On Apr 9, 2010, at 10:04 AM, srinivasula reddy wrote:

> Hi,
>  
> i am compiling the freeswitch1.0.2 its giving compilation errors in libsofia_sip_ua_static project. my enviroment is windows xp.
> any idea
> 
> -- 
> Srinivasula Reddy K



From peter.olsson at visionutveckling.se  Fri Apr  9 08:17:16 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 9 Apr 2010 17:17:16 +0200
Subject: [Freeswitch-users] compile errors
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4C17@cooper>

A good start is to use a current version. Download 1.0.6 and try again.

/Peter


________________________________
Fr?n: srinivasula reddy <srinivas.ksvreddy at gmail.com>
Skickat: den 9 april 2010 17:11
Till: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
?mne: [Freeswitch-users] compile errors

Hi,

i am compiling the freeswitch1.0.2 its giving compilation errors in libsofia_sip_ua_static project. my enviroment is windows xp.
any idea

--
Srinivasula Reddy K
!DSPAM:4bbf439232931558811689!


From 12ukwn at gmail.com  Fri Apr  9 08:24:12 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 9 Apr 2010 17:24:12 +0200
Subject: [Freeswitch-users] git question
Message-ID: <20100409172412.3a0ce3b8@anubis.defcon1>

Hi list,

I now use git but I'm not sure about things I've got to do after an upgrade:
is it mandatory to re-do the whole cycle (bootstrap + configure), or just
configure, or nothing at all but a make install?

-- 
  The best cure for insomnia is to get a lot of sleep. -W. C. Fields


From 12ukwn at gmail.com  Fri Apr  9 08:43:55 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 9 Apr 2010 17:43:55 +0200
Subject: [Freeswitch-users] git other question
Message-ID: <20100409174355.7cc69bf4@anubis.defcon1>

Hi list,

About the contrib tree, shall I build it out FS tree or inside?

-- 
Turn the other cheek.
		-- Jesus Christ


From moises.silva at gmail.com  Fri Apr  9 09:04:43 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 9 Apr 2010 12:04:43 -0400
Subject: [Freeswitch-users] transcoding
In-Reply-To: <528D6591-3E12-469E-8DFA-9AF01F368630@freeswitch.org>
References: <4BBDB50F.9070600@baigtel.com>
	<4FF7CCE3-ABFC-4BF2-A999-7276BD814547@freeswitch.org>
	<x2xc4d05cbe1004081247qc8644eeax1f2afa4667c958e9@mail.gmail.com>
	<528D6591-3E12-469E-8DFA-9AF01F368630@freeswitch.org>
Message-ID: <i2mc4d05cbe1004090904xe9682052z79f836785166e720@mail.gmail.com>

Yes, also may be mod_sofia should be notified ( I think there is a way
already to notify about module loading/unloading ) so the list of codecs is
reloaded, cuz right now if mod_sofia is loaded and later on a codec module
is added, sofia does not know about it, or that's the impression I got when
testing.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, Apr 8, 2010 at 3:59 PM, Brian West <brian at freeswitch.org> wrote:

> You know we should deny load if something is already registered maybe?
>
> Cuz I can see this being a problem.
>
> /b
>
> On Apr 8, 2010, at 2:47 PM, Moises Silva wrote:
>
> and I think mod_g729 (pass-thru codec) should not be loaded either.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Thu, Apr 8, 2010 at 1:58 PM, Brian West <brian at freeswitch.org> wrote:
>
>> make sure mod_g723_1 isn't loaded.
>>
>> /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
>
>
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From david.ponzone at gmail.com  Fri Apr  9 09:06:34 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 9 Apr 2010 18:06:34 +0200
Subject: [Freeswitch-users] git question
In-Reply-To: <20100409172412.3a0ce3b8@anubis.defcon1>
References: <20100409172412.3a0ce3b8@anubis.defcon1>
Message-ID: <79B8F075-489D-4F3B-AB4B-EF0B52FE152A@gmail.com>

Jean-Yves,

make; make install

should be enough

but make install should recompile what's necessary anwyay

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 09/04/2010 ? 17:24, Jean-Yves F. Barbier a ?crit :

> Hi list,
>
> I now use git but I'm not sure about things I've got to do after an  
> upgrade:
> is it mandatory to re-do the whole cycle (bootstrap + configure), or  
> just
> configure, or nothing at all but a make install?
>
> -- 
>  The best cure for insomnia is to get a lot of sleep. -W. C. Fields
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nazim.aghabayov at gmail.com  Fri Apr  9 09:17:37 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 09 Apr 2010 21:17:37 +0500
Subject: [Freeswitch-users] git other question
In-Reply-To: <20100409174355.7cc69bf4@anubis.defcon1>
References: <20100409174355.7cc69bf4@anubis.defcon1>
Message-ID: <4BBF5321.4050707@gmail.com>

Hi,

Majority of code in freeswitch-contrib could be compiled outside of
tree, except may be for modules I guess.

Nazim

On 04/09/2010 08:43 PM, Jean-Yves F. Barbier wrote:
> Hi list,
>
> About the contrib tree, shall I build it out FS tree or inside?
>
>   



From djbinter at yahoo.com  Fri Apr  9 09:19:43 2010
From: djbinter at yahoo.com (DJB)
Date: Fri, 9 Apr 2010 09:19:43 -0700 (PDT)
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
Message-ID: <81019.71788.qm@web37501.mail.mud.yahoo.com>

git pull 
make all
make install

-or-

make current

-djbinter



________________________________
From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, April 8, 2010 8:48:17 PM
Subject: Re: [Freeswitch-users] version number: git checkout

Thanks Milena... I upgraded git now - git version 1.7.0.4

I did a 'get pull && make install' and still the same problem.

Do I have to do a get clone or something?  Ideas?  Thanks

freeswitch:~# git --version
git version 1.7.0.4
freeswitch:~# fs_cli
            _____ ____     ____ _     ___
           |  ___/ ___|   / ___| |   |_ _|
           | |_  \___ \  | |   | |    | |
           |  _|  ___) | | |___| |___ | |
           |_|   |____/   \____|_____|___|

*******************************************************
* Anthony Minessale II, Ken Rice, Michael Jerris      *
* FreeSWITCH (http://www.freeswitch.org)              *
* Paypal Donations Appreciated: paypal at freeswitch.org *
* Brought to you by ClueCon http://www.cluecon.com/   *
*******************************************************

Type /help <enter> to see a list of commands


+OK log level  [7]
freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-)

On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
>
>
> Can anything be done in the freeswitch code so when "git pull" is executed,
> the "--pretty" argument is also set where "--format" is set to make it
> compatible with both older and newer versions of git? or it is all up to
> what git does and nothing to do on fs?
>
>
>
> PS: Mark, the issue you're facing is because of your version of git, the CLI
> shows the freeswitch version properly with git 1.7.0.4, the "format"
> argument isn't recognized by your version of git.
>
>
> 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
>>
>> Git was installed as described on the wiki.  I am using Debian Lenny
>> and Git version 1.5.6.5
>>
>> I just did a git pull and had the same issue... FS still shows
>> FreeSWITCH Version 1.0.head (git-)
>>
>>
>>
>> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
>> > Upgrade your git, then it will show it correctly.
>> > http://wiki.freeswitch.org/wiki/Git_Install
>> > djbinter
>> > ________________________________
>> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
>> > To: freeswitch-users at lists.freeswitch.org
>> > Sent: Wed, April 7, 2010 11:54:27 PM
>> > Subject: [Freeswitch-users] version number: git checkout
>> >
>> > Hi!
>> >
>> > I just used git for the first time ever to checkout FreeSwitch as
>> > described on the wiki at
>> > http://wiki.freeswitch.org/wiki/Installation_Guide
>> >
>> > Now my version number says:
>> > FreeSWITCH Version 1.0.head (git-)
>> >
>> > Is there a mistake in my procedure or the building of FS when using
>> > GIT?  Hard to know the build number of FS with a tag like that!
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Apr  9 09:25:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Apr 2010 11:25:43 -0500
Subject: [Freeswitch-users] move on from asterisk,
	the Random or rand 	function
In-Reply-To: <y2j87f2f3b91004081629ncbe6b11eo8d8886db17c1331f@mail.gmail.com>
References: <u2l9372a8331004081329yba87f64n3b09e96663fe575c@mail.gmail.com>
	<y2j87f2f3b91004081629ncbe6b11eo8d8886db17c1331f@mail.gmail.com>
Message-ID: <l2h191c3a031004090925sebbc48acqd3e6a3b67d3e487a@mail.gmail.com>

moderate news:
there is the expr app that is not exactly intuitive but can generate a
random number as a function variable

${expr(randomize(&x);ceil(random(0,10,&x)))}

This will eval to a random number from 1 to 10 (always start one number less
than where you want to start)

see: http://wiki.freeswitch.org/files/expr.html


On Thu, Apr 8, 2010 at 6:29 PM, Michael Collins <msc at freeswitch.org> wrote:

> Good news, bad news:
>
> The good news is that the XML dialplan is a very efficient way of handling
> call routing. The bad news is that the XML dialplan is not a programming
> language. There is no "rand" function, so you'll need to do something
> different. One option is to use mod_distributor, however it is not "random"
> so that may not work for you. The other option is to call Lua or system or
> something and snag a random number.
>
> -MC
>
>
> On Thu, Apr 8, 2010 at 1:29 PM, Krzysztof Drewicz <
> krzysztofdrewicz at gmail.com> wrote:
>
>> Hi,
>>
>> i'm slowly, but constantly moving from asterisk to fs.
>>
>> Stuck on translating this into a freeswitch:
>>
>> exten => s,1,Random(90:s,3)
>> exten => s,2,Noop(his is hit 10% times)
>> exten => s,3,Dial(SIP/8000)
>>
>> What i need to do is put a bucket on some calling_number + destination
>> numer
>> with some random percent to a voiceguide and hangup.
>>
>> i've found that memcache is very convinent way to put data into
>> dialplan logic in runtime, so i plan to do:
>>
>> <extension name="bucket_part_01" continue="true">
>>        <condition field="${do_bucket}" expression=".+" break="on-true"/>
>>        <condition field="caller_id_number" expression="^123456789$">
>>        <action data="do_bucket=${memcache(get
>> b_${caller_id_number})}" application="set"/>
>>        <action data="bucket_part_02" application="transfer"/>
>>        <action application="hangup"/>
>>        </condition>
>> </extension>
>>
>> So above i check for caller_id_number in the memchache b_123456789 with
>> option 1
>> Then i transfer to anoter extension:
>>
>>  <extension name="bucket_part_02">
>>        <condition field="destination_number" expression="bucket_part_02"/>
>>        <condition field="${czy_robic}" expression="^1$">
>>                <action data="bypass_media=true" application="set"/>
>>                <action data="hits=${memcache(increment
>> hit_bucket_test)}" application="set"/>
>>
>> AND HERE I do need some logic that does if Random() > 50%
>>
>>                <action application="answer"/>
>>                <action
>> data="/usr/local/freeswitch/sounds/pc/weRbusyRightNow.wav"
>> application="playback"/>
>>                <action application="hangup"/>
>>        </condition>
>> </extension>
>>
>>
>> any cleaner approach or someone has done it better?
>> note: don't want to use any lua, api or sth, just plain xml dialplan
>> logic.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From cucku.cucku at yahoo.com.vn  Fri Apr  9 09:29:29 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Sat, 10 Apr 2010 00:29:29 +0800 (SGT)
Subject: [Freeswitch-users] need help on variable and param
Message-ID: <332483.42922.qm@web76207.mail.sg1.yahoo.com>

Hi all

i am confuse the param and variable in freeswitch, sometimes param is used for gateway, sometime variable is used for users
example:
<param>
<param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
</param>

=============


<variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
      <variable name="user_context" value="company-b.org"/>
</variables>

what is difference between param and variable
how to use param and variable
is there a list of variable and param

Thank you


      B?n lu?n mu?n k?t n?i v?i nhi?u b?n b? h?n tr?n blog v? trang web c? nh?n? T?o Pingbox m?i nh?t ngay h?m nay! http://vn.messenger.yahoo.com/pingbox/
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From msc at freeswitch.org  Fri Apr  9 09:42:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 09:42:22 -0700
Subject: [Freeswitch-users] move on from asterisk,
	the Random or rand 	function
In-Reply-To: <l2h191c3a031004090925sebbc48acqd3e6a3b67d3e487a@mail.gmail.com>
References: <u2l9372a8331004081329yba87f64n3b09e96663fe575c@mail.gmail.com>
	<y2j87f2f3b91004081629ncbe6b11eo8d8886db17c1331f@mail.gmail.com>
	<l2h191c3a031004090925sebbc48acqd3e6a3b67d3e487a@mail.gmail.com>
Message-ID: <y2y87f2f3b91004090942zedd6786o95195835fd3cab6c@mail.gmail.com>

As usual, Tony rocks. I forgot that expr has all sorts of cool functions.
Check out that page Tony linked to, it has all sorts of interesting things.
Let me know if you have a valid reason to use atan2 (arc-tangent with
quadrant correction) in your dialplan. :P

-MC

On Fri, Apr 9, 2010 at 9:25 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> moderate news:
> there is the expr app that is not exactly intuitive but can generate a
> random number as a function variable
>
> ${expr(randomize(&x);ceil(random(0,10,&x)))}
>
> This will eval to a random number from 1 to 10 (always start one number
> less than where you want to start)
>
> see: http://wiki.freeswitch.org/files/expr.html
>
>
>
> On Thu, Apr 8, 2010 at 6:29 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Good news, bad news:
>>
>> The good news is that the XML dialplan is a very efficient way of handling
>> call routing. The bad news is that the XML dialplan is not a programming
>> language. There is no "rand" function, so you'll need to do something
>> different. One option is to use mod_distributor, however it is not "random"
>> so that may not work for you. The other option is to call Lua or system or
>> something and snag a random number.
>>
>> -MC
>>
>>
>> On Thu, Apr 8, 2010 at 1:29 PM, Krzysztof Drewicz <
>> krzysztofdrewicz at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> i'm slowly, but constantly moving from asterisk to fs.
>>>
>>> Stuck on translating this into a freeswitch:
>>>
>>> exten => s,1,Random(90:s,3)
>>> exten => s,2,Noop(his is hit 10% times)
>>> exten => s,3,Dial(SIP/8000)
>>>
>>> What i need to do is put a bucket on some calling_number + destination
>>> numer
>>> with some random percent to a voiceguide and hangup.
>>>
>>> i've found that memcache is very convinent way to put data into
>>> dialplan logic in runtime, so i plan to do:
>>>
>>> <extension name="bucket_part_01" continue="true">
>>>        <condition field="${do_bucket}" expression=".+" break="on-true"/>
>>>        <condition field="caller_id_number" expression="^123456789$">
>>>        <action data="do_bucket=${memcache(get
>>> b_${caller_id_number})}" application="set"/>
>>>        <action data="bucket_part_02" application="transfer"/>
>>>        <action application="hangup"/>
>>>        </condition>
>>> </extension>
>>>
>>> So above i check for caller_id_number in the memchache b_123456789 with
>>> option 1
>>> Then i transfer to anoter extension:
>>>
>>>  <extension name="bucket_part_02">
>>>        <condition field="destination_number"
>>> expression="bucket_part_02"/>
>>>        <condition field="${czy_robic}" expression="^1$">
>>>                <action data="bypass_media=true" application="set"/>
>>>                <action data="hits=${memcache(increment
>>> hit_bucket_test)}" application="set"/>
>>>
>>> AND HERE I do need some logic that does if Random() > 50%
>>>
>>>                <action application="answer"/>
>>>                <action
>>> data="/usr/local/freeswitch/sounds/pc/weRbusyRightNow.wav"
>>> application="playback"/>
>>>                <action application="hangup"/>
>>>        </condition>
>>> </extension>
>>>
>>>
>>> any cleaner approach or someone has done it better?
>>> note: don't want to use any lua, api or sth, just plain xml dialplan
>>> logic.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  9 10:01:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:01:59 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270796330100-4875677.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
Message-ID: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>

Can you link to the pastebin where you have a full debug log of this error
occurring?
-MC

On Thu, Apr 8, 2010 at 11:58 PM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hi,
> I'll refresh this subject a little.
> Here is what I found in this matter and tried to use to make this error
> disappear:
>
> execute_on_answer
> record_session
> bridge_pre_execute_aleg_app
> media_bug_answer_req=true
>
> I put these in various configurations, in dialstrings and dialplans.
> Unfortunately the error still appears, none of these solved my issue.
>
> Is this the right direction? Ideally I'd like to control the recording by
> sending commands via event socket, just like I'm doing it now after
> bridging.
>
> Maciej Aniserowicz
> --
> View this message in context:
> http://n2.nabble.com/Error-when-recording-tp4817081p4875677.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 msc at freeswitch.org  Fri Apr  9 10:03:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:03:38 -0700
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <1270823599661-4877398.post@n2.nabble.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
Message-ID: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>

Can you pastebin a full debug log of this call?
-MC

On Fri, Apr 9, 2010 at 7:33 AM, Brian Campbell <bcxml at hotmail.com> wrote:

>
>
> I tried what you mentioned but it did not seem to have any effect. I am
> still losing part of the openning message
>
> Here is my inbound dialplan..
>
> <extension name="Inbound_1234567890">
>  <condition field='destination_number' expression='1234567890'>
>    <action application="playback" data="silence_stream://1000"/>
>    <action application="bridge"
> data="{bypass_media=true}sofia/internal/1234567890 at 127.0.0.1:5060
> ;transport=tcp"/>
>  </condition>
> </extension>
>
>
> Brian
> --
> View this message in context:
> http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4877398.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 jasonchu at avaya.com  Fri Apr  9 08:27:06 2010
From: jasonchu at avaya.com (CHU, XINGJUN (XINGJUN))
Date: Fri, 9 Apr 2010 11:27:06 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
Message-ID: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>

I don't see how this is relevant to the problem?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
Sent: Thursday, April 08, 2010 9:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] An issue when attended transfer to fs

hi,

put an <action application="sleep" data="2000"/> before the instruction to play something

2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com<mailto:jasonchu at avaya.com>>
Hi,

I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.

Now I got a problem when A calls B then attended transfer B to freeswitch,
The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.

Any suggestions are greatly appreciated.

Thanks
Xingjun

_______________________________________________
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

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From noisewaterphd at gmail.com  Fri Apr  9 09:08:48 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 10:08:48 -0600
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
Message-ID: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>

Hi,

I've already read the interop list, but I'm wondering if anyone on here has
anymore experience/info on trunking freeswitch to a Mitel 3300?

Specifically, I want to use freeswitch for acd and sip registrations, and
just use our mitel for switching to the PSTN.

Does anyone have some good info to share?

Thanks,

Kenny
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From kenfulmer at icstechnologysolutions.com  Fri Apr  9 10:07:36 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Fri, 9 Apr 2010 12:07:36 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
Message-ID: <012d01cad807$27f5a0a0$77e0e1e0$@com>

Per your suggestion, I changed the following in the
conf/autoload_configs/acl.conf.xml file:

 

    <list name="domains" default="deny">

      <node type="allow" domain="10.10.3.10"/>

      <node type="allow" domain="10.10.3.11"/>

    </list>

 

10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
However, the calls still fail with the 407 Proxy Authentication Required
message. 

 

I get the following log output when I issue the command, reloadacl:

 

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
rfc1918.auto default (deny)

freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195
Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
wan.auto default (allow)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
nat.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
10.10.3.12/255.255.255.128 (deny) to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
loopback.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8
(allow) [] to list loopback.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
localnet.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
10.10.3.12/255.255.255.128 (allow) to list localnet.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
domains default (deny)

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.10

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.11

 

Am I doing something incorrectly?

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 08, 2010 6:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

 

On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Actually, I did purchase a license and installed it today. One call
establishes at 729. When I hang up the phone and try again, it's 711.

Make sure that the encoder/decoder isn't still in use prior to trying the
second call. After you hang up, do a "show channels" and see if the call is
still "up" or not. Also, do "g729_status" to see if the encoder or decoder
is in use. Keep doing "g729_status" until the 'coders are not in use. If
there is a long delay then open up a JIRA ticket on jira.freeswitch.org.

 

The Proxy Authentication Required is being sent by FreeSwitch to the
internal PBX. I have registration disabled on the FreeSwitch gateway and the
internal server. 

By default the SIP profile will challenge if the IP address of the caller is
not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
"domains" node. Add your PBX's IP address. You'll see an example in the
comments. Once you're done editing, save the file and then go to the fs_cli
and do:
reloadacl reloadxml

Then make a call from PBX to FS and it should go through.
-MC

 

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From maciej.aniserowicz at gmail.com  Fri Apr  9 10:14:35 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 9 Apr 2010 09:14:35 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270833275502-4878370.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4878370.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Apr  9 10:18:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:18:54 -0700
Subject: [Freeswitch-users] need help on variable and param
In-Reply-To: <332483.42922.qm@web76207.mail.sg1.yahoo.com>
References: <332483.42922.qm@web76207.mail.sg1.yahoo.com>
Message-ID: <l2t87f2f3b91004091018nc95fb71au829b6837777fd5ba@mail.gmail.com>

2010/4/9 false <cucku.cucku at yahoo.com.vn>

> Hi all
>
> i am confuse the param and variable in freeswitch, sometimes param is used
> for gateway, sometime variable is used for users
>

Remember that variables in this case are "channel variables" and are almost
always associated with a specific phone call. These variables for a user are
set whenever that user makes a phone call.

Parameters a generally used for configuration purposes. The "dial-string"
parameter is related to the user channel, so when you do something like:
originate user/1001 &park()

It will use the dial-string parameter to turn "user/1001" into a proper
dialstring. If it makes your brain hurt then don't worry about it. :)

Parameters are used in lots of configurations. The most obvious examples are
in the SIP profiles and gateways. Are you working on something specific
where the differences between params and variables are causing you trouble?

-MC
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From vfclists at googlemail.com  Mon Apr 12 05:22:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 12 Apr 2010 13:22:49 +0100
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
Message-ID: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>

Can Freeswitch be installed simply by zipping up the folder and unzipping it
to the destination?

Does it require some DLLs to be installed in the Windows system folder and
some registry entries as well?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From larclap at yahoo.com  Fri Apr  9 14:33:12 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:33:12 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011f01cad82c$420847c0$c618d740$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From jeff at jefflenk.com  Mon Apr 12 08:30:54 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 12 Apr 2010 10:30:54 -0500
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
Message-ID: <COL117-W71213F34122C75703A37CB9120@phx.gbl>


Sure the only requirement is the 2008 CRuntime support. No files or registry settings needed for basic operation.
 


Date: Mon, 12 Apr 2010 13:22:49 +0100
From: vfclists at googlemail.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?


Can Freeswitch be installed simply by zipping up the folder and unzipping it to the destination?

Does it require some DLLs to be installed in the Windows system folder and some registry entries as well?
-- 
Frank Church

=======================
http://devblog.brahmancreations.com
 		 	   		  
_________________________________________________________________
The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail.
http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From freeswitch.org at todandlorna.com  Mon Apr 12 08:39:40 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Mon, 12 Apr 2010 09:39:40 -0600
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
Message-ID: <4BC33EBC.40106@todandlorna.com>

I haven't played with every module, but the core requires nothing else 
to work on Windows, just the base freeswitch directory and all its 
sub-folders.

Cheers,

-Tod Hansmann

On 4/12/2010 6:22 AM, Frank Church wrote:
>
> Can Freeswitch be installed simply by zipping up the folder and 
> unzipping it to the destination?
>
> Does it require some DLLs to be installed in the Windows system folder 
> and some registry entries as well?
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 jalsot at gmail.com  Mon Apr 12 08:41:03 2010
From: jalsot at gmail.com (Tamas)
Date: Mon, 12 Apr 2010 17:41:03 +0200
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <COL117-W71213F34122C75703A37CB9120@phx.gbl>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
	<COL117-W71213F34122C75703A37CB9120@phx.gbl>
Message-ID: <4BC33F0F.8070208@gmail.com>

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From infos at madovsky.org  Sun Apr 11 20:38:15 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 11 Apr 2010 23:38:15 -0400
Subject: [Freeswitch-users] FS pid
Message-ID: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>

Hi all,

how can I change the name and the path of the pid file ?

Thanks

Franck
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From infos at madovsky.org  Sun Apr 11 20:38:15 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 11 Apr 2010 23:38:15 -0400
Subject: [Freeswitch-users] FS pid
Message-ID: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>

Hi all,

how can I change the name and the path of the pid file ?

Thanks

Franck
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From lloyd.aloysius at gmail.com  Mon Apr 12 09:00:17 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 12 Apr 2010 12:00:17 -0400
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <C7E62B51.802E%sean@obscuradigital.com>
References: <C7E62B51.802E%sean@obscuradigital.com>
Message-ID: <g2r8a19bf2e1004120900t758f1a04yfb7d0e3b02271b42@mail.gmail.com>

Here is the syntax I am using inside the ivr menus

 <entry action="menu-exec-app" digits="5" param="directory default
${domain_name} "/>

When the user press 5 .. the Directory Application Will start.
---

Here is the wiki.

http://wiki.freeswitch.org/wiki/Mod_directory

You need to download sound files manually.The users returned MUST be a
member of a group.

Thanks
Lloyd

On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hello,
>
> Trying to figure out how to use the mod directory app within an ivr menu.
>  Not sure of the syntax to use when building the ivr.  Not ever sure it can
> be done.
>
> Also does mod_directory need to be added to a dialplan?
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Mon Apr 12 09:03:49 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 12 Apr 2010 18:03:49 +0200
Subject: [Freeswitch-users] FS pid
In-Reply-To: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>
References: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>
Message-ID: <ADA82617-5F5A-4FD3-A2A5-3D6F26E98BFA@gmail.com>

Look in /etc/init.d/freeswitch.
pidfile is generally generated during startup.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 12/04/2010 ? 05:38, Madovsky a ?crit :

> Hi all,
>
> how can I change the name and the path of the pid file ?
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jonas.gauffin at gmail.com  Mon Apr 12 09:04:22 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 12 Apr 2010 18:04:22 +0200
Subject: [Freeswitch-users] Recommended phones
Message-ID: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>

Hello,

Which phone models are you using?

I'm looking for a budget phone and a standard phone.
Both should work well behind NAT (FS <--> internet <--> router <--> phones).

Regards,
  Jonas
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From david.ponzone at gmail.com  Mon Apr 12 09:07:17 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 12 Apr 2010 18:07:17 +0200
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
	<z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>
Message-ID: <2D93F01C-4643-4B9A-BA73-7020FAA9898A@gmail.com>

I think it would really help now if you give us the model of the SMC  
router.
Some routers have a SIP ALG that cannot be disabled, and those routers  
are ready-for-trash.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 09/04/2010 ? 21:40, Aloysius Lloyd a ?crit :

> I could not find a SIP ALG Setting. I setup the sip-force-expires  
> and ping for the user directory. Only one time registering then lost  
> the connection.
>
> Here is the sofia profile internal status
>
> Call-ID:        7307ef8fa6044407
> User:           202 at abc.com
> Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=192.168.1.51:5060 
> >
> Agent:          Aastra 9143i/2.5.2.30
> Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
> Host:           TestSrv
> IP:             A.B.C.D
> Port:           5060
> Auth-User:      202
> Auth-Realm:     abc.com
> MWI-Account:    202 at abc.com
>
> Please let me know how to fix this issue.
>
> Thanks
> Lloyd
>
>
> On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone  
> <david.ponzone at gmail.com> wrote:
> I dont think there is a lowest value, but 30 seconds is reasonable  
> in most cases.
> You can also add a ping parameter with value 30, in the user config  
> with the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
>> what is the lowest value I can use for sip-force-expires ?
>>
>> Thanks
>> Lloyd
>>
>>
>> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
>> > wrote:
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is  
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com 
>> > wrote:
>> On most low-end routers, the NAT table will expire UDP translations  
>> after 60 sec.
>> Did you configure your phones to send a NAT keep-alive every X  
>> seconds, with X < 60 ?
>> You can also use sip-force-expires on the FS side.
>>
>> In your Asterisk config, do you use qualify=yes ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider  
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for  
>>> Asterisk without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying  
>>> make FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chris.chen2004 at gmail.com  Mon Apr 12 09:08:30 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 12 Apr 2010 12:08:30 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
	<z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>
Message-ID: <z2y507898381004120908v8d6e9e63q63e342eb6f279e6a@mail.gmail.com>

Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable
"rport" under "Network" settings of Aastra phone,and restart the phone, most
likely you should be able to bet back to work unless your SMC Gateway really
sucks.
Please advise if this helps.

Chris


On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
> for the user directory. Only one time registering then lost the connection.
>
> Here is the sofia profile internal status
>
> Call-ID:        7307ef8fa6044407
> User:           202 at abc.com
> Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
> 192.168.1.51:5060>
> Agent:          Aastra 9143i/2.5.2.30
> Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
> Host:           TestSrv
> IP:             A.B.C.D
> Port:           5060
> Auth-User:      202
> Auth-Realm:     abc.com
> MWI-Account:    202 at abc.com
>
> Please let me know how to fix this issue.
>
> Thanks
> Lloyd
>
>
> On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> I dont think there is a lowest value, but 30 seconds is reasonable in most
>> cases.
>> You can also add a ping parameter with value 30, in the user config with
>> the variables.
>> The result is a SIP OPTIONS sent to the phone every X sec.
>> That's quite the equivalent of qualify=yes in Asterisk.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>>
>> what is the lowest value I can use for sip-force-expires ?
>>
>> Thanks
>> Lloyd
>>
>>
>> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>>
>>>
>>> Yes Asterisk config using the qualify=yes.
>>>
>>> I will modify the Registration time out by 50sec and see how it is
>>> behaving.
>>>
>>> Thanks
>>> Lloyd
>>>
>>>
>>>
>>>
>>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>>
>>>> On most low-end routers, the NAT table will expire UDP translations
>>>> after 60 sec.
>>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>>> with X < 60 ?
>>>> You can also use sip-force-expires on the FS side.
>>>>
>>>> In your Asterisk config, do you use qualify=yes ?
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>>
>>>> Hi All,
>>>>
>>>> I am having lots of problem with NAT and FreeSWITCH.
>>>>
>>>> FreeeSWITCH running Public IP.
>>>>
>>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>>> Router/Modem : SMC Gateway.
>>>>
>>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>>> without any problem.
>>>>
>>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>>> FreeSWITCH work.
>>>>
>>>> How to solve this problem.
>>>>
>>>> Thanks,
>>>> Lloyd
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From rupa at rupa.com  Mon Apr 12 09:12:23 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 12 Apr 2010 11:12:23 -0500
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
In-Reply-To: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>
References: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>
Message-ID: <m2zb39273531004120912m73af9c41u63f07e6634ac0cdd@mail.gmail.com>

user_data,<user>@<domain> [var|param|attr] <name>,find user
data,mod_commands

http://wiki.freeswitch.org/wiki/Mod_commands#user_data

On Fri, Apr 9, 2010 at 4:02 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I have the following variables setup for a Extension 201
>
> <variable name="followme" value="yes"/>
> <variable name="followme_number" value="14161235555"/>
>
> How to access these variables when another Extension say 202 calling to 201
> or when an external call transfer to extension 201.
>
> Thanks.
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From lloyd.aloysius at gmail.com  Mon Apr 12 09:18:01 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 12 Apr 2010 12:18:01 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <z2y507898381004120908v8d6e9e63q63e342eb6f279e6a@mail.gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
	<z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>
	<z2y507898381004120908v8d6e9e63q63e342eb6f279e6a@mail.gmail.com>
Message-ID: <w2i8a19bf2e1004120918t77dc8ad4m9d54538cebfe2c52@mail.gmail.com>

David & Chris , thank you for the suggestions.

I think better replace SMC.

Lloyd



On Mon, Apr 12, 2010 at 12:08 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:

> Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable
> "rport" under "Network" settings of Aastra phone,and restart the phone, most
> likely you should be able to bet back to work unless your SMC Gateway really
> sucks.
> Please advise if this helps.
>
> Chris
>
>
>
> On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> I could not find a SIP ALG Setting. I setup the sip-force-expires and
>> ping for the user directory. Only one time registering then lost the
>> connection.
>>
>> Here is the sofia profile internal status
>>
>> Call-ID:        7307ef8fa6044407
>> User:           202 at abc.com
>> Contact:        "Mike Derouin" <sip:202 at A.B.C.D
>> :5060;transport=udp;srcadr=192.168.1.51:5060>
>> Agent:          Aastra 9143i/2.5.2.30
>> Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
>> Host:           TestSrv
>> IP:             A.B.C.D
>> Port:           5060
>> Auth-User:      202
>> Auth-Realm:     abc.com
>> MWI-Account:    202 at abc.com
>>
>> Please let me know how to fix this issue.
>>
>> Thanks
>> Lloyd
>>
>>
>> On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> I dont think there is a lowest value, but 30 seconds is reasonable in
>>> most cases.
>>> You can also add a ping parameter with value 30, in the user config with
>>> the variables.
>>> The result is a SIP OPTIONS sent to the phone every X sec.
>>> That's quite the equivalent of qualify=yes in Asterisk.
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>>>
>>> what is the lowest value I can use for sip-force-expires ?
>>>
>>> Thanks
>>> Lloyd
>>>
>>>
>>> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>>> > wrote:
>>>
>>>>
>>>> Yes Asterisk config using the qualify=yes.
>>>>
>>>> I will modify the Registration time out by 50sec and see how it is
>>>> behaving.
>>>>
>>>> Thanks
>>>> Lloyd
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>>>
>>>>> On most low-end routers, the NAT table will expire UDP translations
>>>>> after 60 sec.
>>>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>>>> with X < 60 ?
>>>>> You can also use sip-force-expires on the FS side.
>>>>>
>>>>> In your Asterisk config, do you use qualify=yes ?
>>>>>
>>>>>  David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>>>
>>>>> Hi All,
>>>>>
>>>>> I am having lots of problem with NAT and FreeSWITCH.
>>>>>
>>>>> FreeeSWITCH running Public IP.
>>>>>
>>>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>>>> Router/Modem : SMC Gateway.
>>>>>
>>>>> The Phone Losing the Registrations. The Environment working for
>>>>> Asterisk without any problem.
>>>>>
>>>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>>>> FreeSWITCH work.
>>>>>
>>>>> How to solve this problem.
>>>>>
>>>>> Thanks,
>>>>> Lloyd
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 vhatz at kinetix.gr  Mon Apr 12 09:22:13 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Mon, 12 Apr 2010 19:22:13 +0300
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
Message-ID: <4BC348B5.8060805@kinetix.gr>

Hello Alfredo,

Please see comments inline:

On 10/4/10 6:51 ??, Alfredo Quiroga-Villamil wrote:
> All:
>
> A while back I tried to solve a false answer supervision issue I was
> intermittently receiving from underlying carriers. Back then I tried
> to find a solution using asterisk but had other pending things and put
> this off until now.
>
> Does anyone have any recommendations on how to possibly handle or get
> around FAS using FS. If I am not mistaken what would be needed is to
> have something that upon receiving the first 200 message, it simply
> ignores it, never propagating it and waits for the next 200. I can
> control this now a little bit better since it's only happening when
> the calls are sent to a couple of GrandStreams (FXO).
>    
As you have already noticed, when you get FAS, you receive a 200 before 
the call is answered. But I have never seen a second 200 when the call 
is really answered by the remote end, because after you receive the 
first (and only) 200 message, the call is considered as being properly 
connected, with RTP going both ways and all. A FAS giving carrier will 
take care to not send you another 200 when the call is answered.

So, I don't believe there is a way to detect and remove FAS 
successfully, because to your equipment, it just looks like a properly 
answered call. What you can do is gather batches of CDRs and extract 
statistics about the time delay from the INVITE you send to the 200 you 
receive. If you measure it too short, then you PROBABLY have FAS, and 
you'd still need to place a test call yourself you verify it.

You will not be able of course to tell which exact calls had FAS, but 
you'll be able to have a clue about whether there is FAS present on the 
route or not. And this means that you'd still have to do a lot of work 
yourself, like watching stats, placing test calls etc.

Even if you were able to find a way to remove FAS on a route, that would 
mean that your termination carrier would still charge you for the time 
difference between the 200 that he sends you and the BYE. And you'd 
still pay for a longer duration than what you would charge to your 
customer if you were able to remove FAS.

This is why my opinion is to watch statistics, place test calls and in 
the event of a carrier giving FAS, remove it from your routing. Sooner 
or later, they will either stop giving you FAS or you will stop doing 
business with them.


Best regards & good luck,
Vlasis Hatzistavrou.


From steveayre at gmail.com  Mon Apr 12 09:24:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 12 Apr 2010 17:24:59 +0100
Subject: [Freeswitch-users] FS pid
In-Reply-To: <ADA82617-5F5A-4FD3-A2A5-3D6F26E98BFA@gmail.com>
References: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>
	<ADA82617-5F5A-4FD3-A2A5-3D6F26E98BFA@gmail.com>
Message-ID: <z2leafa772f1004120924k20b0e121ra3ddbf53abec00f@mail.gmail.com>

Actually that won't be quite enough as the PID file is created by
FreeSWITCH itself when it starts.

The PID file's is placed in the run directory. You can move the PID
file by changing the run directory. If you want to put it in a
different directory, that would be possible but only by rewriting part
of the code.

-Steve


2010/4/12 David Ponzone <david.ponzone at gmail.com>:
> Look in /etc/init.d/freeswitch.
> pidfile is generally generated during startup.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 12/04/2010 ? 05:38, Madovsky a ?crit :
>
> Hi all,
>
> how can I change the name and the path of the pid file ?
>
> Thanks
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From srinivas.ksvreddy at gmail.com  Mon Apr 12 09:25:45 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 12 Apr 2010 21:55:45 +0530
Subject: [Freeswitch-users] freeswitch build errors
Message-ID: <v2wf8af5741004120925g74038c05pf2975efe0a1c1837@mail.gmail.com>

HI,

i have customized freeswitch1.0.2 for dynamcially adding users(1000.xml) and
reloading, so now i cant move for 1.0.6 versoin, now freeswitch1.0.2 giving
some compile errors in libsofia_sip_ua_static project.
any idea,

thanks in advacne

-- 
Srinivasula Reddy K
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brian at freeswitch.org  Mon Apr 12 09:30:10 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Apr 2010 11:30:10 -0500
Subject: [Freeswitch-users] freeswitch build errors
In-Reply-To: <v2wf8af5741004120925g74038c05pf2975efe0a1c1837@mail.gmail.com>
References: <v2wf8af5741004120925g74038c05pf2975efe0a1c1837@mail.gmail.com>
Message-ID: <04F6C157-6A3C-49F4-8383-4923D1A972F0@freeswitch.org>

Have you heard of this thing called XML_CURL that does exactly that and it already came with 1.0.2 and is even more powerful in 1.0.6.

I'm guessing it was your misunderstanding of how to use FreeSWITCH that lead to your patching the source in this way.  BTW your 1.0.2 shouldn't be used we fixed a few security concerns since then.

/b

On Apr 12, 2010, at 11:25 AM, srinivasula reddy wrote:

> HI,
>  
> i have customized freeswitch1.0.2 for dynamcially adding users(1000.xml) and reloading, so now i cant move for 1.0.6 versoin, now freeswitch1.0.2 giving some compile errors in libsofia_sip_ua_static project.
> any idea,
>  
> thanks in advacne
> 
> -- 
> Srinivasula Reddy K



From david.ponzone at gmail.com  Mon Apr 12 09:35:25 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 12 Apr 2010 18:35:25 +0200
Subject: [Freeswitch-users] FS pid
In-Reply-To: <z2leafa772f1004120924k20b0e121ra3ddbf53abec00f@mail.gmail.com>
References: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>
	<ADA82617-5F5A-4FD3-A2A5-3D6F26E98BFA@gmail.com>
	<z2leafa772f1004120924k20b0e121ra3ddbf53abec00f@mail.gmail.com>
Message-ID: <CD8AF267-DB2A-46EB-BC41-C56635E28F0A@gmail.com>

Steve,

you're right.
I got confused by some unnecessary config lines in the Debian-style  
init file provided in the source tree.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 12/04/2010 ? 18:24, Steven Ayre a ?crit :

> Actually that won't be quite enough as the PID file is created by
> FreeSWITCH itself when it starts.
>
> The PID file's is placed in the run directory. You can move the PID
> file by changing the run directory. If you want to put it in a
> different directory, that would be possible but only by rewriting part
> of the code.
>
> -Steve
>
>
> 2010/4/12 David Ponzone <david.ponzone at gmail.com>:
>> Look in /etc/init.d/freeswitch.
>> pidfile is generally generated during startup.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 12/04/2010 ? 05:38, Madovsky a ?crit :
>>
>> Hi all,
>>
>> how can I change the name and the path of the pid file ?
>>
>> Thanks
>> Franck
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Apr 11 17:39:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 20:39:19 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
Message-ID: <74497DDA-D84B-4DC1-B736-7AF1FC938782@jerris.com>

Why would we stop and restart from the beginning?


On Apr 7, 2010, at 2:03 PM, CHU, XINGJUN (XINGJUN) wrote:

> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun



From brian at freeswitch.org  Mon Apr 12 09:39:07 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Apr 2010 11:39:07 -0500
Subject: [Freeswitch-users] FS pid
In-Reply-To: <CD8AF267-DB2A-46EB-BC41-C56635E28F0A@gmail.com>
References: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>
	<ADA82617-5F5A-4FD3-A2A5-3D6F26E98BFA@gmail.com>
	<z2leafa772f1004120924k20b0e121ra3ddbf53abec00f@mail.gmail.com>
	<CD8AF267-DB2A-46EB-BC41-C56635E28F0A@gmail.com>
Message-ID: <8FD9337D-7AEA-49D6-BFB1-85F2AFF7FA8A@freeswitch.org>

--with-rundir on configure will let you dictate where it puts the pid file.

/b

On Apr 12, 2010, at 11:35 AM, David Ponzone wrote:

> Steve,
> 
> you're right.
> I got confused by some unnecessary config lines in the Debian-style init file provided in the source tree.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 

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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From garrison at codefix.net  Mon Apr 12 09:42:35 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 12:42:35 -0400
Subject: [Freeswitch-users] Dial Group & Handy-Tone 386
Message-ID: <4BC34D7B.7000606@codefix.net>

I'm trying to set up a dial group using an IP phone (Grandstream GXP 
2020) and a phone via Handy-Tone 386, but there seems to be an issue 
with the Handy-tone.

Routing a DID directly to either extension works without a hitch, and 
both phones ring using the following dial group:

     <extension name="group_dial_test">
       <condition field="destination_number" expression="^2000$">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="bridge" data="group/test@${domain_name}"/>
       </condition>
     </extension>


Incidentally, I've also seen the following syntax, is there a difference?

<action application="bridge" data="${group_call(test@${domain_name}+A)}"/>

The only problem is that when the call is answered by the phone 
connected to the Handy-Tone FXS port, neither end has audio. The GXP 
2020 works in all cases.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From garrison at codefix.net  Mon Apr 12 09:54:05 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 12:54:05 -0400
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
Message-ID: <4BC3502D.4020800@codefix.net>

Jonas Gauffin wrote:

> Which phone models are you using?

> I'm looking for a budget phone and a standard phone. 
> Both should work well behind NAT (FS <--> internet <--> router <--> phones).

I'm really happy with a Grandstream 2020, but I've also had luck with a 
Handy-Tone 386 with two FXS ports; both devices support STUN but since 
my FS is on the local network that hasn't been an issue.

It really doesn't get any more budget than a cheap FXS adapter like the 
Handy-Tone series and an old phone, but they do have issues. 
Voip-info.org has some good information on known issues.

(also see my other post)

-gh


From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 09:58:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Apr 2010 11:58:07 -0500
Subject: [Freeswitch-users] performance comparison between centos and
	freebsd
In-Reply-To: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>
References: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>
Message-ID: <w2l191c3a031004120958h34cf45dfiafa9acf29f2e4f1f@mail.gmail.com>

Either this will help:

http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Threading_Library

or nothing will.

FreeBSD is always a pain to support because it's like a spoiled actor in a
bad movie.
It makes many demands on everyone, it should be called DivaOS.  Still we try
to support it and if it does not
work as well we really can't do much about it.



On Sat, Apr 10, 2010 at 9:14 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.
>
> The expectation is that freeswitch 1.0.5 acting as media proxy would
> perform better in freebsd, but I found that freebsd can only sustain
> half of the total concurrent calls as in centos 5.4 (120 vs 60).
>
> The test is run on both ATOM CPU and VIA c7 and the result is
> relatively the same.
>
> Does anyone know why?  Is this some sort of setting issues in freebsd
> kernel?  I have tried with pure freebsd and pfsense and the result is
> the same.
>
>
>
>
> 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
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 09:59:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Apr 2010 11:59:18 -0500
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <4BC3502D.4020800@codefix.net>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<4BC3502D.4020800@codefix.net>
Message-ID: <4555CE92-2832-4FA4-8660-039B7EFFDC1C@freeswitch.org>


On Apr 12, 2010, at 11:54 AM, Garrison Hoffman wrote:

> I'm really happy with a Grandstream 2020, but I've also had luck with a 
> Handy-Tone 386 with two FXS ports; both devices support STUN but since 
> my FS is on the local network that hasn't been an issue.

Really?  How can anyone be happy with those phones... they don't even make good paperweights.  :P

> It really doesn't get any more budget than a cheap FXS adapter like the 
> Handy-Tone series and an old phone, but they do have issues. 
> Voip-info.org has some good information on known issues.

My test for a good phone is take the handset and walk away from desk with it.  If phone follows with very little effort its a crap phone.  If it stays put for the most part its a good phone.

> 
> -gh

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From sean at obscuradigital.com  Mon Apr 12 09:59:14 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 12 Apr 2010 09:59:14 -0700
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <g2r8a19bf2e1004120900t758f1a04yfb7d0e3b02271b42@mail.gmail.com>
Message-ID: <C7E89F72.80A1%sean@obscuradigital.com>

Hey Lloyd,

When you say download sound files, where from?  And how is that defined in
the ivr menu?

I looked over the wiki but couldn?t figure out the flow.

Plus I keep getting this message after enter the digit 5

2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
I see you have a Sonus!
FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
Sadly, my creator had to spend several hours figuring this out so I thought
you'd like to know that!
Don't worry, DTMF will work but you may want to ask them to fix it......

Thanks
Sean


On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:

> Here is the syntax I am using inside the ivr menus
> 
> ?<entry action="menu-exec-app" digits="5" param="directory default
> ${domain_name} "/>
> 
> When the user press 5 .. the Directory Application Will start.
> ---
> 
> Here is the wiki.?
> 
> http://wiki.freeswitch.org/wiki/Mod_directory
> 
> You need to download sound files manually.The users returned MUST be a member
> of a group.
> 
> Thanks
> Lloyd
> 
> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello,
>> 
>> Trying to figure out how to use the mod directory app within an ivr menu.
>> ?Not sure of the syntax to use when building the ivr. ?Not ever sure it can
>> be done.
>> 
>> Also does mod_directory need to be added to a dialplan?
>> 
>> Thanks
>> Sean 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Mon Apr 12 10:13:14 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 12 Apr 2010 13:13:14 -0400
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <C7E89F72.80A1%sean@obscuradigital.com>
References: <g2r8a19bf2e1004120900t758f1a04yfb7d0e3b02271b42@mail.gmail.com>
	<C7E89F72.80A1%sean@obscuradigital.com>
Message-ID: <i2h8a19bf2e1004121013h7dddc794pb2c20b058f3f2ba1@mail.gmail.com>

You do not need to defined sound files in ivr. Sounds files are used by mod
directory.

 Please read the wiki, in the last paragraph.



*Please note as of 2010/02/24, the default en/us/callie sounds do not
contain the sounds needed for the directory. The current work-around is to
copy**http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
* <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/>* to
{sounds_dir}/en/us/callie/directory/48000*




On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hey Lloyd,
>
> When you say download sound files, where from?  And how is that defined in
> the ivr menu?
>
> I looked over the wiki but couldn?t figure out the flow.
>
> Plus I keep getting this message after enter the digit 5
>
> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I thought
> you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix it......
>
> Thanks
> Sean
>
>
>
> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
>
> Here is the syntax I am using inside the ivr menus
>
>  <entry action="menu-exec-app" digits="5" param="directory default
> ${domain_name} "/>
>
> When the user press 5 .. the Directory Application Will start.
> ---
>
> Here is the wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_directory
>
> You need to download sound files manually.The users returned MUST be a
> member of a group.
>
> Thanks
> Lloyd
>
> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Hello,
>
> Trying to figure out how to use the mod directory app within an ivr menu.
>  Not sure of the syntax to use when building the ivr.  Not ever sure it can
> be done.
>
> Also does mod_directory need to be added to a dialplan?
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From excelsio at gmx.net  Mon Apr 12 10:16:44 2010
From: excelsio at gmx.net (excelsio at gmx.net)
Date: Mon, 12 Apr 2010 19:16:44 +0200
Subject: [Freeswitch-users] (dynamic) user created conference rooms
Message-ID: <20100412171644.52330@gmx.net>

Hi,

I successfully installed the basic conference servicee from the conference script on the wiki:
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR

I don?t want to let my users tell me to create the conference rooms by hand, so I?d like to have a basic authentication system to let them create/delete their conference rooms. The question is, how? Well, I have a centralized user database. Therefore I could authenticate them with the help of RADIUS. But setting up a secured homepage to access the database? To complicated for me. Never done this. So I asked me whether the following would work:

1. users write an email to conference-setup at mydomain.com. They user their corporate email address and the corporate email server.
2. a script on the freeswitch host downloads the email and checks the header of the email: the sender?s address, i.e. user at mydomain.com and the used email server, i.e. email.mydomain.com
(Will this be an adequate spoofing check? The email will be checked before against spam by the email server...). So the users are "authenticated" somehow at least.
3. a script checks the subject for e.g. "ADD", "CHANGE" or "DELETE"
4. the body should contain the "CONFERENCE ROOM NUMBER" in the first line, a "PASSWORD" in the second line, the "CONFERENCE ROOM NAME" in the third line or in c ase a user wants to delete an entry, the old password.
5. another script will take those entries, starts mysql and insert those values in the table of the database

What do you think? Will this work? Has anyone already tried that? Are there easier options? It would be great if you could give some hints.

thanks in advance

Michael
-- 
GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01


From yehavi.bourvine at gmail.com  Mon Apr 12 10:19:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 12 Apr 2010 20:19:26 +0300
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
Message-ID: <w2kd361ca7b1004121019obf5f9451n58efeb49827a545f@mail.gmail.com>

The law of "what you pay is what you get" works here. You want a good phone?
Then pay for a SNOM or Polycom.

                       Regards, __Yehavi:

 2010/4/12 Jonas Gauffin <jonas.gauffin at gmail.com>

> Hello,
>
> Which phone models are you using?
>
> I'm looking for a budget phone and a standard phone.
> Both should work well behind NAT (FS <--> internet <--> router <-->
> phones).
>
> Regards,
>   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
>
>
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From msc at freeswitch.org  Mon Apr 12 10:20:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 10:20:07 -0700
Subject: [Freeswitch-users] Getting git updates
In-Reply-To: <011f01cad82c$420847c0$c618d740$@com>
References: <011f01cad82c$420847c0$c618d740$@com>
Message-ID: <s2j87f2f3b91004121020ycaac9956gfef01975c579bcb2@mail.gmail.com>

You should still use "make current" - it works great on git-based FS repos.
As for the core files, they are really only useful if you have exactly the
same binary freeswitch file that created them. If you are not experiencing
any seg faults then go ahead and delete them. If you are experiencing seg
faults after updating to latest release then definitely open a JIRA ticket.

-MC

On Fri, Apr 9, 2010 at 2:33 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  Once we pull down the full FreeSWITCH version using git, should we
> continue to use ?make current? to update thereafter?
>
>
>
> I started with this command and interrupted it just after it had pulled the
> source changes. I was surprised to see that ?fs_cli? no longer worked. I
> went to freeswitch/bin and it was gone. Are the contents of this directory
> removed in ?make current??
>
>
>
> I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
> to remove these? They are about 200M each.
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From garrison at codefix.net  Mon Apr 12 10:21:08 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 13:21:08 -0400
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <4555CE92-2832-4FA4-8660-039B7EFFDC1C@freeswitch.org>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<4BC3502D.4020800@codefix.net>
	<4555CE92-2832-4FA4-8660-039B7EFFDC1C@freeswitch.org>
Message-ID: <4BC35684.50905@codefix.net>

Brian West wrote:

> Really?  How can anyone be happy with those phones... they don't even 
> make good paperweights.  :P

> My test for a good phone is take the handset and walk away from desk 
> with it.  If phone follows with very little effort its a crap phone.  If 
> it stays put for the most part its a good phone.

I'd prefer not to disclose details, but the GXP 2020 has actually 
survived a high impact collision with a wall, that seems as good a test 
as any. I will concede, though, it's a bit lightweight. If you think you 
may one day be forced to carry your IP phone long distances, this is the 
one for you!

-gh


From lawwton at gmail.com  Mon Apr 12 10:23:34 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Mon, 12 Apr 2010 13:23:34 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <4BC348B5.8060805@kinetix.gr>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
	<4BC348B5.8060805@kinetix.gr>
Message-ID: <o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>

Thanks for the response Vlasis. We run statistical analysis for FAS in
our production routes.

This request would be for a controlled environment where I am using
GrandStreams 488 to convert from Voip to PSTN. I was thinking that
perhaps FS has a way to allow me somehow access to the Signaling
portion of the protocol. If that was the case, I would be able to hack
my way around it and perhaps ignore the first 200 I receive and not
propagate it to the origination gateway. This is the setup I am
looking at:

A => B => GrandStream Device => PSTN

   SIP   SIP

A/B - Both Gateways

This is a test environment I am using, something I am just playing
with. So in every case I will receive a FAS from the GrandStream
device.

Do you know if there is a way to launch an application from the
DialPlan and have access to intercept Signaling; something that would
allow me to see the messages coming in and make decisions based on
them like propagate or do not propagate that message. I know this is a
tough thing since I am literally asking for access to the SIP Stack.

Thanks,

Alfredo

On Mon, Apr 12, 2010 at 12:22 PM, Vlasis Hatzistavrou (KTI)
<vhatz at kinetix.gr> wrote:
> Hello Alfredo,
>
> Please see comments inline:
>
> On 10/4/10 6:51 ??, Alfredo Quiroga-Villamil wrote:
>> All:
>>
>> A while back I tried to solve a false answer supervision issue I was
>> intermittently receiving from underlying carriers. Back then I tried
>> to find a solution using asterisk but had other pending things and put
>> this off until now.
>>
>> Does anyone have any recommendations on how to possibly handle or get
>> around FAS using FS. If I am not mistaken what would be needed is to
>> have something that upon receiving the first 200 message, it simply
>> ignores it, never propagating it and waits for the next 200. I can
>> control this now a little bit better since it's only happening when
>> the calls are sent to a couple of GrandStreams (FXO).
>>
> As you have already noticed, when you get FAS, you receive a 200 before
> the call is answered. But I have never seen a second 200 when the call
> is really answered by the remote end, because after you receive the
> first (and only) 200 message, the call is considered as being properly
> connected, with RTP going both ways and all. A FAS giving carrier will
> take care to not send you another 200 when the call is answered.
>
> So, I don't believe there is a way to detect and remove FAS
> successfully, because to your equipment, it just looks like a properly
> answered call. What you can do is gather batches of CDRs and extract
> statistics about the time delay from the INVITE you send to the 200 you
> receive. If you measure it too short, then you PROBABLY have FAS, and
> you'd still need to place a test call yourself you verify it.
>
> You will not be able of course to tell which exact calls had FAS, but
> you'll be able to have a clue about whether there is FAS present on the
> route or not. And this means that you'd still have to do a lot of work
> yourself, like watching stats, placing test calls etc.
>
> Even if you were able to find a way to remove FAS on a route, that would
> mean that your termination carrier would still charge you for the time
> difference between the 200 that he sends you and the BYE. And you'd
> still pay for a longer duration than what you would charge to your
> customer if you were able to remove FAS.
>
> This is why my opinion is to watch statistics, place test calls and in
> the event of a carrier giving FAS, remove it from your routing. Sooner
> or later, they will either stop giving you FAS or you will stop doing
> business with them.
>
>
> Best regards & good luck,
> Vlasis Hatzistavrou.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 12 10:27:03 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Apr 2010 12:27:03 -0500
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <4BC35684.50905@codefix.net>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<4BC3502D.4020800@codefix.net>
	<4555CE92-2832-4FA4-8660-039B7EFFDC1C@freeswitch.org>
	<4BC35684.50905@codefix.net>
Message-ID: <83A5B040-8CF5-44AF-A309-A230E4672E23@freeswitch.org>

My GXP2020 did not survive the impact of ten 9 mm gold dot hollow point rounds in my lab.  But it was stress relief at its finest.

/b
PS: The lab was really the gun range.

On Apr 12, 2010, at 12:21 PM, Garrison Hoffman wrote:

> I'd prefer not to disclose details, but the GXP 2020 has actually 
> survived a high impact collision with a wall, that seems as good a test 
> as any. I will concede, though, it's a bit lightweight. If you think you 
> may one day be forced to carry your IP phone long distances, this is the 
> one for you!



From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From garrison at codefix.net  Mon Apr 12 10:31:48 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 13:31:48 -0400
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <w2kd361ca7b1004121019obf5f9451n58efeb49827a545f@mail.gmail.com>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<w2kd361ca7b1004121019obf5f9451n58efeb49827a545f@mail.gmail.com>
Message-ID: <4BC35904.30604@codefix.net>

Yehavi Bourvine wrote:
> The law of "what you pay is what you get" works here. You want a good 
> phone? Then pay for a SNOM or Polycom.

Therefore Windows is better than Linux and SCO Unix is better than BSD? 
  I think that law was repealed shortly after the Prohibition Era.


From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From freeswitch-users-list at metik.com  Mon Apr 12 10:40:45 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Mon, 12 Apr 2010 13:40:45 -0400
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
Message-ID: <4BC35B1D.1030403@metik.com>

I would recommend Cisco Linksys and Snom, the quirks they do have are 
less painful and mostly manageable.  However, if you are looking for a 
quality speaker phone, I suggest Polycom or Cisco.   I'm not a fanboy of 
any manufacturer and even though I like the sound quality of most 
Polycom phones and their presence capabilities, I have seen far too many 
of them fail in the field (i.e. displays and memory).

That does not mean that others have not seen the same with the phones 
that I recommend, but a side from a few caveats, they have been largely 
reliable if you avoid firmware roulette. 

My recommendation on ATAs would be either to avoid them or limit their 
use to connect cordless phones or fax machines.  I have deployed a 
significant number (thousands) of the Linksys SPA 2102 units and 
although I have seen many bugs come and go, they provide a healthy 
balance of quality and economy.  If the budget and port density supports 
it, I highly recommend Audiocodes (especially when it comes to faxing).

-metik


Jonas Gauffin wrote:
> Hello,
>
> Which phone models are you using?
>
> I'm looking for a budget phone and a standard phone. 
> Both should work well behind NAT (FS <--> internet <--> router <--> 
> phones).
>
> Regards,
>   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
>   



From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From garrison at codefix.net  Mon Apr 12 11:08:58 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 14:08:58 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
Message-ID: <4BC361BA.504@codefix.net>

Using git://git.freeswitch.org/freeswitch.git
commit e7ff9f8506dbbc57538e562b277f216bc31ecc93
Date:   Fri Apr 9 16:03:54 2010 -0400

[WARNING] switch_ivr_menu.c:704 Invalid Action [menu-say-text]

Indeed, switch_ivr_action_t has no action "menu-say-text"

Has this been removed or is it not yet supported? Using flite in 
greet-long/greet-short works fine, is there any way to use TTS in menu 
actions?


Thanks,

-gh


From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 noisewaterphd at gmail.com  Mon Apr 12 11:11:32 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Mon, 12 Apr 2010 12:11:32 -0600
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
	<m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>
Message-ID: <n2j1c9955401004121111kdccb831ep53511c6bc19b2006@mail.gmail.com>

Hi,

We are wanting to set up a new support department, all of the agents will
register to Freeswitch (SIP). We will use Freeswitch for IVR, and some
custom integration into some systems here.

All calls will come in, and go out through the Mitel (T1/E1 cards).

I should have a SIP Trunk license for the Mitel this week, after which I
might have some more specific issues, but for now I was just putting out a
feeler to make sure there weren't any known issues connecting to the Mitel
stuff.

I've had a few private replies, and drummed up some stuff on IRC, and it
sounds like this should work out quite smoothly.

Thanks,

Kenny

On Fri, Apr 9, 2010 at 5:34 PM, Dan Le <dule.maillist at gmail.com> wrote:

> Can you be a bit more specific what you want to accomplish?
>
> We often trunk to a 3300 and I know many that have done so without issues;
> the configuration is quite straightforward. Gateway configs in FS, SIP Peer
> Profile (+ all the dependent forms) in the 3300 and you're pretty much good
> to go.
> Sorry I don't know what kind of information you're looking for in
> particular.
>
> Dan
>
>
> On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <
> noisewaterphd at gmail.com> wrote:
>
>> Hi,
>>
>> I've already read the interop list, but I'm wondering if anyone on here
>> has anymore experience/info on trunking freeswitch to a Mitel 3300?
>>
>> Specifically, I want to use freeswitch for acd and sip registrations, and
>> just use our mitel for switching to the PSTN.
>>
>> Does anyone have some good info to share?
>>
>> Thanks,
>>
>> Kenny
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 11:19:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Apr 2010 13:19:49 -0500
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
	<q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>
Message-ID: <k2h191c3a031004121119y78de8972h58c4995c1dd3d565@mail.gmail.com>

We debated this and concluded that in most cases in a server environment
that it would be unwise to attempt this floating license scheme when it
could clearly present scalability issues and most companies would not use
g729 unless they had a serious business case for it and it seemed to be an
overly-complex way to end up with dropped calls.


On Fri, Apr 9, 2010 at 4:36 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Can the licenses be transferred between computers?  If I have a cluster of
> 5 freeswitch servers do I need to assign a set of licenses to each
> specifically?
>
> I know that if you get a bunch of Howler's license they sit in a pool,
> attached to one master computer, and your other computers can all use that
> pool of licenses as long as the master server is running (or has been
> running in the last week.) But they are tied to the mac address of that
> master computer.
>
> Cheers,
> Fraser
>
>
>
>
> On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Greetings all,
>>
>> The FreeSWITCH team would like to let everyone know that we do indeed sell
>> g.729 licenses for $10 each. Use this link to initiate a purchase:
>> http://www.freeswitch.org/node/235
>>
>> Note: licenses are available only for Linux-based systems at this time.
>> Please stay tuned for updates.
>>
>> The INSTALL.txt file has very detailed instructions:
>> http://files.freeswitch.org/g729/INSTALL.txt
>>
>> Keep in mind that a single license includes one encoder and one decoder,
>> that is, it can transcode both directions of a single phone call. If you
>> have any other questions please email us here or join us in #freeswitch on
>> irc.freenode.net.
>>  -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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From anthony.minessale at gmail.com  Mon Apr 12 11:42:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Apr 2010 13:42:16 -0500
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
In-Reply-To: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>
References: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>
Message-ID: <n2u191c3a031004121142qa125ca1fn949c686432f0610b@mail.gmail.com>

That is unfortunate for you.

We do not support older revisions in this way and if you need commercial
support you can contact
consulting at freeswitch.org but be advised that there is a 1,000% increase in
pricing for out of tree forks such as yours.

I suggest you start over and do not make any modifications to the base
FreeSWITCH that you cannot share with the community.




On Fri, Apr 9, 2010 at 2:40 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> HI,
> we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
> cant go for 1.0.6, any there idea about compilation errors
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From bcxml at hotmail.com  Mon Apr 12 12:08:30 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Mon, 12 Apr 2010 15:08:30 -0400
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>,
	<20100406040606.78daf9fd@anubis.defcon1>,
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>,
	<1270823599661-4877398.post@n2.nabble.com>,
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <SNT135-w400266A630F3D3A657F526B7120@phx.gbl>


I am resubmitting this just in case it got lost in the pile

 

Here is the link

 

http://pastebin.freeswitch.org/12675

 

Thanks

 

 

Brian Campbell
 


Date: Fri, 9 Apr 2010 10:03:38 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with IVR app starting too soon

Can you pastebin a full debug log of this call?
-MC


On Fri, Apr 9, 2010 at 7:33 AM, Brian Campbell <bcxml at hotmail.com> wrote:



I tried what you mentioned but it did not seem to have any effect. I am
still losing part of the openning message

Here is my inbound dialplan..

<extension name="Inbound_1234567890">
 <condition field='destination_number' expression='1234567890'>
   <action application="playback" data="silence_stream://1000"/>
   <action application="bridge"
data="{bypass_media=true}sofia/internal/1234567890 at 127.0.0.1:5060;transport=tcp"/>
 </condition>
</extension>


Brian
--
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4877398.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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From tomb at cachecomm.com  Mon Apr 12 12:10:27 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 12 Apr 2010 13:10:27 -0600
Subject: [Freeswitch-users] {Scanned} wanpipe 3.6.0.13 openzap will not
	compile
Message-ID: <4BC37023.80300@cachecomm.com>

Freeswitch Gods,
I just bought a B601 Sangoma card. The Wanpipe install went will i get 
this from a hwprobe
-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=1 : HWEC=64 : V=03
2 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=2 : HWEC=64 : V=03

Card Cnt: B601=1 


but when i go to compile the openzap drivers i get the fallowing errors

ozmod_wanpipe_la-ozmod_wanpipe.lo -MD -MP -MF 
.deps/ozmod_wanpipe_la-ozmod_wanpipe.Tpo -c 
src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c  -fPIC -DPIC -o 
.libs/ozmod_wanpipe_la-ozmod_wanpipe.o
cc1: warnings being treated as errors
src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c: In function 'wanpipe_open':
src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:493: warning: implicit 
declaration of function 'sangoma_flush_event_bufs'
make: *** [ozmod_wanpipe_la-ozmod_wanpipe.lo] Error 1

I tried to remove the /usr/include/libsangoma.h 
that still stops on errors. I tried the 3.5 driver branch still the same 
thing.
what i am doing wrong

Tom


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Mon Apr 12 12:23:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 12:23:23 -0700
Subject: [Freeswitch-users] {Scanned} wanpipe 3.6.0.13 openzap will not
	compile
In-Reply-To: <4BC37023.80300@cachecomm.com>
References: <4BC37023.80300@cachecomm.com>
Message-ID: <p2l87f2f3b91004121223q2b642113v1d708c6a211bca06@mail.gmail.com>

Is this a fresh git (or svn) checkout of FreeSWITCH? I just did a make
current this morning on a system with oz & wanpipe. I used version 3.5.10
from Sangoma's site:

http://wiki.sangoma.com/wanpipe-linux-drivers#latest
Specifically: ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.5.10.tgz

Uninstall the wanpipe version you have now using the "./Setup uninstall"
method and then download & install 3.5.10 and try again.
-MC

On Mon, Apr 12, 2010 at 12:10 PM, Tom <tomb at cachecomm.com> wrote:

> Freeswitch Gods,
> I just bought a B601 Sangoma card. The Wanpipe install went will i get
> this from a hwprobe
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=1 : HWEC=64 : V=03
> 2 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=2 : HWEC=64 : V=03
>
> Card Cnt: B601=1
>
>
> but when i go to compile the openzap drivers i get the fallowing errors
>
> ozmod_wanpipe_la-ozmod_wanpipe.lo -MD -MP -MF
> .deps/ozmod_wanpipe_la-ozmod_wanpipe.Tpo -c
> src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c  -fPIC -DPIC -o
> .libs/ozmod_wanpipe_la-ozmod_wanpipe.o
> cc1: warnings being treated as errors
> src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c: In function 'wanpipe_open':
> src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:493: warning: implicit
> declaration of function 'sangoma_flush_event_bufs'
> make: *** [ozmod_wanpipe_la-ozmod_wanpipe.lo] Error 1
>
> I tried to remove the /usr/include/libsangoma.h
> that still stops on errors. I tried the 3.5 driver branch still the same
> thing.
> what i am doing wrong
>
> Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Mon Apr 12 12:34:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 12:34:28 -0700
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <4BC35904.30604@codefix.net>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<w2kd361ca7b1004121019obf5f9451n58efeb49827a545f@mail.gmail.com>
	<4BC35904.30604@codefix.net>
Message-ID: <j2g87f2f3b91004121234wa70c13q36098a48a8798de8@mail.gmail.com>

On Mon, Apr 12, 2010 at 10:31 AM, Garrison Hoffman <garrison at codefix.net>wrote:

> Yehavi Bourvine wrote:
> > The law of "what you pay is what you get" works here. You want a good
> > phone? Then pay for a SNOM or Polycom.
>
> Therefore Windows is better than Linux and SCO Unix is better than BSD?
>  I think that law was repealed shortly after the Prohibition Era.
>

Sort of. "Payment" comes in many forms, not just $$.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From kond at nstel.ru  Mon Apr 12 12:33:20 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 12 Apr 2010 23:33:20 +0400
Subject: [Freeswitch-users] call bargein (urgent call)
In-Reply-To: <20100409070730.C3CD31151B@mail.nstel.ru>
Message-ID: <20100412193319.E37C911492@mail.nstel.ru>

Hi all,
i'm still at a loss regarding incarnation of my idea (below).
Can anybody please point me to an analogous example?
Thanks and regards,
Nikolay.


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Kondratyev
Sent: Friday, April 09, 2010 11:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call bargein (urgent call)


Yes, that is exactly what i want.
Nikolay.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, April 09, 2010 10:48 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call bargein (urgent call)





On Thu, Apr 8, 2010 at 11:36 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:


Hi all,
 
i need a sort of  "call barge in" functianality, ar may be "urgent call"
will be better name... 
I know about existing "eavesdrop" application, but it does not suite me
because of two reasons: 
1. callee must be on active call, or eavesdrop will fail.
2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If
1003 hangs up, the call will terminate.
 
Existing "intercept" application also does not suite.
 
So i need the following: 
A dispatcher must be able to call any user urgently, that is if a user is
idle, it just should be a call, and when a user A is talking to user B, and
dispatcher makes urgent call to A, all three must occur in a conference.
I'm sure it's possible, but i'm rather new to FS and looks like i'm lost
while searching how to achive it....
 
Can anybody please advise how to do it?


So in other words, if the dispatcher calls A, and A's phone is idle it just
rings normally.
If the dispatcher calls A while A is connected to B, then you want to throw
A, B, and the dispatcher all into a conference?
Just confirming that I understand what is happening.
-MC



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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vfclists at googlemail.com  Mon Apr 12 05:22:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 12 Apr 2010 13:22:49 +0100
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
Message-ID: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>

Can Freeswitch be installed simply by zipping up the folder and unzipping it
to the destination?

Does it require some DLLs to be installed in the Windows system folder and
some registry entries as well?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From infos at madovsky.org  Sun Apr 11 20:38:15 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 11 Apr 2010 23:38:15 -0400
Subject: [Freeswitch-users] FS pid
Message-ID: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>

Hi all,

how can I change the name and the path of the pid file ?

Thanks

Franck
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 13:15:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 13:15:15 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <20100410143328.2c419080@anubis.defcon1>
References: <20100410143328.2c419080@anubis.defcon1>
Message-ID: <w2r87f2f3b91004121315r96ee8de2l330268e33ab05325@mail.gmail.com>

Update to latest and try again. Use top -H to see if a particular thread is
causing the issue and report back...
-MC

On Sat, Apr 10, 2010 at 5:33 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200)
> Started.
> ===============
>
> Hi list,
>
> since I left the old fashion bulding to use git (last version compiled
> on Debian lenny), FS is chewing 100% CPU just after launch.
>
> not any red line in console, nor registered device, nor anything in the log
> file.
>
> what could cause this behaviour?
>
> --
> I do not take drugs -- I am drugs.
>                -- Salvador Dali
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andrewkt at aktzero.com  Mon Apr 12 13:19:39 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Mon, 12 Apr 2010 16:19:39 -0400
Subject: [Freeswitch-users] sip uri incomming calls
In-Reply-To: <20100409194136.268140@gmx.net>
References: <20100409194136.268140@gmx.net>
Message-ID: <4BC3805B.5080901@aktzero.com>

On 4/9/2010 3:41 PM, thomas.ji at gmx.at wrote:
> hello list!
> can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
> what do i have to change in which files?
>    

For an existing extension YYY in dialplan/public/YYY.xml, you can send a 
call in as: YYY at server_ip_or_hostname:5080

-- 
Andrew Thompson



From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From tomb at cachecomm.com  Mon Apr 12 13:26:03 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 12 Apr 2010 14:26:03 -0600
Subject: [Freeswitch-users] {Scanned} Re: {Scanned} Re: {Scanned} wanpipe
 3.6.0.13 openzap will not	compile
In-Reply-To: <p2l87f2f3b91004121223q2b642113v1d708c6a211bca06@mail.gmail.com>
References: <4BC37023.80300@cachecomm.com>
	<p2l87f2f3b91004121223q2b642113v1d708c6a211bca06@mail.gmail.com>
Message-ID: <4BC381DB.70605@cachecomm.com>

Michael Collins wrote:
> Is this a fresh git (or svn) checkout of FreeSWITCH? I just did a make 
> current this morning on a system with oz & wanpipe. I used version 
> 3.5.10 from Sangoma's site:
>
> http://wiki.sangoma.com/wanpipe-linux-drivers#latest
> Specifically: 
> ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.5.10.tgz
>
> Uninstall the wanpipe version you have now using the "./Setup 
> uninstall" method and then download & install 3.5.10 and try again.
> -MC
>
> On Mon, Apr 12, 2010 at 12:10 PM, Tom <tomb at cachecomm.com 
> <mailto:tomb at cachecomm.com>> wrote:
>
>     Freeswitch Gods,
>     I just bought a B601 Sangoma card. The Wanpipe install went will i get
>     this from a hwprobe
>     -------------------------------
>     | Wanpipe Hardware Probe Info |
>     -------------------------------
>     1 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=1 : HWEC=64 : V=03
>     2 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=2 : HWEC=64 : V=03
>
>     Card Cnt: B601=1
>
>
>     but when i go to compile the openzap drivers i get the fallowing
>     errors
>
>     ozmod_wanpipe_la-ozmod_wanpipe.lo -MD -MP -MF
>     .deps/ozmod_wanpipe_la-ozmod_wanpipe.Tpo -c
>     src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c  -fPIC -DPIC -o
>     .libs/ozmod_wanpipe_la-ozmod_wanpipe.o
>     cc1: warnings being treated as errors
>     src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c: In function 'wanpipe_open':
>     src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:493: warning: implicit
>     declaration of function 'sangoma_flush_event_bufs'
>     make: *** [ozmod_wanpipe_la-ozmod_wanpipe.lo] Error 1
>
>     I tried to remove the /usr/include/libsangoma.h
>     that still stops on errors. I tried the 3.5 driver branch still
>     the same
>     thing.
>     what i am doing wrong
>
>     Tom
>
>     _______________________________________________
>     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
>   
MC,
i just got off the phone with Sangoma they said the the B601 card will 
only work with the 3.6 driver. Which will not work with openzap is that 
right.


Thanks

Tom


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From dyatsin at sangoma.com  Mon Apr 12 13:45:08 2010
From: dyatsin at sangoma.com (David Yat Sin)
Date: Mon, 12 Apr 2010 16:45:08 -0400
Subject: [Freeswitch-users] {Scanned} Re: {Scanned} Re: {Scanned} wanpipe
 3.6.0.13 openzap will not	compile
In-Reply-To: <4BC381DB.70605@cachecomm.com>
References: <4BC37023.80300@cachecomm.com>	<p2l87f2f3b91004121223q2b642113v1d708c6a211bca06@mail.gmail.com>
	<4BC381DB.70605@cachecomm.com>
Message-ID: <4BC38654.7020807@sangoma.com>

Hi Tom,
The B601 will not work with sangoma_prid because the B601 does not have 
a hardware HDLC framer implemented (which is required by sangoma_prid).

If you need a PRI interface, you can still use the B601 and configure 
the T1/E1 port in "zaptel mode" and use libpri instead.

If you need a RBS interface, the B601 will work fine.

Regards,

*David Yat Sin, **BEng**, **Software Engineer**
*Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON 
L3R 9T3 Canada
t. 1 905 474 1990 x 119 | e. _dyatsin at sangoma.com 
<mailto:dyatsin at sangoma.com>_


On 4/12/2010 4:26 PM, Tom wrote:
> Michael Collins wrote:
>    
>> Is this a fresh git (or svn) checkout of FreeSWITCH? I just did a make
>> current this morning on a system with oz&  wanpipe. I used version
>> 3.5.10 from Sangoma's site:
>>
>> http://wiki.sangoma.com/wanpipe-linux-drivers#latest
>> Specifically:
>> ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.5.10.tgz
>>
>> Uninstall the wanpipe version you have now using the "./Setup
>> uninstall" method and then download&  install 3.5.10 and try again.
>> -MC
>>
>> On Mon, Apr 12, 2010 at 12:10 PM, Tom<tomb at cachecomm.com
>> <mailto:tomb at cachecomm.com>>  wrote:
>>
>>      Freeswitch Gods,
>>      I just bought a B601 Sangoma card. The Wanpipe install went will i get
>>      this from a hwprobe
>>      -------------------------------
>>      | Wanpipe Hardware Probe Info |
>>      -------------------------------
>>      1 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=1 : HWEC=64 : V=03
>>      2 . AFT-B601-SH : SLOT=4 : BUS=3 : IRQ=90 : PORT=2 : HWEC=64 : V=03
>>
>>      Card Cnt: B601=1
>>
>>
>>      but when i go to compile the openzap drivers i get the fallowing
>>      errors
>>
>>      ozmod_wanpipe_la-ozmod_wanpipe.lo -MD -MP -MF
>>      .deps/ozmod_wanpipe_la-ozmod_wanpipe.Tpo -c
>>      src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c  -fPIC -DPIC -o
>>      .libs/ozmod_wanpipe_la-ozmod_wanpipe.o
>>      cc1: warnings being treated as errors
>>      src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c: In function 'wanpipe_open':
>>      src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:493: warning: implicit
>>      declaration of function 'sangoma_flush_event_bufs'
>>      make: *** [ozmod_wanpipe_la-ozmod_wanpipe.lo] Error 1
>>
>>      I tried to remove the /usr/include/libsangoma.h
>>      that still stops on errors. I tried the 3.5 driver branch still
>>      the same
>>      thing.
>>      what i am doing wrong
>>
>>      Tom
>>
>>      _______________________________________________
>>      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
>>
>>      
> MC,
> i just got off the phone with Sangoma they said the the B601 card will
> only work with the 3.6 driver. Which will not work with openzap is that
> right.
>
>
> Thanks
>
> Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Mon Apr 12 13:53:38 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 12 Apr 2010 13:53:38 -0700
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <i2h8a19bf2e1004121013h7dddc794pb2c20b058f3f2ba1@mail.gmail.com>
Message-ID: <C7E8D662.847B%sean@obscuradigital.com>

Ok I?ve downloaded the sounds and setup my ivr menu, but now it doesn?t
recognizes any incoming DTMF tones.  I?ve run a wireshark capture and know
that DTMF tones are successful when calling into a conference line, but
nothing on the ivr menu.

Thoughts?
Thanks
Sean


On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:

> You do not need to defined sound files in ivr. Sounds files are used by mod
> directory.
> 
> ?Please read the wiki, in the last paragraph.
> 
> 
> 
> Please note?as of 2010/02/24, the default en/us/callie sounds do not contain
> the sounds needed for the directory. The current work-around is to
> copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
> <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/> ?to
> {sounds_dir}/en/us/callie/directory/48000
> 
> 
> 
> 
> On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hey Lloyd,
>> 
>> When you say download sound files, where from? ?And how is that defined in
>> the ivr menu?
>> 
>> I looked over the wiki but couldn?t figure out the flow.
>> 
>> Plus I keep getting this message after enter the digit 5
>> 
>> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
>> I see you have a Sonus!
>> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
>> Sadly, my creator had to spend several hours figuring this out so I thought
>> you'd like to know that!
>> Don't worry, DTMF will work but you may want to ask them to fix it......
>> 
>> Thanks
>> Sean
>> 
>> 
>> 
>> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
>> <http://lloyd.aloysius at gmail.com> > wrote:
>> 
>>> Here is the syntax I am using inside the ivr menus
>>> 
>>> ?<entry action="menu-exec-app" digits="5" param="directory default
>>> ${domain_name} "/>
>>> 
>>> When the user press 5 .. the Directory Application Will start.
>>> ---
>>> 
>>> Here is the wiki.?
>>> 
>>> http://wiki.freeswitch.org/wiki/Mod_directory
>>> 
>>> You need to download sound files manually.The users returned MUST be a
>>> member of a group.
>>> 
>>> Thanks
>>> Lloyd
>>> 
>>> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
>>> <http://sean at obscuradigital.com> > wrote:
>>>> Hello,
>>>> 
>>>> Trying to figure out how to use the mod directory app within an ivr menu.
>>>> ?Not sure of the syntax to use when building the ivr. ?Not ever sure it can
>>>> be done.
>>>> 
>>>> Also does mod_directory need to be added to a dialplan?
>>>> 
>>>> Thanks
>>>> Sean 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From lloyd.aloysius at gmail.com  Mon Apr 12 13:59:28 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 12 Apr 2010 16:59:28 -0400
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <C7E8D662.847B%sean@obscuradigital.com>
References: <i2h8a19bf2e1004121013h7dddc794pb2c20b058f3f2ba1@mail.gmail.com>
	<C7E8D662.847B%sean@obscuradigital.com>
Message-ID: <s2k8a19bf2e1004121359id7645891mdb25abd55183cbe4@mail.gmail.com>

Could you post your dialplan here.


On Mon, Apr 12, 2010 at 4:53 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Ok I?ve downloaded the sounds and setup my ivr menu, but now it doesn?t
> recognizes any incoming DTMF tones.  I?ve run a wireshark capture and know
> that DTMF tones are successful when calling into a conference line, but
> nothing on the ivr menu.
>
> Thoughts?
> Thanks
> Sean
>
>
>
> On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
>
> You do not need to defined sound files in ivr. Sounds files are used by mod
> directory.
>
>  Please read the wiki, in the last paragraph.
>
>
>
> *Please note as of 2010/02/24, the default en/us/callie sounds do not
> contain the sounds needed for the directory. The current work-around is to
> copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
> * <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/>
> * to {sounds_dir}/en/us/callie/directory/48000
> *
>
>
>
>
> On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Hey Lloyd,
>
> When you say download sound files, where from?  And how is that defined in
> the ivr menu?
>
> I looked over the wiki but couldn?t figure out the flow.
>
> Plus I keep getting this message after enter the digit 5
>
> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I thought
> you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix it......
>
> Thanks
> Sean
>
>
>
> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com <
> http://lloyd.aloysius at gmail.com> > wrote:
>
> Here is the syntax I am using inside the ivr menus
>
>  <entry action="menu-exec-app" digits="5" param="directory default
> ${domain_name} "/>
>
> When the user press 5 .. the Directory Application Will start.
> ---
>
> Here is the wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_directory
>
> You need to download sound files manually.The users returned MUST be a
> member of a group.
>
> Thanks
> Lloyd
>
>
> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com <
> http://sean at obscuradigital.com> > wrote:
>
> Hello,
>
> Trying to figure out how to use the mod directory app within an ivr menu.
>  Not sure of the syntax to use when building the ivr.  Not ever sure it can
> be done.
>
> Also does mod_directory need to be added to a dialplan?
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From robert.hadley at teotech.com  Mon Apr 12 14:04:26 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 12 Apr 2010 14:04:26 -0700
Subject: [Freeswitch-users] FW:  Mod directory
Message-ID: <13C1415E7B544698847EF99C548CE92C@greyhawk.tonecommander.com>

Hi Sean,

 

I don't know about your DTMF issue but I also had to add a directory
extension to the dialplan to go with the change to the IVR

 

IVR:

   <entry action="menu-exec-app" digits="5" param="transfer directory XML
default"/>    <!-- Directory -->

 

dialplan/defaults.xml:

    <!-- dial by name directory extension -->

    <extension name="directory">

      <condition field="destination_number" expression="^directory$">

       <action application="answer"/>

       <action application="sleep" data="500"/>

       <action application="directory" data="default $${domain} default"/>

      </condition>

    </extension>

 

Regards,

Robert

 

  _____  

From: Sean Holt [mailto:sean at obscuradigital.com] 
Sent: Monday, April 12, 2010 1:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Mod directory

 

Ok I've downloaded the sounds and setup my ivr menu, but now it doesn't
recognizes any incoming DTMF tones.  I've run a wireshark capture and know
that DTMF tones are successful when calling into a conference line, but
nothing on the ivr menu.

Thoughts?
Thanks
Sean


On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:

You do not need to defined sound files in ivr. Sounds files are used by mod
directory.

 Please read the wiki, in the last paragraph.



Please note as of 2010/02/24, the default en/us/callie sounds do not contain
the sounds needed for the directory. The current work-around is to
copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
<http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/>  to
{sounds_dir}/en/us/callie/directory/48000




On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:

Hey Lloyd,

When you say download sound files, where from?  And how is that defined in
the ivr menu?

I looked over the wiki but couldn't figure out the flow.

Plus I keep getting this message after enter the digit 5

2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
I see you have a Sonus!
FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
Sadly, my creator had to spend several hours figuring this out so I thought
you'd like to know that!
Don't worry, DTMF will work but you may want to ask them to fix it......

Thanks
Sean



On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
<http://lloyd.aloysius at gmail.com> > wrote:

Here is the syntax I am using inside the ivr menus

 <entry action="menu-exec-app" digits="5" param="directory default
${domain_name} "/>

When the user press 5 .. the Directory Application Will start.
---

Here is the wiki. 

http://wiki.freeswitch.org/wiki/Mod_directory

You need to download sound files manually.The users returned MUST be a
member of a group.

Thanks
Lloyd

On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
<http://sean at obscuradigital.com> > wrote:

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean 

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

 

  _____  

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


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

 

  _____  

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

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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 sean at obscuradigital.com  Mon Apr 12 14:15:01 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 12 Apr 2010 14:15:01 -0700
Subject: [Freeswitch-users] Mod directory
In-Reply-To: <s2k8a19bf2e1004121359id7645891mdb25abd55183cbe4@mail.gmail.com>
Message-ID: <C7E8DB65.8481%sean@obscuradigital.com>

Ok here?s my ivr menu*****

  <configuration name="ivr.conf" description="IVR menus">
    <menus>
      <menu name="auto_attendant_1"
        greet-long="ivr/ivr-welcome_to_freeswitch.wav"
        greet-short="ivr/ivr-welcome_to_freeswitch.wav"
        timeout="10000"
        invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
        exit-sound="voicemail/vm-goodbye.wav"
        inter-digit-timeout="2000"
        max-failures="3"
        digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
      <entry action="menu-exec-app" digits="2" param="transfer 2013 XML
context_1"/>    <!-- FS echo -->
      <entry action="menu-exec-app" digits="3" param="transfer 9999 XML
default"/>    <!-- MOH -->
      <entry action="menu-sub" digits="4" param="demo_ivr_submenu"/>  <!--
demo sub menu -->
      <entry action="menu-exec-app" digits="5" param="transfer 1234*256
enum"/>    <!-- Screaming monkeys -->
      <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
param="transfer $1 XML default"/> <!-- dial ext & x-fer -->
      <entry action="menu-top" digits="9"/>    <!-- Repeat this menu -->
      <entry action="menu-exec-app" digits="7" param="directory default
context_1"/>
     </menu>
        <!-- Demo IVR, Sub Menu -->
    <menu name="demo_ivr_submenu"
        greet-long="phrase:demo_ivr_sub_menu"
        greet-short="phrase:demo_ivr_sub_menu_short"
        invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
        exit-sound="voicemail/vm-goodbye.wav"
        timeout="15000"
        max-failures="3">
      <entry action="menu-top" digits="*"/>
     </menu>
    </menus>
  </configuration>
</include>

And here?s my dialplan*****

 <extension name="main_number_39" continue="true">
      <condition field="destination_number" expression="^4157960191$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="ivr" data="auto_attendant_1"/>
      </condition>
    </extension>

Thanks
Sean


On 4/12/10 1:59 PM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:

> Could you post your dialplan here.
> 
> 
> On Mon, Apr 12, 2010 at 4:53 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Ok I?ve downloaded the sounds and setup my ivr menu, but now it doesn?t
>> recognizes any incoming DTMF tones. ?I?ve run a wireshark capture and know
>> that DTMF tones are successful when calling into a conference line, but
>> nothing on the ivr menu.
>> 
>> Thoughts?
>> Thanks
>> Sean
>> 
>> 
>> 
>> On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
>> <http://lloyd.aloysius at gmail.com> > wrote:
>> 
>>> You do not need to defined sound files in ivr. Sounds files are used by mod
>>> directory.
>>> 
>>> ?Please read the wiki, in the last paragraph.
>>> 
>>> 
>>> 
>>> Please note?as of 2010/02/24, the default en/us/callie sounds do not contain
>>> the sounds needed for the directory. The current work-around is to
>>> copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
>>> <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/>
>>> <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/> ?to
>>> {sounds_dir}/en/us/callie/directory/48000
>>> 
>>> 
>>> 
>>> 
>>> On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com
>>> <http://sean at obscuradigital.com> > wrote:
>>>> Hey Lloyd,
>>>> 
>>>> When you say download sound files, where from? ?And how is that defined in
>>>> the ivr menu?
>>>> 
>>>> I looked over the wiki but couldn?t figure out the flow.
>>>> 
>>>> Plus I keep getting this message after enter the digit 5
>>>> 
>>>> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
>>>> I see you have a Sonus!
>>>> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
>>>> Sadly, my creator had to spend several hours figuring this out so I thought
>>>> you'd like to know that!
>>>> Don't worry, DTMF will work but you may want to ask them to fix it......
>>>> 
>>>> Thanks
>>>> Sean
>>>> 
>>>> 
>>>> 
>>>> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
>>>> <http://lloyd.aloysius at gmail.com>  <http://lloyd.aloysius at gmail.com> >
>>>> wrote:
>>>> 
>>>>> Here is the syntax I am using inside the ivr menus
>>>>> 
>>>>> ?<entry action="menu-exec-app" digits="5" param="directory default
>>>>> ${domain_name} "/>
>>>>> 
>>>>> When the user press 5 .. the Directory Application Will start.
>>>>> ---
>>>>> 
>>>>> Here is the wiki.?
>>>>> 
>>>>> http://wiki.freeswitch.org/wiki/Mod_directory
>>>>> 
>>>>> You need to download sound files manually.The users returned MUST be a
>>>>> member of a group.
>>>>> 
>>>>> Thanks
>>>>> Lloyd
>>>>> 
>>>>> 
>>>>> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
>>>>> <http://sean at obscuradigital.com>  <http://sean at obscuradigital.com> >
>>>>> wrote:
>>>>>> Hello,
>>>>>> 
>>>>>> Trying to figure out how to use the mod directory app within an ivr menu.
>>>>>> ?Not sure of the syntax to use when building the ivr. ?Not ever sure it
>>>>>> can be done.
>>>>>> 
>>>>>> Also does mod_directory need to be added to a dialplan?
>>>>>> 
>>>>>> Thanks
>>>>>> Sean 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Mon Apr 12 14:30:21 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 12 Apr 2010 14:30:21 -0700
Subject: [Freeswitch-users] FW:  Mod directory
In-Reply-To: <13C1415E7B544698847EF99C548CE92C@greyhawk.tonecommander.com>
Message-ID: <C7E8DEFD.8485%sean@obscuradigital.com>

Ok well seems that my DTMF are working again, but after making the changes
Robert recommended I?m now getting this error message

Invalid Application directory

Maybe my confusion is in the default $${domain} default line.  Not sure what
default represents.

I feel I?ve almost got this figured out.  Any thoughts why I might get this
error?
Thanks,
Sean
On 4/12/10 2:04 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:

> Hi Sean,
>  
> I don?t know about your DTMF issue but I also had to add a directory extension
> to the dialplan to go with the change to the IVR
>  
> IVR:
>    <entry action="menu-exec-app" digits="5" param="transfer directory XML
> default"/>    <!-- Directory -->
>  
> dialplan/defaults.xml:
>     <!-- dial by name directory extension -->
>     <extension name="directory">
>       <condition field="destination_number" expression="^directory$">
>        <action application="answer"/>
>        <action application="sleep" data="500"/>
>        <action application="directory" data="default $${domain} default"/>
>       </condition>
>     </extension>
>  
> Regards,
> Robert
>  
> 
> 
> From: Sean Holt [mailto:sean at obscuradigital.com]
> Sent: Monday, April 12, 2010 1:54 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Mod directory
>  
> Ok I?ve downloaded the sounds and setup my ivr menu, but now it doesn?t
> recognizes any incoming DTMF tones.  I?ve run a wireshark capture and know
> that DTMF tones are successful when calling into a conference line, but
> nothing on the ivr menu.
> 
> Thoughts?
> Thanks
> Sean
> 
> 
> On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
> You do not need to defined sound files in ivr. Sounds files are used by mod
> directory.
> 
>  Please read the wiki, in the last paragraph.
> 
> 
> 
> Please note as of 2010/02/24, the default en/us/callie sounds do not contain
> the sounds needed for the directory. The current work-around is to
> copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
> <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/>  to
> {sounds_dir}/en/us/callie/directory/48000
> 
> 
> 
> 
> On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:
> Hey Lloyd,
> 
> When you say download sound files, where from?  And how is that defined in the
> ivr menu?
> 
> I looked over the wiki but couldn?t figure out the flow.
> 
> Plus I keep getting this message after enter the digit 5
> 
> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I thought
> you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix it......
> 
> Thanks
> Sean
> 
> 
> 
> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
> <http://lloyd.aloysius at gmail.com> > wrote:
> Here is the syntax I am using inside the ivr menus
> 
>  <entry action="menu-exec-app" digits="5" param="directory default
> ${domain_name} "/>
> 
> When the user press 5 .. the Directory Application Will start.
> ---
> 
> Here is the wiki.
> 
> http://wiki.freeswitch.org/wiki/Mod_directory
> 
> You need to download sound files manually.The users returned MUST be a member
> of a group.
> 
> Thanks
> Lloyd
> 
> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
> <http://sean at obscuradigital.com> > wrote:
> Hello,
> 
> Trying to figure out how to use the mod directory app within an ivr menu.  Not
> sure of the syntax to use when building the ivr.  Not ever sure it can be
> done.
> 
> Also does mod_directory need to be added to a dialplan?
> 
> Thanks
> Sean 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From larclap at yahoo.com  Mon Apr 12 14:34:17 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 12 Apr 2010 14:34:17 -0700
Subject: [Freeswitch-users] Updating via git
Message-ID: <00f701cada87$ea2a47c0$be7ed740$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and fs_cli was gone. Are the contents of this
directory removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

 

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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From robert.hadley at teotech.com  Mon Apr 12 14:45:49 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 12 Apr 2010 14:45:49 -0700
Subject: [Freeswitch-users] FW:  Mod directory
In-Reply-To: <C7E8DEFD.8485%sean@obscuradigital.com>
References: <13C1415E7B544698847EF99C548CE92C@greyhawk.tonecommander.com>
	<C7E8DEFD.8485%sean@obscuradigital.com>
Message-ID: <C11A1FF2F7044D3F95FC0DE35D788CEF@greyhawk.tonecommander.com>

Hi Sean,

 

The default $${domain} default came right out of the wiki

 

http://wiki.freeswitch.org/wiki/Mod_directory

 

I'm not sure about the Invalid Application directory.  Check that you've
enabled mod_directory in both the build (uncomment mod_directory in
module.conf) and at runtime in conf/autoload_config/modules.conf.xml
(uncomment mod_directory).

 

Also, don't forget as per the wiki you need to set the
effective_caller_id_name to a name (first last) or add new param
directory_full_name in the directory/default/lineId.xml files.

 

Regards,

Robert

 

 

  _____  

From: Sean Holt [mailto:sean at obscuradigital.com] 
Sent: Monday, April 12, 2010 2:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FW: Mod directory

 

Ok well seems that my DTMF are working again, but after making the changes
Robert recommended I'm now getting this error message

Invalid Application directory

Maybe my confusion is in the default $${domain} default line.  Not sure what
default represents.

I feel I've almost got this figured out.  Any thoughts why I might get this
error?
Thanks,
Sean
On 4/12/10 2:04 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:

Hi Sean,
 
I don't know about your DTMF issue but I also had to add a directory
extension to the dialplan to go with the change to the IVR
 
IVR:
   <entry action="menu-exec-app" digits="5" param="transfer directory XML
default"/>    <!-- Directory -->

dialplan/defaults.xml:
    <!-- dial by name directory extension -->
    <extension name="directory">
      <condition field="destination_number" expression="^directory$">
       <action application="answer"/>
       <action application="sleep" data="500"/>
       <action application="directory" data="default $${domain} default"/>
      </condition>
    </extension>
 
Regards,
Robert
 

  _____  

From: Sean Holt [mailto:sean at obscuradigital.com] 
Sent: Monday, April 12, 2010 1:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Mod directory

Ok I've downloaded the sounds and setup my ivr menu, but now it doesn't
recognizes any incoming DTMF tones.  I've run a wireshark capture and know
that DTMF tones are successful when calling into a conference line, but
nothing on the ivr menu.

Thoughts?
Thanks
Sean


On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
You do not need to defined sound files in ivr. Sounds files are used by mod
directory.

 Please read the wiki, in the last paragraph.



Please note as of 2010/02/24, the default en/us/callie sounds do not contain
the sounds needed for the directory. The current work-around is to
copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
<http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/> to
{sounds_dir}/en/us/callie/directory/48000




On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:
Hey Lloyd,

When you say download sound files, where from?  And how is that defined in
the ivr menu?

I looked over the wiki but couldn't figure out the flow.

Plus I keep getting this message after enter the digit 5

2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
I see you have a Sonus!
FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
Sadly, my creator had to spend several hours figuring this out so I thought
you'd like to know that!
Don't worry, DTMF will work but you may want to ask them to fix it......

Thanks
Sean



On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
<http://lloyd.aloysius at gmail.com> > wrote:
Here is the syntax I am using inside the ivr menus

 <entry action="menu-exec-app" digits="5" param="directory default
${domain_name} "/>

When the user press 5 .. the Directory Application Will start.
---

Here is the wiki. 

http://wiki.freeswitch.org/wiki/Mod_directory

You need to download sound files manually.The users returned MUST be a
member of a group.

Thanks
Lloyd

On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
<http://sean at obscuradigital.com> > wrote:
Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean 

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

  _____  


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

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

  _____  


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

  _____  

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

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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From jjmartres at gmail.com  Mon Apr 12 13:34:24 2010
From: jjmartres at gmail.com (=?UTF-8?Q?Martr=C3=A8s_Jean=2DJacques?=)
Date: Mon, 12 Apr 2010 22:34:24 +0200
Subject: [Freeswitch-users] Can FS listen dtmf during bridge call ?
Message-ID: <m2g95237e3b1004121334h92c48783rf817ea6d059d211c@mail.gmail.com>

Hi all,

Does FS as the capabilities to listen dtmf during a bridge call ?

For example, call leg a and call leg b are bridged through  FS.
FS receive a specific dtmf from call leg a then play hold_music to call leg
b.

Is it possible and how can I do it ? Using an IVR app or just a diaplan ?

All my thanks to the one who enlighten me :)

/jean-jacques

-- 
MARTRES Jean-Jacques
email : jjmartres at gmail.com
website : http://www.jeanjacques.martres.info
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From david.ponzone at gmail.com  Mon Apr 12 15:01:03 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 13 Apr 2010 00:01:03 +0200
Subject: [Freeswitch-users] Can FS listen dtmf during bridge call ?
In-Reply-To: <m2g95237e3b1004121334h92c48783rf817ea6d059d211c@mail.gmail.com>
References: <m2g95237e3b1004121334h92c48783rf817ea6d059d211c@mail.gmail.com>
Message-ID: <30DE2803-25F0-4A4E-A7C4-AAB04C8E1D63@gmail.com>

Jean-Jacques,

yes FS can do that.
Look at bind_meta_app.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 12/04/2010 ? 22:34, Martr?s Jean-Jacques a ?crit :

> Hi all,
>
> Does FS as the capabilities to listen dtmf during a bridge call ?
>
> For example, call leg a and call leg b are bridged through  FS.
> FS receive a specific dtmf from call leg a then play hold_music to  
> call leg b.
>
> Is it possible and how can I do it ? Using an IVR app or just a  
> diaplan ?
>
> All my thanks to the one who enlighten me :)
>
> /jean-jacques
>
> -- 
> MARTRES Jean-Jacques
> email : jjmartres at gmail.com
> website : http://www.jeanjacques.martres.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

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From nandy1925 at gmail.com  Mon Apr 12 15:00:48 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 13 Apr 2010 06:00:48 +0800
Subject: [Freeswitch-users] Can FS listen dtmf during bridge call ?
In-Reply-To: <m2g95237e3b1004121334h92c48783rf817ea6d059d211c@mail.gmail.com>
References: <m2g95237e3b1004121334h92c48783rf817ea6d059d211c@mail.gmail.com>
Message-ID: <x2l7d0bfd8c1004121500i734c879fyf057d6cea5a6cdcc@mail.gmail.com>

have you considered bind_meta_app? you can find the details here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app

include this in the dialplan before you call "bridge".
-nandy

2010/4/13 Martr?s Jean-Jacques <jjmartres at gmail.com>

> Hi all,
>
> Does FS as the capabilities to listen dtmf during a bridge call ?
>
> For example, call leg a and call leg b are bridged through  FS.
> FS receive a specific dtmf from call leg a then play hold_music to call leg
> b.
>
> Is it possible and how can I do it ? Using an IVR app or just a diaplan ?
>
> All my thanks to the one who enlighten me :)
>
> /jean-jacques
>
> --
> MARTRES Jean-Jacques
> email : jjmartres at gmail.com
> website : http://www.jeanjacques.martres.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
>
>
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From mike at jerris.com  Sun Apr 11 17:39:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 20:39:19 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
Message-ID: <74497DDA-D84B-4DC1-B736-7AF1FC938782@jerris.com>

Why would we stop and restart from the beginning?


On Apr 7, 2010, at 2:03 PM, CHU, XINGJUN (XINGJUN) wrote:

> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun



From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Mon Apr 12 15:17:24 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 12 Apr 2010 15:17:24 -0700
Subject: [Freeswitch-users] FW:  Mod directory
In-Reply-To: <C11A1FF2F7044D3F95FC0DE35D788CEF@greyhawk.tonecommander.com>
Message-ID: <C7E8EA04.8491%sean@obscuradigital.com>

Sometimes it?s the simplest things that one overlooks.  I guess enabling
mod_directory would be critical....doh!

Anyways mod_directory is answering my calls now but still not getting any
results when looking up a name/extension

New error is sampling rate doesn?t match

Not sure if this is affecting my search, but at least I?m closer to my end
goal

Thanks
Sean


On 4/12/10 2:45 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:

> Hi Sean,
>  
> The default $${domain} default came right out of the wiki
>  
> http://wiki.freeswitch.org/wiki/Mod_directory
>  
> I?m not sure about the Invalid Application directory.  Check that you?ve
> enabled mod_directory in both the build (uncomment mod_directory in
> module.conf) and at runtime in conf/autoload_config/modules.conf.xml
> (uncomment mod_directory).
>  
> Also, don?t forget as per the wiki you need to set the
> effective_caller_id_name to a name (first last) or add new param
> directory_full_name in the directory/default/lineId.xml files.
>  
> Regards,
> Robert
>  
>  
> 
> 
> From: Sean Holt [mailto:sean at obscuradigital.com]
> Sent: Monday, April 12, 2010 2:30 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FW: Mod directory
>  
> Ok well seems that my DTMF are working again, but after making the changes
> Robert recommended I?m now getting this error message
> 
> Invalid Application directory
> 
> Maybe my confusion is in the default $${domain} default line.  Not sure what
> default represents.
> 
> I feel I?ve almost got this figured out.  Any thoughts why I might get this
> error?
> Thanks,
> Sean
> On 4/12/10 2:04 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:
> Hi Sean,
>  
> I don?t know about your DTMF issue but I also had to add a directory extension
> to the dialplan to go with the change to the IVR
>  
> IVR:
>    <entry action="menu-exec-app" digits="5" param="transfer directory XML
> default"/>    <!-- Directory -->
> 
> dialplan/defaults.xml:
>     <!-- dial by name directory extension -->
>     <extension name="directory">
>       <condition field="destination_number" expression="^directory$">
>        <action application="answer"/>
>        <action application="sleep" data="500"/>
>        <action application="directory" data="default $${domain} default"/>
>       </condition>
>     </extension>
>  
> Regards,
> Robert
>  
> 
> 
> From: Sean Holt [mailto:sean at obscuradigital.com]
> Sent: Monday, April 12, 2010 1:54 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Mod directory
> 
> Ok I?ve downloaded the sounds and setup my ivr menu, but now it doesn?t
> recognizes any incoming DTMF tones.  I?ve run a wireshark capture and know
> that DTMF tones are successful when calling into a conference line, but
> nothing on the ivr menu.
> 
> Thoughts?
> Thanks
> Sean
> 
> 
> On 4/12/10 10:13 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
> You do not need to defined sound files in ivr. Sounds files are used by mod
> directory.
> 
>  Please read the wiki, in the last paragraph.
> 
> 
> 
> Please note as of 2010/02/24, the default en/us/callie sounds do not contain
> the sounds needed for the directory. The current work-around is to
> copyhttp://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/
> <http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/> to
> {sounds_dir}/en/us/callie/directory/48000
> 
> 
> 
> 
> On Mon, Apr 12, 2010 at 12:59 PM, Sean Holt <sean at obscuradigital.com> wrote:
> Hey Lloyd,
> 
> When you say download sound files, where from?  And how is that defined in the
> ivr menu?
> 
> I looked over the wiki but couldn?t figure out the flow.
> 
> Plus I keep getting this message after enter the digit 5
> 
> 2010-04-12 09:52:05.373930 [WARNING] sofia_glue.c:3290 Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I thought
> you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix it......
> 
> Thanks
> Sean
> 
> 
> 
> On 4/12/10 9:00 AM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com
> <http://lloyd.aloysius at gmail.com> > wrote:
> Here is the syntax I am using inside the ivr menus
> 
>  <entry action="menu-exec-app" digits="5" param="directory default
> ${domain_name} "/>
> 
> When the user press 5 .. the Directory Application Will start.
> ---
> 
> Here is the wiki.
> 
> http://wiki.freeswitch.org/wiki/Mod_directory
> 
> You need to download sound files manually.The users returned MUST be a member
> of a group.
> 
> Thanks
> Lloyd
> 
> On Sat, Apr 10, 2010 at 4:19 PM, Sean Holt <sean at obscuradigital.com
> <http://sean at obscuradigital.com> > wrote:
> Hello,
> 
> Trying to figure out how to use the mod directory app within an ivr menu.  Not
> sure of the syntax to use when building the ivr.  Not ever sure it can be
> done.
> 
> Also does mod_directory need to be added to a dialplan?
> 
> Thanks
> Sean 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vfclists at googlemail.com  Mon Apr 12 05:22:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 12 Apr 2010 13:22:49 +0100
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
Message-ID: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>

Can Freeswitch be installed simply by zipping up the folder and unzipping it
to the destination?

Does it require some DLLs to be installed in the Windows system folder and
some registry entries as well?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From 12ukwn at gmail.com  Mon Apr 12 15:24:48 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 13 Apr 2010 00:24:48 +0200
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <w2r87f2f3b91004121315r96ee8de2l330268e33ab05325@mail.gmail.com>
References: <20100410143328.2c419080@anubis.defcon1>
	<w2r87f2f3b91004121315r96ee8de2l330268e33ab05325@mail.gmail.com>
Message-ID: <20100413002448.0aa172a2@anubis.defcon1>

Le Mon, 12 Apr 2010 13:15:15 -0700,
Michael Collins <msc at freeswitch.org> a ?crit :

Ok Mike, I've done this:
git pull
make current
and relaunched: still 100% CPU used :(
FreeSWITCH Version 1.0.head (git-5e31f52 2010-04-12 16:11:09 -0400) Started.

a top -H with a 0.5s delay shows apparently 8 process with the name
'freeswitch', 4 of 'em stays on top of CPU consumption, the other 4 are
varying.

after having FS stopped, I must reset the graphic console I use in order to
re-gain visibility of what I type.

launch is done through: .../freeswitch -u freeswitch -g freeswitch -c

shutdown stay so long on 'Event binding deleted for
mod_local_stream:SHUTDOWN' that I killed FS.

So I'm commented out 'mod_local_stream' and... It worked: CPU's down to
normal!?? (but I'm still obliged to reset my console)

If you have a clue about that I'd be glad to ear about it :)

> Update to latest and try again. Use top -H to see if a particular thread
> is causing the issue and report back...
> -MC
> 
> On Sat, Apr 10, 2010 at 5:33 AM, Jean-Yves F. Barbier
> <12ukwn at gmail.com>wrote:
> 
> > FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200)
> > Started.
> > ===============
> >
> > Hi list,
> >
> > since I left the old fashion bulding to use git (last version compiled
> > on Debian lenny), FS is chewing 100% CPU just after launch.
> >
> > not any red line in console, nor registered device, nor anything in the
> > log file.
> >
> > what could cause this behaviour?
> >
> > --
> > I do not take drugs -- I am drugs.
> >                -- Salvador Dali
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >



-- 
You own a dog, but you can only feed a cat.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From msc at freeswitch.org  Mon Apr 12 15:47:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 15:47:09 -0700
Subject: [Freeswitch-users] Updating via git
In-Reply-To: <00f701cada87$ea2a47c0$be7ed740$@com>
References: <00f701cada87$ea2a47c0$be7ed740$@com>
Message-ID: <z2s87f2f3b91004121547v3afcc37dk10c6759220a675af@mail.gmail.com>

I thought I answered this already, so check your history. In short: still
use 'make current' even if you're on git. The core files are most likely
nothing to worry about - you can delete them.
-MC

On Mon, Apr 12, 2010 at 2:34 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  Once we pull down the full FreeSWITCH version using git, should we
> continue to use 'make current' to update thereafter?
>
>
>
> I started with this command and interrupted it just after it had pulled the
> source changes. I was surprised to see that "fs_cli" no longer worked. I
> went to freeswitch/bin and fs_cli was gone. Are the contents of this
> directory removed in 'make current'?
>
>
>
> I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
> to remove these? They are about 200M each.
>
>
>
> Thanks, Lars
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Apr 12 16:02:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 16:02:12 -0700
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC361BA.504@codefix.net>
References: <4BC361BA.504@codefix.net>
Message-ID: <z2o87f2f3b91004121602m4b996dbas26e0e0fd02c75c2e@mail.gmail.com>

On Mon, Apr 12, 2010 at 11:08 AM, Garrison Hoffman <garrison at codefix.net>wrote:

> Using git://git.freeswitch.org/freeswitch.git
> commit e7ff9f8506dbbc57538e562b277f216bc31ecc93
> Date:   Fri Apr 9 16:03:54 2010 -0400
>
> [WARNING] switch_ivr_menu.c:704 Invalid Action [menu-say-text]
>
> Indeed, switch_ivr_action_t has no action "menu-say-text"
>
> Has this been removed or is it not yet supported? Using flite in
> greet-long/greet-short works fine, is there any way to use TTS in menu
> actions?
>
>
> Thanks,
>
> -gh
>
menu-exec-app
-MC
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew at hijacked.us  Mon Apr 12 16:10:10 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 12 Apr 2010 19:10:10 -0400
Subject: [Freeswitch-users] performance comparison between centos and
	freebsd
In-Reply-To: <w2l191c3a031004120958h34cf45dfiafa9acf29f2e4f1f@mail.gmail.com>
References: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>
	<w2l191c3a031004120958h34cf45dfiafa9acf29f2e4f1f@mail.gmail.com>
Message-ID: <20100412231010.GE3321@hijacked.us>

On Mon, Apr 12, 2010 at 11:58:07AM -0500, Anthony Minessale wrote:
> Either this will help:
> 
> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Threading_Library
> 
> or nothing will.
> 
> FreeBSD is always a pain to support because it's like a spoiled actor in a
> bad movie.
> It makes many demands on everyone, it should be called DivaOS.  Still we try
> to support it and if it does not
> work as well we really can't do much about it.
> 
>
Also, you completely neglected to mention what version of FreeBSD you're
using (you should probably be using 8.x).

Andrew


From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Mon Apr 12 16:16:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 16:16:22 -0700
Subject: [Freeswitch-users] (dynamic) user created conference rooms
In-Reply-To: <20100412171644.52330@gmx.net>
References: <20100412171644.52330@gmx.net>
Message-ID: <p2l87f2f3b91004121616l1b6d3060md5c9403f7930391a@mail.gmail.com>

The first way (Web page for users) is harder for you, easier for the users.
The second way (emails) is easier for you but much harder for the users. If
you go with the second method you're going to have to do a lot of work on
parsing the emails and your users will find a way to break it. You are
better off doing it correctly and just letting your users have a little Web
page to manage their conf rooms.
-MC

On Mon, Apr 12, 2010 at 10:16 AM, <excelsio at gmx.net> wrote:

> Hi,
>
> I successfully installed the basic conference servicee from the conference
> script on the wiki:
> http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>
> I don?t want to let my users tell me to create the conference rooms by
> hand, so I?d like to have a basic authentication system to let them
> create/delete their conference rooms. The question is, how? Well, I have a
> centralized user database. Therefore I could authenticate them with the help
> of RADIUS. But setting up a secured homepage to access the database? To
> complicated for me. Never done this. So I asked me whether the following
> would work:
>
> 1. users write an email to conference-setup at mydomain.com. They user their
> corporate email address and the corporate email server.
> 2. a script on the freeswitch host downloads the email and checks the
> header of the email: the sender?s address, i.e. user at mydomain.com and the
> used email server, i.e. email.mydomain.com
> (Will this be an adequate spoofing check? The email will be checked before
> against spam by the email server...). So the users are "authenticated"
> somehow at least.
> 3. a script checks the subject for e.g. "ADD", "CHANGE" or "DELETE"
> 4. the body should contain the "CONFERENCE ROOM NUMBER" in the first line,
> a "PASSWORD" in the second line, the "CONFERENCE ROOM NAME" in the third
> line or in c ase a user wants to delete an entry, the old password.
> 5. another script will take those entries, starts mysql and insert those
> values in the table of the database
>
> What do you think? Will this work? Has anyone already tried that? Are there
> easier options? It would be great if you could give some hints.
>
> thanks in advance
>
> Michael
> --
> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From noisewaterphd at gmail.com  Mon Apr 12 16:22:39 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Mon, 12 Apr 2010 17:22:39 -0600
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
	<y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>
Message-ID: <t2r1c9955401004121622ia85b5dcfj16ad932540a451ef@mail.gmail.com>

Can you explain a little bit about what your issues with this type of setup
were?  I'm very curious, any info will certainly help me.

Thanks

On Fri, Apr 9, 2010 at 2:11 PM, Brent Paddon <brent at overthewire.com.au>wrote:

> We had a lot of problems getting this to work properly for us (exact same
> Mitel box), but the last time we looked at it was probably 12 months ago.
>  We ended up with an asterisk box in between FS and the Mitel.  I would like
> to know if you have more success than us - perhaps we can revisit this one.
>
> Brent
>
> On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <
> noisewaterphd at gmail.com> wrote:
>
>> Hi,
>>
>> I've already read the interop list, but I'm wondering if anyone on here
>> has anymore experience/info on trunking freeswitch to a Mitel 3300?
>>
>> Specifically, I want to use freeswitch for acd and sip registrations, and
>> just use our mitel for switching to the PSTN.
>>
>> Does anyone have some good info to share?
>>
>> Thanks,
>>
>> Kenny
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> --
> Brent Paddon
>
> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
> www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 16:24:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 16:24:53 -0700
Subject: [Freeswitch-users] Multi-tenanting
In-Reply-To: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>
References: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>
Message-ID: <u2g87f2f3b91004121624q41770c60j16cad44fbe0b6d35@mail.gmail.com>

Just in case, be sure to read this:
http://wiki.freeswitch.org/wiki/Multi-tenant

The OP there uses the XML files for all of the configuration. While I
recommend getting to know the XML files I would also say that using
mod_xml_curl would give you a measure of power and flexibility for keeping
your directory and dialplan information in a database.

As far as being too good to be true... it almost is! The catch really is the
learning curve, although if you've already dealt with Asterisk, Mitel,
ShoreTel, etc. then you've got a few battle scars already, so a steep
learning curve is nothing to fear, eh? :)

Be sure to join #freeswitch at irc.freenode.net if you want to chat in
real-time with other FreeSWITCH users. As for gotchas: be sure you
understand domains and if necessary have a good DNS server that does SRV
records properly.

Welcome to FreeSWITCH, it's going to be a wild ride! :)

-MC

On Fri, Apr 9, 2010 at 3:42 PM, Kenneth Noisewater
<noisewaterphd at gmail.com>wrote:

> Hi All,
>
> I'm just a few days into my FreeSwitch investigation, and so far I have to
> say, it seems almost too good to be true! Kudos to everyone involved. I've
> got  quite a bit of experience with other systems, ranging from Asterisk to
> Mitel/Shoretel type systems, and FreeSwitch is really looking good.
>
> Anyway...
>
> So pouring over configs, it seems it would be really simple to set up a
> multi tenant system with the whole 'domain' concept. Does this domain model
> maintain good seperation throughout the system? Is there anything to be
> aware of in doing a multi tenant setup?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mcampbellsmith at gmail.com  Mon Apr 12 16:33:51 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Apr 2010 09:33:51 +1000
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
	<B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>
Message-ID: <t2h33c87fa31004121633n9a1f70ftb455c511b5b210ef@mail.gmail.com>

I did a complete rebuild:
git pull && ./bootstrap.sh && ./configure && make install

this worked - but I'm not sure it was the correct way?


On Mon, Apr 12, 2010 at 3:05 AM, Michael Jerris <mike at jerris.com> wrote:
> If anyone is still having this issue, please open a bug for me on jira and
> provide privately via email information to remotely access the machine to
> troubleshoot.
> Mike
> On Apr 9, 2010, at 12:19 PM, DJB wrote:
>
> git pull
> make all
> make install
> -or-
> make current
> -djbinter
> ________________________________
> From:?Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To:?freeswitch-users at lists.freeswitch.org
> Sent:?Thu, April 8, 2010 8:48:17 PM
> Subject:?Re: [Freeswitch-users] version number: git checkout
>
> Thanks Milena... I upgraded git now - git version 1.7.0.4
>
> I did a 'get pull && make install' and still the same problem.
>
> Do I have to do a get clone or something?? Ideas?? Thanks
>
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
> ? ? ? ? ? ? _____ ____? ? ____ _? ? ___
> ? ? ? ? ? |? ___/ ___|? / ___| |? |_ _|
> ? ? ? ? ? | |_? \___ \? | |? | |? ? | |
> ? ? ? ? ? |? _|? ___) | | |___| |___ | |
> ? ? ? ? ? |_|? |____/? \____|_____|___|
>
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris? ? ? *
> * FreeSWITCH (http://www.freeswitch.org)? ? ? ? ? ? ? *
> * Paypal Donations Appreciated:?paypal at freeswitch.org?*
> * Brought to you by ClueCon?http://www.cluecon.com/? *
> *******************************************************
>
> Type /help <enter> to see a list of commands
>
>
> +OK log level? [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
>
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
>>
>>
>> Can anything be done in the freeswitch code so when "git pull" is
>> executed,
>> the "--pretty"?argument is?also set?where "--format" is set to make it
>> compatible with both older and newer versions of git? or it is all up to
>> what git does and nothing to do on fs?
>>
>>
>>
>> PS: Mark,?the issue you're facing is because of your version of git, the
>> CLI
>> shows the freeswitch version properly with git 1.7.0.4, the "format"
>> argument isn't recognized by your version of git.
>>
>>
>> 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
>>>
>>> Git was installed as described on the wiki. ?I am using Debian Lenny
>>> and Git version 1.5.6.5
>>>
>>> I just did a git pull and had the same issue... FS still shows
>>> FreeSWITCH Version 1.0.head (git-)
>>>
>>>
>>>
>>> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
>>> > Upgrade your git, then it will show it correctly.
>>> >?http://wiki.freeswitch.org/wiki/Git_Install
>>> > djbinter
>>> > ________________________________
>>> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
>>> > To:?freeswitch-users at lists.freeswitch.org
>>> > Sent: Wed, April 7, 2010 11:54:27 PM
>>> > Subject: [Freeswitch-users] version number: git checkout
>>> >
>>> > Hi!
>>> >
>>> > I just used git for the first time ever to checkout FreeSwitch as
>>> > described on the wiki at
>>> >?http://wiki.freeswitch.org/wiki/Installation_Guide
>>> >
>>> > Now my version number says:
>>> > FreeSWITCH Version 1.0.head (git-)
>>> >
>>> > Is there a mistake in my procedure or the building of FS when using
>>> > GIT?? Hard to know the build number of FS with a tag like that!
>>> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 12 16:42:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 16:42:04 -0700
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <t2h33c87fa31004121633n9a1f70ftb455c511b5b210ef@mail.gmail.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
	<B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>
	<t2h33c87fa31004121633n9a1f70ftb455c511b5b210ef@mail.gmail.com>
Message-ID: <l2r87f2f3b91004121642x1ccd4b26o671d8bc244372eb5@mail.gmail.com>

On Mon, Apr 12, 2010 at 4:33 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> I did a complete rebuild:
> git pull && ./bootstrap.sh && ./configure && make install
>
> this worked - but I'm not sure it was the correct way?
>
> You may have had to do that once but I think from now on you'll only need
to do make current.
-MC
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From msc at freeswitch.org  Mon Apr 12 16:48:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Apr 2010 16:48:03 -0700
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
	<4BC348B5.8060805@kinetix.gr>
	<o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
Message-ID: <o2s87f2f3b91004121648v6c217038j33c045292e342a90@mail.gmail.com>

2010/4/12 Alfredo Quiroga-Villamil <lawwton at gmail.com>

> Thanks for the response Vlasis. We run statistical analysis for FAS in
> our production routes.
>
> This request would be for a controlled environment where I am using
> GrandStreams 488 to convert from Voip to PSTN. I was thinking that
> perhaps FS has a way to allow me somehow access to the Signaling
> portion of the protocol. If that was the case, I would be able to hack
> my way around it and perhaps ignore the first 200 I receive and not
> propagate it to the origination gateway. This is the setup I am
> looking at:
>
> A => B => GrandStream Device => PSTN
>
>   SIP   SIP
>
> A/B - Both Gateways
>
> This is a test environment I am using, something I am just playing
> with. So in every case I will receive a FAS from the GrandStream
> device.
>
> Do you know if there is a way to launch an application from the
> DialPlan and have access to intercept Signaling; something that would
> allow me to see the messages coming in and make decisions based on
> them like propagate or do not propagate that message. I know this is a
> tough thing since I am literally asking for access to the SIP Stack.
>

I think those last two sentences describe a natural feature of a SIP proxy.
I'd check out OpenSIPS to see if it has the stack-level control and
functionality that you need.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From garrison at codefix.net  Mon Apr 12 16:52:31 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Mon, 12 Apr 2010 19:52:31 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC361BA.504@codefix.net>
References: <4BC361BA.504@codefix.net>
Message-ID: <4BC3B23F.5070700@codefix.net>

I got annoyed at the thought of using the application API to accomplish 
something already built-in, so a bit of wading through the source gave 
up the answer:

<entry action="menu-play-sound" digits="3" param="say: You pressed 3"/>

'Twould be nice if someone updated demo_ivr.xml with the correct syntax.

-gh


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From anthony.minessale at gmail.com  Mon Apr 12 17:21:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Apr 2010 19:21:05 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <20100413002448.0aa172a2@anubis.defcon1>
References: <20100410143328.2c419080@anubis.defcon1>
	<w2r87f2f3b91004121315r96ee8de2l330268e33ab05325@mail.gmail.com>
	<20100413002448.0aa172a2@anubis.defcon1>
Message-ID: <s2x191c3a031004121721s3a3b05ebwe3c803b5206a5510@mail.gmail.com>

did you run make sounds-install ?


On Mon, Apr 12, 2010 at 5:24 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Le Mon, 12 Apr 2010 13:15:15 -0700,
> Michael Collins <msc at freeswitch.org> a ?crit :
>
> Ok Mike, I've done this:
> git pull
> make current
> and relaunched: still 100% CPU used :(
> FreeSWITCH Version 1.0.head (git-5e31f52 2010-04-12 16:11:09 -0400)
> Started.
>
> a top -H with a 0.5s delay shows apparently 8 process with the name
> 'freeswitch', 4 of 'em stays on top of CPU consumption, the other 4 are
> varying.
>
> after having FS stopped, I must reset the graphic console I use in order to
> re-gain visibility of what I type.
>
> launch is done through: .../freeswitch -u freeswitch -g freeswitch -c
>
> shutdown stay so long on 'Event binding deleted for
> mod_local_stream:SHUTDOWN' that I killed FS.
>
> So I'm commented out 'mod_local_stream' and... It worked: CPU's down to
> normal!?? (but I'm still obliged to reset my console)
>
> If you have a clue about that I'd be glad to ear about it :)
>
> > Update to latest and try again. Use top -H to see if a particular thread
> > is causing the issue and report back...
> > -MC
> >
> > On Sat, Apr 10, 2010 at 5:33 AM, Jean-Yves F. Barbier
> > <12ukwn at gmail.com>wrote:
> >
> > > FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200)
> > > Started.
> > > ===============
> > >
> > > Hi list,
> > >
> > > since I left the old fashion bulding to use git (last version compiled
> > > on Debian lenny), FS is chewing 100% CPU just after launch.
> > >
> > > not any red line in console, nor registered device, nor anything in the
> > > log file.
> > >
> > > what could cause this behaviour?
> > >
> > > --
> > > I do not take drugs -- I am drugs.
> > >                -- Salvador Dali
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
>
>
>
> --
> You own a dog, but you can only feed a cat.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Mon Apr 12 18:02:19 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Mon, 12 Apr 2010 21:02:19 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <o2s87f2f3b91004121648v6c217038j33c045292e342a90@mail.gmail.com>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
	<4BC348B5.8060805@kinetix.gr>
	<o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
	<o2s87f2f3b91004121648v6c217038j33c045292e342a90@mail.gmail.com>
Message-ID: <l2o5fe6fa8f1004121802v4c9a651sfafa1c1f8486edb6@mail.gmail.com>

Thanks Michael. Indeed, OpenSips has exactly what I need in this case.

Appreciate the response.

Alfredo

On Mon, Apr 12, 2010 at 7:48 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> 2010/4/12 Alfredo Quiroga-Villamil <lawwton at gmail.com>
>>
>> Thanks for the response Vlasis. We run statistical analysis for FAS in
>> our production routes.
>>
>> This request would be for a controlled environment where I am using
>> GrandStreams 488 to convert from Voip to PSTN. I was thinking that
>> perhaps FS has a way to allow me somehow access to the Signaling
>> portion of the protocol. If that was the case, I would be able to hack
>> my way around it and perhaps ignore the first 200 I receive and not
>> propagate it to the origination gateway. This is the setup I am
>> looking at:
>>
>> A => B => GrandStream Device => PSTN
>>
>> ? SIP ? SIP
>>
>> A/B - Both Gateways
>>
>> This is a test environment I am using, something I am just playing
>> with. So in every case I will receive a FAS from the GrandStream
>> device.
>>
>> Do you know if there is a way to launch an application from the
>> DialPlan and have access to intercept Signaling; something that would
>> allow me to see the messages coming in and make decisions based on
>> them like propagate or do not propagate that message. I know this is a
>> tough thing since I am literally asking for access to the SIP Stack.
>
> I think those last two sentences describe a natural feature of a SIP proxy.
> I'd check out OpenSIPS to see if it has the stack-level control and
> functionality that you need.
> -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 mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From 12ukwn at gmail.com  Mon Apr 12 19:51:38 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Tue, 13 Apr 2010 04:51:38 +0200
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <s2x191c3a031004121721s3a3b05ebwe3c803b5206a5510@mail.gmail.com>
References: <20100410143328.2c419080@anubis.defcon1>
	<w2r87f2f3b91004121315r96ee8de2l330268e33ab05325@mail.gmail.com>
	<20100413002448.0aa172a2@anubis.defcon1>
	<s2x191c3a031004121721s3a3b05ebwe3c803b5206a5510@mail.gmail.com>
Message-ID: <20100413045138.19760612@anubis.defcon1>

Le Mon, 12 Apr 2010 19:21:05 -0500,
Anthony Minessale <anthony.minessale at gmail.com> a ?crit :

Yep, I installed sounds.

I just finished to recompiled by the quick'n'dirty method (svn-17188),
and all is fine!

> did you run make sounds-install ?

-- 
You can only live once, but if you do it right, once is enough.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 17:39:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 20:39:19 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
Message-ID: <74497DDA-D84B-4DC1-B736-7AF1FC938782@jerris.com>

Why would we stop and restart from the beginning?


On Apr 7, 2010, at 2:03 PM, CHU, XINGJUN (XINGJUN) wrote:

> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun



From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From yehavi.bourvine at gmail.com  Mon Apr 12 20:57:33 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 13 Apr 2010 06:57:33 +0300
Subject: [Freeswitch-users] Recommended phones
In-Reply-To: <j2g87f2f3b91004121234wa70c13q36098a48a8798de8@mail.gmail.com>
References: <l2od021d3be1004120904p7960d368r804b99f0d0fa9b4f@mail.gmail.com>
	<w2kd361ca7b1004121019obf5f9451n58efeb49827a545f@mail.gmail.com>
	<4BC35904.30604@codefix.net>
	<j2g87f2f3b91004121234wa70c13q36098a48a8798de8@mail.gmail.com>
Message-ID: <u2gd361ca7b1004122057mde39b63dz87bde8842e092609@mail.gmail.com>

Over the last three years we've tested quite a few brands of phones. Most of
them were real crap, only very few were reliable. From the better ones our
users preffered the Polycoms.
The Polycoms (and SNOMs) are probably the most expensive ones. Polycom has
the central boot server provisioning method which saves on initial setup
manpower (SNOM proobably has the same - didn't test).
Polycoms are also quite intiuitive for the user, so less support
personnel... This is currently crucial to us as we are deploying them using
already busy staff.

BTW, we did not tried Aastra as they are quite rare here.

                                 Regards, __Yehavi:

 2010/4/12 Michael Collins <msc at freeswitch.org>

>
>
>  On Mon, Apr 12, 2010 at 10:31 AM, Garrison Hoffman <garrison at codefix.net>wrote:
>
>> Yehavi Bourvine wrote:
>> > The law of "what you pay is what you get" works here. You want a good
>> > phone? Then pay for a SNOM or Polycom.
>>
>> Therefore Windows is better than Linux and SCO Unix is better than BSD?
>>  I think that law was repealed shortly after the Prohibition Era.
>>
>
> Sort of. "Payment" comes in many forms, not just $$.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at googlemail.com  Mon Apr 12 05:22:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 12 Apr 2010 13:22:49 +0100
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
Message-ID: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>

Can Freeswitch be installed simply by zipping up the folder and unzipping it
to the destination?

Does it require some DLLs to be installed in the Windows system folder and
some registry entries as well?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cucku.cucku at yahoo.com.vn  Mon Apr 12 21:09:08 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Tue, 13 Apr 2010 12:09:08 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?q?V=E1=BB=81=3A__need_help_on_variable?=
 =?utf-8?q?_and_param?=
In-Reply-To: <l2t87f2f3b91004091018nc95fb71au829b6837777fd5ba@mail.gmail.com>
References: <332483.42922.qm@web76207.mail.sg1.yahoo.com>
	<l2t87f2f3b91004091018nc95fb71au829b6837777fd5ba@mail.gmail.com>
Message-ID: <875671.76956.qm@web76214.mail.sg1.yahoo.com>

Hi Michael

thank you for your explanation
so the Variables are use for phone call, the params are used for gateway, and sip config file

i found the link that lists of all Variables
http://wiki.freeswitch.org/wiki/Category:Variable

so when i config the gateway/ create new sip profile . 
can i use the variable on the link above to config as a params for gateway/sip profile,  
if not where i can get the list of valid params for gateway + sip configuration

i check the 1000.xml, there is a tag <param> </param>, could you do a brief explain it
<include>
  <user id="1000">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

Thank you


________________________________
T?: Michael Collins <msc at freeswitch.org>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 0:18:54, Th? B?y, 10 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] need help on variable and param




2010/4/9 false <cucku.cucku at yahoo.com.vn>

Hi all
>
>
>i am confuse the param and variable in freeswitch, sometimes param is used for gateway, sometime variable is used for users

Remember that variables in this case are "channel variables" and are almost always associated with a specific phone call. These variables for a user are set whenever that user makes a phone call.

Parameters a generally used for configuration purposes. The "dial-string" parameter is related to the user channel, so when you do something like:
originate user/1001 &park()

It will use the dial-string parameter to turn "user/1001" into a proper dialstring. If it makes your brain hurt then don't worry about it. :)

Parameters are used in lots of configurations. The most obvious examples are in the SIP profiles and gateways. Are you working on something specific where the differences between params and variables are causing you trouble?

-MC

__________________________________________________
B?n C? S? D?ng Yahoo! Kh?ng?
M?t m?i v? th? r?c?  Yahoo! Th? c? ch??ng tr?nh b?o v? ch?ng th? r?c h?u hi?u nh?t tr?n m?ng 
http://vn.mail.yahoo.com 
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From infos at madovsky.org  Mon Apr 12 21:01:01 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 13 Apr 2010 00:01:01 -0400
Subject: [Freeswitch-users] 100& LSB compliant FS init script
Message-ID: <7FE4080A46DF4F57820DF3587C71E550@MOBILEE1705>

For a lot of Linux distrib it should work....
interesting for cluster mans

enjoy

Franck
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From infos at madovsky.org  Sun Apr 11 20:38:15 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 11 Apr 2010 23:38:15 -0400
Subject: [Freeswitch-users] FS pid
Message-ID: <48C3DBB008A1412C964E6F5535FD9D01@MOBILEE1705>

Hi all,

how can I change the name and the path of the pid file ?

Thanks

Franck
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 17:39:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 20:39:19 -0400
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
Message-ID: <74497DDA-D84B-4DC1-B736-7AF1FC938782@jerris.com>

Why would we stop and restart from the beginning?


On Apr 7, 2010, at 2:03 PM, CHU, XINGJUN (XINGJUN) wrote:

> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun



From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From mike at jerris.com  Mon Apr 12 21:48:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Apr 2010 00:48:10 -0400
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <4BC33F0F.8070208@gmail.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>
	<COL117-W71213F34122C75703A37CB9120@phx.gbl>
	<4BC33F0F.8070208@gmail.com>
Message-ID: <087B501F-8809-4CD7-A426-42F39D7568A9@jerris.com>


On Apr 12, 2010, at 11:41 AM, Tamas wrote:

> Is it possible to link against msvcrt.dll instead of compiler specific ones (e.g. msvcrt90.dll)?

Maybe, but I have not seen an easy way to do this with later compilers, and the old ones are generally pretty broken.  I doubt it would compile at all.

> In case we don't use any console stuff, is there a way to link without msvcrt at all?

Not sure the connection between console and msvcrt.dll.  We need a runtime regardless unless we somehow write c code that uses no functions from the c runtime.

Why is it an issue to depend on msvcrt90.dll?

> 
> Regards,
>     Tamas
> 
> Jeff Lenk ?rta:
>> 
>> Sure the only requirement is the 2008 CRuntime support. No files or registry settings needed for basic operation.
>>  
>> Date: Mon, 12 Apr 2010 13:22:49 +0100
>> From: vfclists at googlemail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
>> 
>> 
>> Can Freeswitch be installed simply by zipping up the folder and unzipping it to the destination?
>> 
>> Does it require some DLLs to be installed in the Windows system folder and some registry entries as well?
>> 

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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From mike at jerris.com  Mon Apr 12 21:52:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Apr 2010 00:52:37 -0400
Subject: [Freeswitch-users] call bargein (urgent call)
In-Reply-To: <20100412193319.E37C911492@mail.nstel.ru>
References: <20100412193319.E37C911492@mail.nstel.ru>
Message-ID: <5C3F423C-3B2C-46BE-9ADB-CBFDE45C0952@jerris.com>

You could do this with some clever application of mod_limit, and transferring everyone to a conference.

Mike

On Apr 12, 2010, at 3:33 PM, Nikolay Kondratyev wrote:

> Hi all,
> i'm still at a loss regarding incarnation of my idea (below).
> Can anybody please point me to an analogous example?
> Thanks and regards,
> Nikolay.
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> Sent: Friday, April 09, 2010 11:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] call bargein (urgent call)
> 
> Yes, that is exactly what i want.
> Nikolay.
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Friday, April 09, 2010 10:48 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] call bargein (urgent call)
> 
> 
> 
> On Thu, Apr 8, 2010 at 11:36 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:
> Hi all,
>  
> i need a sort of  "call barge in" functianality, ar may be "urgent call" will be better name... 
> I know about existing "eavesdrop" application, but it does not suite me because of two reasons:
> 1. callee must be on active call, or eavesdrop will fail.
> 2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If 1003 hangs up, the call will terminate.
>  
> Existing "intercept" application also does not suite.
>  
> So i need the following:
> A dispatcher must be able to call any user urgently, that is if a user is idle, it just should be a call, and when a user A is talking to user B, and dispatcher makes urgent call to A, all three must occur in a conference.
> I'm sure it's possible, but i'm rather new to FS and looks like i'm lost while searching how to achive it....
>  
> Can anybody please advise how to do it?
> 
> So in other words, if the dispatcher calls A, and A's phone is idle it just rings normally.
> If the dispatcher calls A while A is connected to B, then you want to throw A, B, and the dispatcher all into a conference?
> Just confirming that I understand what is happening.
> -MC
\
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From larclap at yahoo.com  Fri Apr  9 14:33:12 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:33:12 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011f01cad82c$420847c0$c618d740$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From jalsot at gmail.com  Tue Apr 13 00:50:10 2010
From: jalsot at gmail.com (Tamas)
Date: Tue, 13 Apr 2010 09:50:10 +0200
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <087B501F-8809-4CD7-A426-42F39D7568A9@jerris.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>	<COL117-W71213F34122C75703A37CB9120@phx.gbl>	<4BC33F0F.8070208@gmail.com>
	<087B501F-8809-4CD7-A426-42F39D7568A9@jerris.com>
Message-ID: <4BC42232.5070907@gmail.com>

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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From t.mahe at telemaque.fr  Tue Apr 13 03:10:05 2010
From: t.mahe at telemaque.fr (=?ISO-8859-15?Q?Tristan_Mah=E9?=)
Date: Tue, 13 Apr 2010 12:10:05 +0200
Subject: [Freeswitch-users] getting spammed from the list
Message-ID: <4BC442FD.8090506@telemaque.fr>

Hi everyone,

just a message because it's been 4 days now that I receive 
retransmission of mails sent to the ML ( not every mail, just a few, but 
I got them hundred of times )

Exemple of these mails:
- [Freeswitch-users] phocketsphinx from lua sent by janvb at live.com on 
10.04.2010 at 01:51
- Re: [Freeswitch-users] need help on variable and param sent by 
msc at freeswitch.org on 09.04.2010 at 19:18
- Re: [Freeswitch-users] Problem with IVR app starting too soon sent by 
bcxml at hotmail.com on 09.04.2010 at 19:45

Am I the only one impacted ?

Regards,

Gled.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From zolotov at altron.ua  Tue Apr 13 03:16:49 2010
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Tue, 13 Apr 2010 13:16:49 +0300
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <4BC442FD.8090506@telemaque.fr>
References: <4BC442FD.8090506@telemaque.fr>
Message-ID: <4BC44491.4090404@altron.ua>

  Nope, not only You. I have the same problem.

Tristan Mah? ?????:
> Hi everyone,
>
> just a message because it's been 4 days now that I receive 
> retransmission of mails sent to the ML ( not every mail, just a few, but 
> I got them hundred of times )
>
> Exemple of these mails:
> - [Freeswitch-users] phocketsphinx from lua sent by janvb at live.com on 
> 10.04.2010 at 01:51
> - Re: [Freeswitch-users] need help on variable and param sent by 
> msc at freeswitch.org on 09.04.2010 at 19:18
> - Re: [Freeswitch-users] Problem with IVR app starting too soon sent by 
> bcxml at hotmail.com on 09.04.2010 at 19:45
>
> Am I the only one impacted ?
>
> Regards,
>
> Gled.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From linux4michelle at tamay-dogan.net  Tue Apr 13 03:16:16 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 13 Apr 2010 12:16:16 +0200
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <4BC442FD.8090506@telemaque.fr>
References: <4BC442FD.8090506@telemaque.fr>
Message-ID: <20100413101616.GW12939@tamay-dogan.net>

Hello Tristan Mah?,

Am 2010-04-13 12:10:05, hacktest Du folgendes herunter:
> Hi everyone,
> 
> just a message because it's been 4 days now that I receive 
> retransmission of mails sent to the ML ( not every mail, just a few, but 
> I got them hundred of times )
> 
> Exemple of these mails:
> - [Freeswitch-users] phocketsphinx from lua sent by janvb at live.com on 
> 10.04.2010 at 01:51
> - Re: [Freeswitch-users] need help on variable and param sent by 
> msc at freeswitch.org on 09.04.2010 at 19:18
> - Re: [Freeswitch-users] Problem with IVR app starting too soon sent by 
> bcxml at hotmail.com on 09.04.2010 at 19:45
> 
> Am I the only one impacted ?

Do you have a message with a comlete header?

There are some peoples which are using GMail and forward the messages to
another account and  those  accounts  are  bouncing  or  forwarding  the
messages to the "To:" header instead to the maibox...

This is a configuration problem of the receiving server...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
Gesch. Michelle Konzack          Gesch. Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz               Kinzigstra?e 17
67100 Strasbourg/France         77694 Kehl/Germany
Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From bcxml at hotmail.com  Fri Apr  9 14:22:25 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 13:22:25 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270848145402-4879666.post@n2.nabble.com>


Here is the link 

http://pastebin.freeswitch.org/12675


Brian 
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4879666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From t.mahe at telemaque.fr  Tue Apr 13 03:41:44 2010
From: t.mahe at telemaque.fr (=?ISO-8859-15?Q?Tristan_Mah=E9?=)
Date: Tue, 13 Apr 2010 12:41:44 +0200
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <20100413101616.GW12939@tamay-dogan.net>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
Message-ID: <4BC44A68.9040209@telemaque.fr>

Hi Michelle,

We own our mailserver there, and I can assure it comes from FS ML 
server. You'll be able to see that from the headers below.

Here are two headers from mails I received a few minutes ago:

-----------------------------------------------------------------------

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X-Originating-IP: [83.109.87.96]
From: Jan Berger <janvb at live.com>
To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 10 Apr 2010 01:51:10 +0200
Importance: Normal
In-Reply-To: <1270833275502-4878370.post at n2.nabble.com>
References: <1269856895179-4817081.post at n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F at freeswitch.org>,
	<4BB1A30A.4060701 at gmail.com>,
	<1270796330100-4875677.post at n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42 at mail.gmail.com>,
	<1270833275502-4878370.post at n2.nabble.com>
MIME-Version: 1.0
X-OriginalArrivalTime: 09 Apr 2010 23:51:10.0485 (UTC)
	FILETIME=[87B5AC50:01CAD83F]
Subject: [Freeswitch-users] phocketsphinx from lua
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----------------------------------------------------------------------------

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	(envelope-from <mike at jerris.com>) id 1O12Bp-0006e2-Ob
	for freeswitch-users at lists.freeswitch.org;
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	Sun, 11 Apr 2010 10:06:10 -0700 (PDT)
From: Michael Jerris <mike at jerris.com>
Mime-Version: 1.0 (Apple Message framework v1078)
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In-Reply-To: <81019.71788.qm at web37501.mail.mud.yahoo.com>
To: freeswitch-users at lists.freeswitch.org
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88 at mail.gmail.com>
	<738759.37967.qm at web37505.mail.mud.yahoo.com>
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Errors-To: freeswitch-users-bounces at lists.freeswitch.org





Le 13.04.2010 12:16, Michelle Konzack a ?crit :
> Hello Tristan Mah?,
>
> Am 2010-04-13 12:10:05, hacktest Du folgendes herunter:
>> Hi everyone,
>>
>> just a message because it's been 4 days now that I receive
>> retransmission of mails sent to the ML ( not every mail, just a few, but
>> I got them hundred of times )
>>
>> Exemple of these mails:
>> - [Freeswitch-users] phocketsphinx from lua sent by janvb at live.com on
>> 10.04.2010 at 01:51
>> - Re: [Freeswitch-users] need help on variable and param sent by
>> msc at freeswitch.org on 09.04.2010 at 19:18
>> - Re: [Freeswitch-users] Problem with IVR app starting too soon sent by
>> bcxml at hotmail.com on 09.04.2010 at 19:45
>>
>> Am I the only one impacted ?
>
> Do you have a message with a comlete header?
>
> There are some peoples which are using GMail and forward the messages to
> another account and  those  accounts  are  bouncing  or  forwarding  the
> messages to the "To:" header instead to the maibox...
>
> This is a configuration problem of the receiving server...
>
> Thanks, Greetings and nice Day/Evening
>      Michelle Konzack
>      Systemadministrator
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Cordialement,

**********************************
* Tristan Mah?                   *
* T?l?maque - Service Voix       *
* Tel: +33 4.92.90.99.85         *
* Mob: +33 6.24.16.43.01         *
* Fax: +33 4.92.90.91.46         *
**********************************


From bcxml at hotmail.com  Tue Apr 13 03:40:48 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 13 Apr 2010 06:40:48 -0400
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <20100413101616.GW12939@tamay-dogan.net>
References: <4BC442FD.8090506@telemaque.fr>,
	<20100413101616.GW12939@tamay-dogan.net>
Message-ID: <SNT135-w2B475F9CA560B57695990B7110@phx.gbl>


I am having the same problem

 

Actually I initially sent the message 'Problem with IVR app starting too soon' which is now being received many many times

 

 

Brian Campbell
 
> Date: Tue, 13 Apr 2010 12:16:16 +0200
> From: linux4michelle at tamay-dogan.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] getting spammed from the list
> 
> Hello Tristan Mah?,
> 
> Am 2010-04-13 12:10:05, hacktest Du folgendes herunter:
> > Hi everyone,
> > 
> > just a message because it's been 4 days now that I receive 
> > retransmission of mails sent to the ML ( not every mail, just a few, but 
> > I got them hundred of times )
> > 
> > Exemple of these mails:
> > - [Freeswitch-users] phocketsphinx from lua sent by janvb at live.com on 
> > 10.04.2010 at 01:51
> > - Re: [Freeswitch-users] need help on variable and param sent by 
> > msc at freeswitch.org on 09.04.2010 at 19:18
> > - Re: [Freeswitch-users] Problem with IVR app starting too soon sent by 
> > bcxml at hotmail.com on 09.04.2010 at 19:45
> > 
> > Am I the only one impacted ?
> 
> Do you have a message with a comlete header?
> 
> There are some peoples which are using GMail and forward the messages to
> another account and those accounts are bouncing or forwarding the
> messages to the "To:" header instead to the maibox...
> 
> This is a configuration problem of the receiving server...
> 
> Thanks, Greetings and nice Day/Evening
> Michelle Konzack
> Systemadministrator
> 
> -- 
> ##################### Debian GNU/Linux Consultant ######################
> Development of Intranet and Embedded Systems with Debian GNU/Linux
> 
> itsystems at tdnet France itsystems at tdnet UG (haftungsbeschr?nkt)
> Gesch. Michelle Konzack Gesch. Michelle Konzack
> 
> Apt. 917 (homeoffice)
> 50, rue de Soultz Kinzigstra?e 17
> 67100 Strasbourg/France 77694 Kehl/Germany
> Tel: +33-6-61925193 mobil Tel: +49-177-9351947 mobil
> Tel: +33-9-52705884 fix
> 
> <http://www.itsystems.tamay-dogan.net/> <http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/> <http://www.can4linux.org/>
> 
> Jabber linux4michelle at jabber.ccc.de
> ICQ #328449886
> 
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
 		 	   		  
_________________________________________________________________
Got a phone? Get Hotmail & Messenger for mobile!
http://go.microsoft.com/?linkid=9724464
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From lloyd.aloysius at gmail.com  Fri Apr  9 14:02:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 17:02:37 -0400
Subject: [Freeswitch-users] Read Dialed Extension Variable Values
Message-ID: <l2t8a19bf2e1004091402rbd1b5ac0jdbc48b64d29b397@mail.gmail.com>

Hi All,

I have the following variables setup for a Extension 201

<variable name="followme" value="yes"/>
<variable name="followme_number" value="14161235555"/>

How to access these variables when another Extension say 202 calling to 201
or when an external call transfer to extension 201.

Thanks.
Lloyd
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From linux4michelle at tamay-dogan.net  Tue Apr 13 03:53:26 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 13 Apr 2010 12:53:26 +0200
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <4BC44A68.9040209@telemaque.fr>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
	<4BC44A68.9040209@telemaque.fr>
Message-ID: <20100413105326.GY12939@tamay-dogan.net>

Hello Tristan and "postmaster of <ostag.org>",

Am 2010-04-13 12:41:44, hacktest Du folgendes herunter:
> Hi Michelle,
> 
> We own our mailserver there, and I can assure it comes from FS ML 
> server. You'll be able to see that from the headers below.
> 
> Here are two headers from mails I received a few minutes ago:
<snip>

I have checked my mailserver too and found arround 1740 dupes...  which
where detected by my spamfilers as dulicated.

It seems there is a problem on <yoda.ostag.org>, because  the  first  or
second received header are intact, but it changed after it was  received
by <yoda.ostag.org>.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
Gesch. Michelle Konzack          Gesch. Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz               Kinzigstra?e 17
67100 Strasbourg/France         77694 Kehl/Germany
Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From yivzhenko at mksat.net  Tue Apr 13 03:54:31 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Tue, 13 Apr 2010 13:54:31 +0300
Subject: [Freeswitch-users] Bad HTTP POST data in  xml_cdr
Message-ID: <201004131354.31349.yivzhenko@mksat.net>

I use xml_cdr for logging
And i have problems with non-english letters in variable 'caller_id_name'

HTTP POST data terminates ".......20%3Ccaller_id_name%3E%"
 when variable 'caller_id_name' contains non-english letters
so some variables do not get to HTTP CGI program

Can i restrict users with non-english letters in 'caller_id_name' or rewrite 
this variable for a-leg?








From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From msc at freeswitch.org  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From larclap at yahoo.com  Fri Apr  9 14:33:12 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:33:12 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011f01cad82c$420847c0$c618d740$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 maciej.aniserowicz at gmail.com  Sat Apr 10 13:29:48 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Apr 2010 12:29:48 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
Message-ID: <1270931388804-4883374.post@n2.nabble.com>


Sure, it's here: http://pastebin.freeswitch.org/12566 .
(sorry it's posted so late, I posted this earlier via nabble but the post
was in a "pending" state for several days)
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4883374.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From thomas.ji at gmx.at  Fri Apr  9 12:41:36 2010
From: thomas.ji at gmx.at (thomas.ji at gmx.at)
Date: Fri, 09 Apr 2010 21:41:36 +0200
Subject: [Freeswitch-users] sip uri incomming calls
Message-ID: <20100409194136.268140@gmx.net>

hello list!
can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
what do i have to change in which files?

thank you very much for your help!

thomas
-- 
GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From brent at overthewire.com.au  Tue Apr 13 04:31:26 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Tue, 13 Apr 2010 21:31:26 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <t2r1c9955401004121622ia85b5dcfj16ad932540a451ef@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com> 
	<y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com> 
	<t2r1c9955401004121622ia85b5dcfj16ad932540a451ef@mail.gmail.com>
Message-ID: <p2sad139bd21004130431k2318950dr2356c656181453be@mail.gmail.com>

For us, at the time, it was primarily getting the Mitel to keep a solid
registration (this was over the Internet, and if I recall through NAT too).
 One thing I vaguely remember was that there was a problem identified with
the nonce coming back not matching on subsequent registrations after the
first.

We knew nothing about the Mitel box, and the people who did were a little
unhelpful.  Putting the Asterisk box in the middle fixed all issues at the
time, but this has not been revisited for 12 months.  Perhaps things are a
little different now, or you can get away without needing registration ?

Brent

On Tue, Apr 13, 2010 at 9:22 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Can you explain a little bit about what your issues with this type of setup
> were?  I'm very curious, any info will certainly help me.
>
> Thanks
>
>
> On Fri, Apr 9, 2010 at 2:11 PM, Brent Paddon <brent at overthewire.com.au>wrote:
>
>> We had a lot of problems getting this to work properly for us (exact same
>> Mitel box), but the last time we looked at it was probably 12 months ago.
>>  We ended up with an asterisk box in between FS and the Mitel.  I would like
>> to know if you have more success than us - perhaps we can revisit this one.
>>
>> Brent
>>
>> On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <
>> noisewaterphd at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I've already read the interop list, but I'm wondering if anyone on here
>>> has anymore experience/info on trunking freeswitch to a Mitel 3300?
>>>
>>> Specifically, I want to use freeswitch for acd and sip registrations, and
>>> just use our mitel for switching to the PSTN.
>>>
>>> Does anyone have some good info to share?
>>>
>>> Thanks,
>>>
>>> Kenny
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> --
>> Brent Paddon
>>
>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
>> www.overthewire.com.au
>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 itamar at ispbrasil.com.br  Tue Apr 13 04:36:25 2010
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Tue, 13 Apr 2010 08:36:25 -0300
Subject: [Freeswitch-users] sip uri incomming calls
In-Reply-To: <20100409194136.268140@gmx.net>
References: <20100409194136.268140@gmx.net>
Message-ID: <x2ud3ceea381004130436j592331c5h380ac41a6ae6a73@mail.gmail.com>

On Fri, Apr 9, 2010 at 4:41 PM,  <thomas.ji at gmx.at> wrote:
> hello list!
> can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet.
> what do i have to change in which files?
>
> thank you very much for your help!
>
> thomas
> --
> GMX.at - ?sterreichs FreeMail-Dienst mit ?ber 2 Mio Mitgliedern
> E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail

look ->


http://www.voip-info.org/wiki/view/DNS+SRV


-- 
------------

Itamar Reis Peixoto

e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vhatz at kinetix.gr  Tue Apr 13 04:45:37 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 13 Apr 2010 14:45:37 +0300
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>	<4BC348B5.8060805@kinetix.gr>
	<o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
Message-ID: <4BC45961.6020305@kinetix.gr>

Hello Alfredo,


On 12/4/10 8:23 ??, Alfredo Quiroga-Villamil wrote:
> Thanks for the response Vlasis. We run statistical analysis for FAS in
> our production routes.
>
> This request would be for a controlled environment where I am using
> GrandStreams 488 to convert from Voip to PSTN. I was thinking that
> perhaps FS has a way to allow me somehow access to the Signaling
> portion of the protocol. If that was the case, I would be able to hack
> my way around it and perhaps ignore the first 200 I receive and not
> propagate it to the origination gateway. This is the setup I am
> looking at:
>
>    

Are you absolutely certain that when your carrier (or test rig) gives 
FAS you receive two 200 messages instead of one? Have you verified it 
using Wireshark captures?

> A =>  B =>  GrandStream Device =>  PSTN
>
>     SIP   SIP
>
> A/B - Both Gateways
>
> This is a test environment I am using, something I am just playing
> with. So in every case I will receive a FAS from the GrandStream
> device.
>
> Do you know if there is a way to launch an application from the
> DialPlan and have access to intercept Signaling; something that would
> allow me to see the messages coming in and make decisions based on
> them like propagate or do not propagate that message. I know this is a
> tough thing since I am literally asking for access to the SIP Stack.
>
>    

Perhaps there is a way to create an app to detect when a SIP 200 message 
arrives, but I still have reservations whether 2 such messages arrive... 
I'd like to help you in this, so, do you have a packet capture file to 
share where 2 200 SIP messaes are sent?

Regards,
Vlasis.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From fraserredmond at gmail.com  Fri Apr  9 14:36:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 22:36:19 +0100
Subject: [Freeswitch-users] UPDATE: G.729 Codec Licensing *IS* Available
In-Reply-To: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
References: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>
Message-ID: <q2lc32ee1cf1004091436tc655d29bh3317d714923ca9cd@mail.gmail.com>

Can the licenses be transferred between computers?  If I have a cluster of 5
freeswitch servers do I need to assign a set of licenses to each
specifically?

I know that if you get a bunch of Howler's license they sit in a pool,
attached to one master computer, and your other computers can all use that
pool of licenses as long as the master server is running (or has been
running in the last week.) But they are tied to the mac address of that
master computer.

Cheers,
Fraser




On Thu, Apr 8, 2010 at 10:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Greetings all,
>
> The FreeSWITCH team would like to let everyone know that we do indeed sell
> g.729 licenses for $10 each. Use this link to initiate a purchase:
> http://www.freeswitch.org/node/235
>
> Note: licenses are available only for Linux-based systems at this time.
> Please stay tuned for updates.
>
> The INSTALL.txt file has very detailed instructions:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Keep in mind that a single license includes one encoder and one decoder,
> that is, it can transcode both directions of a single phone call. If you
> have any other questions please email us here or join us in #freeswitch on
> irc.freenode.net.
>  -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 ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From mike at jerris.com  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 12:23:35 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 22:23:35 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
Message-ID: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>

Hi,
Thanks for the hints.

When I am doing this, calling party *does not hear any ringtone* during this
pause.

session:execute("pre_answer");
session:execute("sleep","5000");


I tried to add the following (before or after pre-answer), but same result -
only silence:
session:setVariable("ringback", "%(2000,4000,440,480)");


I need ringtone. What am I doing wrong?

Thank you,
Vitalie

2010/4/8 David Ponzone <david.ponzone at gmail.com>

> Perhaps:
> pre_ answer
> then
> sleep ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>
> Hi dear community,
>
> I am using a Lua script that is being executed when a call reaches a
> particular extension, say 1001.
> It works ok, but it answers immediately when call reaches the system.
>
> How can I make it to wait 5-10 seconds (so the caller will hear several
> ringtones) and only after that the Lua script should answer and start the
> processing?
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From 12ukwn at gmail.com  Sat Apr 10 05:33:28 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 10 Apr 2010 14:33:28 +0200
Subject: [Freeswitch-users] 100% CPU
Message-ID: <20100410143328.2c419080@anubis.defcon1>

FreeSWITCH Version 1.0.head (git-555b205 2010-04-09 16:54:29 +0200) Started.
===============

Hi list,

since I left the old fashion bulding to use git (last version compiled 
on Debian lenny), FS is chewing 100% CPU just after launch.

not any red line in console, nor registered device, nor anything in the log
file.

what could cause this behaviour?

-- 
I do not take drugs -- I am drugs.
		-- Salvador Dali


From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From dule.maillist at gmail.com  Fri Apr  9 16:34:10 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Fri, 9 Apr 2010 19:34:10 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <m2m914fc92a1004091634waafe6e2fk8ff1cb4cb685ae69@mail.gmail.com>

Can you be a bit more specific what you want to accomplish?

We often trunk to a 3300 and I know many that have done so without issues;
the configuration is quite straightforward. Gateway configs in FS, SIP Peer
Profile (+ all the dependent forms) in the 3300 and you're pretty much good
to go.
Sorry I don't know what kind of information you're looking for in
particular.

Dan


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From vetali100 at gmail.com  Sun Apr 11 13:22:11 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 11 Apr 2010 23:22:11 +0300
Subject: [Freeswitch-users] How to make Lua script wait few seconds and
	then answer a call
In-Reply-To: <h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
References: <l2ra858e7691004071417q6737cf4nfee94f5be8d460ec@mail.gmail.com>
	<FA330144-96B8-46EB-8441-6BC6EFF8FB9D@gmail.com>
	<h2ua858e7691004111223yabc1c6b6t4f5ca7394ee14d0e@mail.gmail.com>
Message-ID: <y2pa858e7691004111322i6f2f17bes6440ee9e7779e1f7@mail.gmail.com>

[RESOLVED]

Looks like the following resolved my problem, calling party can hear
ringtone now:

session:execute("ring_ready");
session:execute("sleep","5000");

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

Thank you all,
Vitalie



2010/4/11 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
> Thanks for the hints.
>
> When I am doing this, calling party *does not hear any ringtone* during
> this pause.
>
> session:execute("pre_answer");
> session:execute("sleep","5000");
>
>
> I tried to add the following (before or after pre-answer), but same result
> - only silence:
> session:setVariable("ringback", "%(2000,4000,440,480)");
>
>
> I need ringtone. What am I doing wrong?
>
> Thank you,
> Vitalie
>
> 2010/4/8 David Ponzone <david.ponzone at gmail.com>
>
> Perhaps:
>> pre_ answer
>> then
>> sleep ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/04/2010 ? 23:17, Vitalii Colosov a ?crit :
>>
>> Hi dear community,
>>
>> I am using a Lua script that is being executed when a call reaches a
>> particular extension, say 1001.
>> It works ok, but it answers immediately when call reaches the system.
>>
>> How can I make it to wait 5-10 seconds (so the caller will hear several
>> ringtones) and only after that the Lua script should answer and start the
>> processing?
>>
>> Thank you,
>> Vitalie
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From lawwton at gmail.com  Tue Apr 13 05:30:18 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Tue, 13 Apr 2010 08:30:18 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
In-Reply-To: <4BC45961.6020305@kinetix.gr>
References: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>
	<4BC348B5.8060805@kinetix.gr>
	<o2s5fe6fa8f1004121023jceed81c7tf7d85a4c5fc03685@mail.gmail.com>
	<4BC45961.6020305@kinetix.gr>
Message-ID: <v2q5fe6fa8f1004130530k6932227eu2433e88c8336ea39@mail.gmail.com>

Hello Vlasis. I'll set something up this afternoon and send it to you
to make sure I am interpreting this correctly. Essentially the issue I
am experiencing in this case seems to be due to the VOIP ==> PSTN
conversion being done by the GrandStream.

I wasn't really using wireshark; just looking at the SIP trace after
having enabled SIP debug messages. I'll setup a test today and capture
the output of it and send it your way. If I am not mistaken what I am
seeing is a 200 OK message being sent by the GrandStream immediately
after doing the bridge from the VOIP leg to the PSTN side. Once the
bridge is created and the call sent over the PSTN, if an answer is
received, the GrandStream will then propagate a second 200.

A second set of eyes wouldn't hurt before I try to implement a hack
for this using OpenSips. In this case I seem to have the ability as
Michael previously pointed out to intercept Signaling and manipulate
it if needed. The basic idea I have is since I will always receive a
200 OK, (assuming I was correctly looking that day at the SIP trace)
is to simply ignore the first one and propagate just the second one
received. Which could also be a 486 or whatever other message. At that
point I wouldn't care, I would just propagate whatever I get after the
first 200.

I really appreciate the help with this since this could be a tricky
issue at times. Simplified in this case since I will always receive
the same result.

Alfredo

2010/4/13 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>:
> Hello Alfredo,
>
>
> On 12/4/10 8:23 ??, Alfredo Quiroga-Villamil wrote:
>> Thanks for the response Vlasis. We run statistical analysis for FAS in
>> our production routes.
>>
>> This request would be for a controlled environment where I am using
>> GrandStreams 488 to convert from Voip to PSTN. I was thinking that
>> perhaps FS has a way to allow me somehow access to the Signaling
>> portion of the protocol. If that was the case, I would be able to hack
>> my way around it and perhaps ignore the first 200 I receive and not
>> propagate it to the origination gateway. This is the setup I am
>> looking at:
>>
>>
>
> Are you absolutely certain that when your carrier (or test rig) gives
> FAS you receive two 200 messages instead of one? Have you verified it
> using Wireshark captures?
>
>> A => ?B => ?GrandStream Device => ?PSTN
>>
>> ? ? SIP ? SIP
>>
>> A/B - Both Gateways
>>
>> This is a test environment I am using, something I am just playing
>> with. So in every case I will receive a FAS from the GrandStream
>> device.
>>
>> Do you know if there is a way to launch an application from the
>> DialPlan and have access to intercept Signaling; something that would
>> allow me to see the messages coming in and make decisions based on
>> them like propagate or do not propagate that message. I know this is a
>> tough thing since I am literally asking for access to the SIP Stack.
>>
>>
>
> Perhaps there is a way to create an app to detect when a SIP 200 message
> arrives, but I still have reservations whether 2 such messages arrive...
> I'd like to help you in this, so, do you have a packet capture file to
> share where 2 200 SIP messaes are sent?
>
> Regards,
> Vlasis.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From vfclists at googlemail.com  Mon Apr 12 05:22:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 12 Apr 2010 13:22:49 +0100
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
Message-ID: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>

Can Freeswitch be installed simply by zipping up the folder and unzipping it
to the destination?

Does it require some DLLs to be installed in the Windows system folder and
some registry entries as well?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From troy at tlainvestments.com  Fri Apr  9 19:25:41 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 9 Apr 2010 19:25:41 -0700
Subject: [Freeswitch-users] An issue when attended transfer to fs
In-Reply-To: <47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
References: <47AB18AC0F23934383F2BBA7EE3D8D7421F7277057@DC-US1MBEX4.global.avaya.com>
	<h2ic29b930b1004080656s5d90bdfema2ed2c1b23ad5b@mail.gmail.com>
	<47AB18AC0F23934383F2BBA7EE3D8D7421F72B656C@DC-US1MBEX4.global.avaya.com>
Message-ID: <4D572B03-33E0-4751-8154-03C528C3FB14@tlainvestments.com>

Also, make sure you did <action application="answer" /> at some point before starting to play the file.  If you start to play a file without answering first, the first bit of the file is cut off.  Best practice seems to be something like:

<action application="answer" />
<action applicatino="sleep" data="1000"/>
<action application="playback" data="sounds/soundfile.wav"/>

-Troy


On Apr 9, 2010, at 8:27 AM, CHU, XINGJUN (XINGJUN) wrote:

> I don't see how this is relevant to the problem?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
> Sent: Thursday, April 08, 2010 9:57 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] An issue when attended transfer to fs
>  
> hi,
>  
> put an <action application="sleep" data="2000"/> before the instruction to play something
>  
> 
> 2010/4/7 CHU, XINGJUN (XINGJUN) <jasonchu at avaya.com>
> Hi,
> 
> I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.
> 
> Now I got a problem when A calls B then attended transfer B to freeswitch,
> The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.
> 
> Any suggestions are greatly appreciated.
> 
> Thanks
> Xingjun
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  9 10:50:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:50:50 -0700
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
	<4BB1A30A.4060701@gmail.com>
	<1270796330100-4875677.post@n2.nabble.com>
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Sure, it's here: http://pastebin.freeswitch.org/12566 .
>

I was not able to reproduce your symptoms, but I also don't have two
gateways to test with right now. However, I'm wondering if possibly you are
starting the uuid_record too quickly. Try doing the uuid_transfer without
the bgapi and then doing the uuid_record. I'm curious to see what happens.

For the record, I used your exact syntax, including uuids. The only
difference is that I used a pair of locally registered phones and did
"user/1001" and "user/1003" for my dialstrings.
-MC
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 17:30:27 2010
From: ash at url.net.au (Ash)
Date: Sun, 11 Apr 2010 10:30:27 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail fialover
Message-ID: <6147291A-E1C6-4B14-81F5-8141F79CBD02@url.net.au>


Hi All,

I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server.  Once the two calls are connected I would like it to play file that says something like "you are connected".  I have this component working by using a dynamic conference bridge.  The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line.  This is the dialplan I am trying:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>
          <action application="answer"/>
          <action application="playback" data="/tmp/pleasewait.wav"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxxx"/>
          <action application="log" data="INFO Voicemail"/>
          <action application="voicemail" data="default v2e 1000"/>
        </condition>
      </extension>
I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. 

Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line?  

Cheers,

Ash.
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From mattdfong at gmail.com  Sat Apr 10 21:13:10 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Apr 2010 21:13:10 -0700
Subject: [Freeswitch-users] Millisecond Precision for Lua Script
Message-ID: <i2j4256bf831004102113kd04ae2bemadd666f0b948cdef@mail.gmail.com>

I'm wondering if there is a way to get millisecond precision for a lua
script in freeswitch. I noticed that the api has a strepoch time, but it
only gives second precision. Lua does not have anything natively w/o adding
an extension, and just wondered if I'm over looking a method. Thanks.

--matt
hello hunter corp.
hosted predictive dialer - http://www.hellohunter.com
voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
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From brent at overthewire.com.au  Fri Apr  9 13:11:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 10 Apr 2010 06:11:33 +1000
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2nad139bd21004091311je4755c49r2ecbb025c7f49522@mail.gmail.com>

We had a lot of problems getting this to work properly for us (exact same
Mitel box), but the last time we looked at it was probably 12 months ago.
 We ended up with an asterisk box in between FS and the Mitel.  I would like
to know if you have more success than us - perhaps we can revisit this one.

Brent

On Sat, Apr 10, 2010 at 2:08 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From larclap at yahoo.com  Fri Apr  9 15:23:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 15:23:58 -0700 (PDT)
Subject: [Freeswitch-users] Getting git updates
Message-ID: <33344.52276.qm@web57613.mail.re1.yahoo.com>

Once we pull down the full FreeSWITCH version using git,
should we continue to use ?make current? to update thereafter?
 
I started with this command and interrupted it just after it
had pulled the source changes. I was surprised to see that ?fs_cli? no longer
worked. I went to freeswitch/bin and it was gone. Are the contents of this
directory removed in ?make current??
 
I also saw core files (core.pid?) in the freeswitch/bin
folder. Is it safe to remove these? They are about 200M each.
 
Thanks, Lars


From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Apr  9 10:22:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:22:07 -0700
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <g2s87f2f3b91004091022saa04aebp62482c2b6bbaea87@mail.gmail.com>

On Fri, Apr 9, 2010 at 10:07 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
I'm afraid so. :)

<node type="allow" cidr="10.10.3.10/32"/>
<node type="allow" cidr="10.10.3.11/32"/>

You're not allowing a domain in, you're allowing an IP address.
-MC
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From kevin at johnnyvoip.com  Fri Apr  9 14:33:33 2010
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 9 Apr 2010 17:33:33 -0400
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <y2xab57a3e81004091433h9ed905dbpb9a313b19e467f3d@mail.gmail.com>

Hi Kenny,

I've done this before. It's relatively straightforward if you are
experienced with the 3300.

You should simply need to setup a SIP trunk in the 3300 and point it to the
FS box. There are a few ways to deal with authentication depending on your
needs. You can setup ARS in the 3300 to push calls to the FS box, and from
the FS box you can makes calls across a gateway to the 3300 and out the the
PSTN.

If you have any questions please feel free to give me a buzz at the number
below.

Regards,
   Kevin Green

JohnnyVoIP
Cell:  613 866 0706
http://www.johnnyvoip.com


On Fri, Apr 9, 2010 at 12:08 PM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Apr 11 10:02:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:02:43 -0400
Subject: [Freeswitch-users] git clone via http
In-Reply-To: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
References: <x2s15b9404e1004080633h24e61d59v41f6eea658a7ba73@mail.gmail.com>
Message-ID: <E56DA82F-7BFF-47CE-B64F-34F001513443@jerris.com>

I am still working on this, should be soon.


On Apr 8, 2010, at 9:33 AM, mayamatakeshi wrote:

> Is there any chance of getting freeswitch using git thru http? 
> I need to go to internet thru a proxy so I cannot access it (unless I install some sort of git_proxy).



From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
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From bcxml at hotmail.com  Fri Apr  9 10:45:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Fri, 9 Apr 2010 09:45:10 -0800 (PST)
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <1270835110926-4878543.post@n2.nabble.com>


Here is the link

http://pastebin.freeswitch.org/12675


Brian
-- 
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4878543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mike at jerris.com  Sun Apr 11 10:05:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Apr 2010 13:05:52 -0400
Subject: [Freeswitch-users] version number: git checkout
In-Reply-To: <81019.71788.qm@web37501.mail.mud.yahoo.com>
References: <w2s33c87fa31004072354x2bbb5d68l23fd231f4b4c1c88@mail.gmail.com>
	<738759.37967.qm@web37505.mail.mud.yahoo.com>
	<s2m33c87fa31004080601j5b542168m702e868743999185@mail.gmail.com>
	<w2jc29b930b1004080651qdedc5402hfe96653eadfe6636@mail.gmail.com>
	<m2z33c87fa31004082048hc0513cc1z20bbbb958e81baad@mail.gmail.com>
	<81019.71788.qm@web37501.mail.mud.yahoo.com>
Message-ID: <B545758D-C2D6-4D6A-8C9C-D9D595787983@jerris.com>

If anyone is still having this issue, please open a bug for me on jira and provide privately via email information to remotely access the machine to troubleshoot.

Mike

On Apr 9, 2010, at 12:19 PM, DJB wrote:

> git pull 
> make all
> make install
> 
> -or-
> 
> make current
> 
> -djbinter
> 
> From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, April 8, 2010 8:48:17 PM
> Subject: Re: [Freeswitch-users] version number: git checkout
> 
> Thanks Milena... I upgraded git now - git version 1.7.0.4
> 
> I did a 'get pull && make install' and still the same problem.
> 
> Do I have to do a get clone or something?  Ideas?  Thanks
> 
> freeswitch:~# git --version
> git version 1.7.0.4
> freeswitch:~# fs_cli
>             _____ ____    ____ _    ___
>           |  ___/ ___|  / ___| |  |_ _|
>           | |_  \___ \  | |  | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|  |____/  \____|_____|___|
> 
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/  *
> *******************************************************
> 
> Type /help <enter> to see a list of commands
> 
> 
> +OK log level  [7]
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-)
> 
> On Thu, Apr 8, 2010 at 11:51 PM, Milena <testeador01 at gmail.com> wrote:
> >
> >
> > Can anything be done in the freeswitch code so when "git pull" is executed,
> > the "--pretty" argument is also set where "--format" is set to make it
> > compatible with both older and newer versions of git? or it is all up to
> > what git does and nothing to do on fs?
> >
> >
> >
> > PS: Mark, the issue you're facing is because of your version of git, the CLI
> > shows the freeswitch version properly with git 1.7.0.4, the "format"
> > argument isn't recognized by your version of git.
> >
> >
> > 2010/4/8 Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >>
> >> Git was installed as described on the wiki.  I am using Debian Lenny
> >> and Git version 1.5.6.5
> >>
> >> I just did a git pull and had the same issue... FS still shows
> >> FreeSWITCH Version 1.0.head (git-)
> >>
> >>
> >>
> >> On Thu, Apr 8, 2010 at 5:14 PM, DJB <djbinter at yahoo.com> wrote:
> >> > Upgrade your git, then it will show it correctly.
> >> > http://wiki.freeswitch.org/wiki/Git_Install
> >> > djbinter
> >> > ________________________________
> >> > From: Mark Campbell-Smith <mcampbellsmith at gmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Sent: Wed, April 7, 2010 11:54:27 PM
> >> > Subject: [Freeswitch-users] version number: git checkout
> >> >
> >> > Hi!
> >> >
> >> > I just used git for the first time ever to checkout FreeSwitch as
> >> > described on the wiki at
> >> > http://wiki.freeswitch.org/wiki/Installation_Guide
> >> >
> >> > Now my version number says:
> >> > FreeSWITCH Version 1.0.head (git-)
> >> >
> >> > Is there a mistake in my procedure or the building of FS when using
> >> > GIT?  Hard to know the build number of FS with a tag like that!
> >> >
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 lawwton at gmail.com  Sat Apr 10 08:51:39 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 10 Apr 2010 11:51:39 -0400
Subject: [Freeswitch-users] False Answer Supervision (FAS) - Question
Message-ID: <i2q5fe6fa8f1004100851y12792828zc405ed24cf00652c@mail.gmail.com>

All:

A while back I tried to solve a false answer supervision issue I was
intermittently receiving from underlying carriers. Back then I tried
to find a solution using asterisk but had other pending things and put
this off until now.

Does anyone have any recommendations on how to possibly handle or get
around FAS using FS. If I am not mistaken what would be needed is to
have something that upon receiving the first 200 message, it simply
ignores it, never propagating it and waits for the next 200. I can
control this now a little bit better since it's only happening when
the calls are sent to a couple of GrandStreams (FXO).

My idea is to perhaps write a little application and add it to the
DialPlan. The little script/app. would have the logic to ignore the
first 200 and hopefully get around this issue. Is that something that
is doable? Any other ideas on how to do this assuming is even possible
to do it?

Thanks in advance,

Alfredo


From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From woodydickson at gmail.com  Sat Apr 10 07:14:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 10 Apr 2010 22:14:52 +0800
Subject: [Freeswitch-users] performance comparison between centos and freebsd
Message-ID: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>

Hi,

I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.

The expectation is that freeswitch 1.0.5 acting as media proxy would
perform better in freebsd, but I found that freebsd can only sustain
half of the total concurrent calls as in centos 5.4 (120 vs 60).

The test is run on both ATOM CPU and VIA c7 and the result is
relatively the same.

Does anyone know why?  Is this some sort of setting issues in freebsd
kernel?  I have tried with pure freebsd and pfsense and the result is
the same.




Woody


From srinivas.ksvreddy at gmail.com  Fri Apr  9 12:40:58 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 10 Apr 2010 01:10:58 +0530
Subject: [Freeswitch-users] freeswitch1.0.2 compile errors
Message-ID: <n2nf8af5741004091240wf81d48fcm29d0fafd2f2bd9dc@mail.gmail.com>

HI,
we have customized Freeswitch.Managed project in Freeswitch1.0.2, so now i
cant go for 1.0.6, any there idea about compilation errors

-- 
Srinivasula Reddy K
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From lloyd.aloysius at gmail.com  Fri Apr  9 12:40:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 9 Apr 2010 15:40:38 -0400
Subject: [Freeswitch-users] NAT Problems
In-Reply-To: <C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
References: <s2l8a19bf2e1004081400y7bf40b2z2b50106db1f72c8a@mail.gmail.com>
	<23C88E92-10CB-49D4-8DE9-365F198D7CE1@gmail.com>
	<k2j8a19bf2e1004081829if332cff8ybbc3df1fbb230088@mail.gmail.com>
	<w2r8a19bf2e1004081831sfd01c080g6731a1b001c5c890@mail.gmail.com>
	<C8FE9E51-B929-4B9C-B2A1-1FD21B17E187@gmail.com>
Message-ID: <z2i8a19bf2e1004091240q72e174dbrbc3411fbcb6479b7@mail.gmail.com>

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/04/2010 ? 03:31, Aloysius Lloyd a ?crit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 ? 23:00, Aloysius Lloyd a ?crit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tjardick at vanderkraan.net  Sun Apr 11 14:35:34 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 11 Apr 2010 23:35:34 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>

Use the CIDR XML key not domain:

http://wiki.freeswitch.org/wiki/Acl

Regards,

Tj

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:

> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From ash at url.net.au  Sat Apr 10 03:50:43 2010
From: ash at url.net.au (Ash)
Date: Sat, 10 Apr 2010 20:50:43 +1000
Subject: [Freeswitch-users] Dynamic Conference with voicemail failover
Message-ID: <0B3EA4F7-9229-4A08-A89A-63071BDC421E@url.net.au>


Hi All,

I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way.

I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected.  I have this part working by using a dynamic bridge conference.  The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge.  This is the dialplan I am using:

      <extension name="9000">
        <condition field="destination_number" expression="^9000$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="continue_on_fail=true"/>
          <action application="set" data="ignore_early_media=true"/>

          <action application="answer"/>
          <action application="set" data="call_timeout=5"/>
          <action application="ring_ready"/>
          <action application="conference" data="bridge:_uuid_ at conf_1:sofia/gateway/primary/614xxxxxxxxx"/>
         <action application="voicemail" data="default v2e 1000"/>

        </condition>
      </extension>

What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call.  Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan?


Thanks in advance.

Ash.

From gavin.henry at gmail.com  Fri Apr  9 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 9 Apr 2010 23:47:01 +0100
Subject: [Freeswitch-users] SIP Trunk Mitel 3300
In-Reply-To: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
References: <q2w1c9955401004090908hd39da1c3j57da107fbc34726a@mail.gmail.com>
Message-ID: <x2h13ca621c1004091547h8d5b4042le449eec9c66fb8de@mail.gmail.com>

It works perfect. What have you tried?

Thanks,

Gavin.

On 09/04/2010, Kenneth Noisewater <noisewaterphd at gmail.com> wrote:
> Hi,
>
> I've already read the interop list, but I'm wondering if anyone on here has
> anymore experience/info on trunking freeswitch to a Mitel 3300?
>
> Specifically, I want to use freeswitch for acd and sip registrations, and
> just use our mitel for switching to the PSTN.
>
> Does anyone have some good info to share?
>
> Thanks,
>
> Kenny
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From noisewaterphd at gmail.com  Fri Apr  9 15:42:14 2010
From: noisewaterphd at gmail.com (Kenneth Noisewater)
Date: Fri, 9 Apr 2010 16:42:14 -0600
Subject: [Freeswitch-users] Multi-tenanting
Message-ID: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>

Hi All,

I'm just a few days into my FreeSwitch investigation, and so far I have to
say, it seems almost too good to be true! Kudos to everyone involved. I've
got  quite a bit of experience with other systems, ranging from Asterisk to
Mitel/Shoretel type systems, and FreeSwitch is really looking good.

Anyway...

So pouring over configs, it seems it would be really simple to set up a
multi tenant system with the whole 'domain' concept. Does this domain model
maintain good seperation throughout the system? Is there anything to be
aware of in doing a multi tenant setup?

Thanks,

Kenny
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From sean at obscuradigital.com  Sat Apr 10 13:19:13 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 10 Apr 2010 13:19:13 -0700
Subject: [Freeswitch-users] Mod directory
Message-ID: <C7E62B51.802E%sean@obscuradigital.com>

Hello,

Trying to figure out how to use the mod directory app within an ivr menu.
Not sure of the syntax to use when building the ivr.  Not ever sure it can
be done.

Also does mod_directory need to be added to a dialplan?

Thanks
Sean
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From larclap at yahoo.com  Fri Apr  9 14:05:10 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 9 Apr 2010 14:05:10 -0700
Subject: [Freeswitch-users] Getting git updates
Message-ID: <011a01cad828$573ce320$05b6a960$@com>

Once we pull down the full FreeSWITCH version using git, should we continue
to use 'make current' to update thereafter?

 

I started with this command and interrupted it just after it had pulled the
source changes. I was surprised to see that "fs_cli" no longer worked. I
went to freeswitch/bin and it was gone. Are the contents of this directory
removed in 'make current'?

 

I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe
to remove these? They are about 200M each.

 

Thanks, Lars

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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Fri Apr  9 16:51:10 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 10 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] phocketsphinx from lua
In-Reply-To: <1270833275502-4878370.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>,
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>,
	<4BB1A30A.4060701@gmail.com>,
	<1270796330100-4875677.post@n2.nabble.com>,
	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>,
	<1270833275502-4878370.post@n2.nabble.com>
Message-ID: <SNT135-w50CE48A6B89481F86FFE17D2150@phx.gbl>



hi, 

 

Does anyone have a sample of using phocketsphinx from lua?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From fraserredmond at gmail.com  Fri Apr  9 12:13:05 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 9 Apr 2010 20:13:05 +0100
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <012d01cad807$27f5a0a0$77e0e1e0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
Message-ID: <i2nc32ee1cf1004091213r4797c250j4020914c6c8eedbb@mail.gmail.com>

I think you'll want to change it to:

    <list name="domains" default="deny">

      <node type="allow" cidr ="10.10.3.10/32"/>

      <node type="allow" cidr ="10.10.3.11/32"/>

    </list>


Cheers,
Fraser




On Fri, Apr 9, 2010 at 6:07 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 brian at freeswitch.org  Tue Apr 13 06:20:51 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Apr 2010 08:20:51 -0500
Subject: [Freeswitch-users] mail queue
Message-ID: <EB492B54-F3C9-41D1-AFA0-F9DE59F4A87A@freeswitch.org>

I dumped the mail queue this morning in hopes it would clear up anyone getting duplicates.  The log file reached 2 gigs and the MTA decided to just exit since the OS wouldn't let it open the logfile for append over the weekend.  Which caused this problem in the first place.

Sorry for any trouble.

Thanks,
Brian



From scaram at hotmail.de  Tue Apr 13 06:53:44 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Tue, 13 Apr 2010 15:53:44 +0200
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
Message-ID: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>


hi , I have a problem with long duration calls. After 11 hours and 35 minutes the call ends because of MEDIA_TIMEOUT. It seems to be reproduceable. 
Any idea? I use SIP phone SIEMENS Gigaset S685. 

2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/1010 at 192.168.1.234 [BREAK]
2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/1010 at 192.168.1.234] has been answered
2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182 sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/1010 at 192.168.1.234 [KILL]
2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1010 at 192.168.1.234 [BREAK]
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1010 at 192.168.1.234) State HANGUP
2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/internal/1010 at 192.168.1.234
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP -> CS_REPORTING
2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1010 at 192.168.1.234 [BREAK]
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/1010 at 192.168.1.234) State REPORTING
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause: MEDIA_TIMEOUT
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING -> CS_DESTROY
2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3 (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/1010 at 192.168.1.234) Ended
2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore original codec.
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/1010 at 192.168.1.234) State DESTROY
2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255 sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1010 at 192.168.1.234 Standard DESTROY
2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
 		 	   		  
_________________________________________________________________
http://redirect.gimas.net/?n=M1004xSkyDrive2
Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
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From steveayre at gmail.com  Tue Apr 13 08:23:44 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 13 Apr 2010 16:23:44 +0100
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
Message-ID: <u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>

Which version of FS are you running? From the line numbers it isn't
the latest, so you might try updating and see if you can still
reproduce.

Given that the hangup appears to happen on mod_sofia.c:744, there is a
MEDIA_TIMEOUT near that line which is triggered if FS times out trying
to read a video frame from RTP. That will occur when the RTP stack
thinks it has missed too many RTP packets.

Do you have an RTP packet trace for the call?

-Steve



On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
> hi , I have a problem with long duration calls. After 11 hours and 35
> minutes the call ends because of MEDIA_TIMEOUT. It seems to be
> reproduceable.
> Any idea? I use SIP phone SIEMENS Gigaset S685.
>
> 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
> sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
> 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/1010 at 192.168.1.234 [BREAK]
> 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
> [sofia/internal/1010 at 192.168.1.234] has been answered
> 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
> sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
> EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
> 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
> sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
> 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
> sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
> 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
> sofia/internal/1010 at 192.168.1.234 [KILL]
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1010 at 192.168.1.234 [BREAK]
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/1010 at 192.168.1.234) State HANGUP
> 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
> 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
> sofia/internal/1010 at 192.168.1.234
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
> (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP -> CS_REPORTING
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1010 at 192.168.1.234 [BREAK]
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> (sofia/internal/1010 at 192.168.1.234) State REPORTING
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause: MEDIA_TIMEOUT
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
> (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING -> CS_DESTROY
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3
> (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
> 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3
> (sofia/internal/1010 at 192.168.1.234) Ended
> 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore original
> codec.
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> (sofia/internal/1010 at 192.168.1.234) State DESTROY
> 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
> sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1010 at 192.168.1.234 Standard DESTROY
> 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
>
> ________________________________
> Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 13 08:53:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Apr 2010 10:53:48 -0500
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <20100413105326.GY12939@tamay-dogan.net>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
	<4BC44A68.9040209@telemaque.fr>
	<20100413105326.GY12939@tamay-dogan.net>
Message-ID: <y2r191c3a031004130853u3bd2cafejeac02d85c17dbe80@mail.gmail.com>

You guys talk too much =D

The log file reached 2gb and even in 2010 in the age of multiple terabyte
media on a 64 bit box it somehow still is a problem with the file size
causing software to break.

The problem has been fixed.



On Tue, Apr 13, 2010 at 5:53 AM, Michelle Konzack <
linux4michelle at tamay-dogan.net> wrote:

> Hello Tristan and "postmaster of <ostag.org>",
>
> Am 2010-04-13 12:41:44, hacktest Du folgendes herunter:
> > Hi Michelle,
> >
> > We own our mailserver there, and I can assure it comes from FS ML
> > server. You'll be able to see that from the headers below.
> >
> > Here are two headers from mails I received a few minutes ago:
> <snip>
>
> I have checked my mailserver too and found arround 1740 dupes...  which
> where detected by my spamfilers as dulicated.
>
> It seems there is a problem on <yoda.ostag.org>, because  the  first  or
> second received header are intact, but it changed after it was  received
> by <yoda.ostag.org>.
>
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>
> --
> ##################### Debian GNU/Linux Consultant ######################
>   Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
> Gesch. Michelle Konzack          Gesch. Michelle Konzack
>
> Apt. 917 (homeoffice)
> 50, rue de Soultz               Kinzigstra?e 17
> 67100 Strasbourg/France         77694 Kehl/Germany
> Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
> Tel: +33-9-52705884 fix
>
> <http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ    #328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Tue Apr 13 09:26:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Apr 2010 12:26:17 -0400
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <4BC42232.5070907@gmail.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>	<COL117-W71213F34122C75703A37CB9120@phx.gbl>	<4BC33F0F.8070208@gmail.com>
	<087B501F-8809-4CD7-A426-42F39D7568A9@jerris.com>
	<4BC42232.5070907@gmail.com>
Message-ID: <6CAC623D-40BE-4125-A183-2AC4F651599C@jerris.com>

iirc, the suggestion from msft is to always include the dll in your lib dir, NOT install it on the base system with the redist.

Mike

On Apr 13, 2010, at 3:50 AM, Tamas wrote:

> 
> Michael Jerris ?rta:
>> 
>> 
>> On Apr 12, 2010, at 11:41 AM, Tamas wrote:
>> 
>>> Is it possible to link against msvcrt.dll instead of compiler specific ones (e.g. msvcrt90.dll)?
>> 
>> Maybe, but I have not seen an easy way to do this with later compilers, and the old ones are generally pretty broken.  I doubt it would compile at all.
> I see. I've read several stories about good and bad results linking against msvcrt.dll. Some big players do that, though (like mozilla - checked xulrunner).
>> 
>>> In case we don't use any console stuff, is there a way to link without msvcrt at all?
>> 
>> Not sure the connection between console and msvcrt.dll.  We need a runtime regardless unless we somehow write c code that uses no functions from the c runtime.
>> 
>> Why is it an issue to depend on msvcrt90.dll?
> It is a minor thing. For a FS based softphone client you have to handle installation of these dll files too. Right now using vcredist_x86.exe seems to be the right way, but in such a case windows will have another package the user might wonder how it get there - or can delete later when does not know why is it there. This is just from maximalistic point of view :)
> 
>> 
>>> 
>>> Regards,
>>>     Tamas
>>> 
>>> Jeff Lenk ?rta:
>>>> 
>>>> Sure the only requirement is the 2008 CRuntime support. No files or registry settings needed for basic operation.
>>>>  
>>>> Date: Mon, 12 Apr 2010 13:22:49 +0100
>>>> From: vfclists at googlemail.com
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
>>>> 
>>>> 
>>>> Can Freeswitch be installed simply by zipping up the folder and unzipping it to the destination?
>>>> 
>>>> Does it require some DLLs to be installed in the Windows system folder and some registry entries as well?

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From fraserredmond at gmail.com  Tue Apr 13 09:52:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Tue, 13 Apr 2010 17:52:19 +0100
Subject: [Freeswitch-users] media_bug_answer_req ignored if aleg is already
	answered.
Message-ID: <t2tc32ee1cf1004130952zda85c9f5z5c05197a7d232916@mail.gmail.com>

If you have a dialplan which answers the incoming call, runs through some
IVR's/javascript/etc, then bridges a call to another ext, the variable
media_bug_answer_req is ignored.

media_bug_answer_req  stops the ringtone from being included in a session
recording, specifically it tells the recording to not start until after it
is answered - but it is basing that on either leg being answered (or at
least the current leg), rather than on both legs (or the b-leg) being
answered.

Is that something I should log in Jira, or is there something I'm missing?

(I tried running the record_session on the b-leg via execute_on_answer, but
then I couldn't find a way to get it to stop recording, so the audio file
was never created.)

Cheers,
Fraser
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From scaram at hotmail.de  Tue Apr 13 10:03:55 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Tue, 13 Apr 2010 19:03:55 +0200
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>,
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
Message-ID: <SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>


I am running the last stable release 1.0.4 and could reproduce the failure. Sometimes it happens in a call of 4 minutes duration.

I don't have a rtp packet trace but there is one more reason for MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls down to the winsock function recvfrom. It also could be a socket error.  If the recvfrom function returns -2 a MEDIA_TIMEOUT will happen. Maybe something was wrong with the network hardware cables, switch ..
I changed my hardware. Test is running...

How can I enable a rtp packet trace? It think I should check this too.


> Date: Tue, 13 Apr 2010 16:23:44 +0100
> From: steveayre at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
> 
> Which version of FS are you running? From the line numbers it isn't
> the latest, so you might try updating and see if you can still
> reproduce.
> 
> Given that the hangup appears to happen on mod_sofia.c:744, there is a
> MEDIA_TIMEOUT near that line which is triggered if FS times out trying
> to read a video frame from RTP. That will occur when the RTP stack
> thinks it has missed too many RTP packets.
> 
> Do you have an RTP packet trace for the call?
> 
> -Steve
> 
> 
> 
> On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
> > hi , I have a problem with long duration calls. After 11 hours and 35
> > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
> > reproduceable.
> > Any idea? I use SIP phone SIEMENS Gigaset S685.
> >
> > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
> > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
> > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal
> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
> > [sofia/internal/1010 at 192.168.1.234] has been answered
> > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
> > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
> > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
> > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
> > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
> > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
> > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
> > sofia/internal/1010 at 192.168.1.234 [KILL]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
> > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > (sofia/internal/1010 at 192.168.1.234) State HANGUP
> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
> > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
> > sofia/internal/1010 at 192.168.1.234
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
> > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
> > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP -> CS_REPORTING
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > (sofia/internal/1010 at 192.168.1.234) State REPORTING
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
> > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause: MEDIA_TIMEOUT
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
> > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING -> CS_DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3
> > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3
> > (sofia/internal/1010 at 192.168.1.234) Ended
> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close Channel
> > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore original
> > codec.
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > (sofia/internal/1010 at 192.168.1.234) State DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
> > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
> > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
> >
> > ________________________________
> > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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://redirect.gimas.net/?n=M1004xSkyDrive2
Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
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From anthony.minessale at gmail.com  Tue Apr 13 10:07:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Apr 2010 12:07:14 -0500
Subject: [Freeswitch-users] media_bug_answer_req ignored if aleg is
	already answered.
In-Reply-To: <t2tc32ee1cf1004130952zda85c9f5z5c05197a7d232916@mail.gmail.com>
References: <t2tc32ee1cf1004130952zda85c9f5z5c05197a7d232916@mail.gmail.com>
Message-ID: <o2v191c3a031004131007p642feb2ek24daa849e2e054c3@mail.gmail.com>

Doesn't your subject line say it all?
If it's already answered, how is it going to wait for it to be answered
before it starts recording?

Indeed, the variable applies only to the one session you set it on.
Remember, a call is a combination of 2 legs and each leg has its own
independent settings.

I am not sure what getting it to stop has to do with the file being created
so I think your attempt to make it work on the b-leg in your case was the
right direction but you must have done something wrong because the file is
created instantly when you start recording and will stop when the call hangs
up.


On Tue, Apr 13, 2010 at 11:52 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> If you have a dialplan which answers the incoming call, runs through some
> IVR's/javascript/etc, then bridges a call to another ext, the variable
> media_bug_answer_req is ignored.
>
> media_bug_answer_req  stops the ringtone from being included in a session
> recording, specifically it tells the recording to not start until after it
> is answered - but it is basing that on either leg being answered (or at
> least the current leg), rather than on both legs (or the b-leg) being
> answered.
>
> Is that something I should log in Jira, or is there something I'm missing?
>
> (I tried running the record_session on the b-leg via execute_on_answer, but
> then I couldn't find a way to get it to stop recording, so the audio file
> was never created.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From jalsot at gmail.com  Tue Apr 13 10:13:05 2010
From: jalsot at gmail.com (Tamas)
Date: Tue, 13 Apr 2010 19:13:05 +0200
Subject: [Freeswitch-users] Embedded Freeswitch Installation for Windows?
In-Reply-To: <6CAC623D-40BE-4125-A183-2AC4F651599C@jerris.com>
References: <g2y7abab2411004120522yefd7d21er72b6795e5c99169d@mail.gmail.com>	<COL117-W71213F34122C75703A37CB9120@phx.gbl>	<4BC33F0F.8070208@gmail.com>	<087B501F-8809-4CD7-A426-42F39D7568A9@jerris.com>	<4BC42232.5070907@gmail.com>
	<6CAC623D-40BE-4125-A183-2AC4F651599C@jerris.com>
Message-ID: <4BC4A621.4000907@gmail.com>

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From anthony.minessale at gmail.com  Tue Apr 13 10:19:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Apr 2010 12:19:10 -0500
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
Message-ID: <x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>

The last stable release is actually 1.0.6


On Tue, Apr 13, 2010 at 12:03 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  I am running the last stable release 1.0.4 and could reproduce the
> failure. Sometimes it happens in a call of 4 minutes duration.
>
> I don't have a rtp packet trace but there is one more reason for
> MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls down to
> the winsock function recvfrom. It also could be a socket error.  If the
> recvfrom function returns -2 a MEDIA_TIMEOUT will happen. Maybe something
> was wrong with the network hardware cables, switch ..
> I changed my hardware. Test is running...
>
> How can I enable a rtp packet trace? It think I should check this too.
>
>
> > Date: Tue, 13 Apr 2010 16:23:44 +0100
> > From: steveayre at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
>
> >
> > Which version of FS are you running? From the line numbers it isn't
> > the latest, so you might try updating and see if you can still
> > reproduce.
> >
> > Given that the hangup appears to happen on mod_sofia.c:744, there is a
> > MEDIA_TIMEOUT near that line which is triggered if FS times out trying
> > to read a video frame from RTP. That will occur when the RTP stack
> > thinks it has missed too many RTP packets.
> >
> > Do you have an RTP packet trace for the call?
> >
> > -Steve
> >
> >
> >
> > On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
> > > hi , I have a problem with long duration calls. After 11 hours and 35
> > > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
> > > reproduceable.
> > > Any idea? I use SIP phone SIEMENS Gigaset S685.
> > >
> > > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
> > > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
> > > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send
> signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
> > > [sofia/internal/1010 at 192.168.1.234] has been answered
> > > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
> > > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
> > > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
> > > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
> > > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
> > > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
> > > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
> > > sofia/internal/1010 at 192.168.1.234 [KILL]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send
> signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
> > > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > > (sofia/internal/1010 at 192.168.1.234) State HANGUP
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
> > > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
> > > sofia/internal/1010 at 192.168.1.234
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
> > > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause:
> MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
> > > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP ->
> CS_REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send
> signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > > (sofia/internal/1010 at 192.168.1.234) State REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
> > > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause:
> MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
> > > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING ->
> CS_DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3
> > > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external
> entities
> > > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session
> 3
> > > (sofia/internal/1010 at 192.168.1.234) Ended
> > > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close
> Channel
> > > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore
> original
> > > codec.
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > > (sofia/internal/1010 at 192.168.1.234) State DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
> > > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
> > > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
> > >
> > > ________________________________
> > > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!<http://redirect.gimas.net/?n=M1004SkyDrive>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From pankajanand18 at gmail.com  Tue Apr 13 10:47:40 2010
From: pankajanand18 at gmail.com (pankaj anand)
Date: Tue, 13 Apr 2010 23:17:40 +0530
Subject: [Freeswitch-users] Reinvite time
Message-ID: <u2k809ad7ab1004131047z7f4ae7a0kdb353207bfdecc15@mail.gmail.com>

hi,
  Freeswitch sends invite message after every 60 seconds as a keep alive
signal. Can the time 60 seconds can be increased or removed completely.


any help would appreciated.

with regards
Pankaj anand
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From msc at freeswitch.org  Tue Apr 13 13:46:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Apr 2010 13:46:26 -0700
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC3B23F.5070700@codefix.net>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
Message-ID: <u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>

On Mon, Apr 12, 2010 at 4:52 PM, Garrison Hoffman <garrison at codefix.net>wrote:

> I got annoyed at the thought of using the application API to accomplish
> something already built-in, so a bit of wading through the source gave
> up the answer:
>
> <entry action="menu-play-sound" digits="3" param="say: You pressed 3"/>
>
> 'Twould be nice if someone updated demo_ivr.xml with the correct syntax.
>
> -gh
>

FYI,the wiki had the correct syntax all along:
http://wiki.freeswitch.org/wiki/IVR_Menu

However, I just updated demo_ivr.xml so that it has the correct syntax, too.
Thanks for digging.
-MC
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From stevendt at primrosebank.net  Tue Apr 13 14:19:14 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 13 Apr 2010 22:19:14 +0100
Subject: [Freeswitch-users] Cisco 7911G
Message-ID: <73733B3332CD4C9B883A79E0A1A46CE5@bp1.ad.bp.com>

Hi,

does anyone have a working config file to use a Cisco 7911G with FreeSWITCH please ?

The SEP<MAC>.CNF.XML file is significantly different to the 7940 format and I can't get the phone to register with FreeSWITCH, if anyone has a working version, I'd appreciate a copy please ?

regards
Dave
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From kward at binarysignal.com  Tue Apr 13 14:11:07 2010
From: kward at binarysignal.com (Kurt Ward)
Date: Tue, 13 Apr 2010 14:11:07 -0700
Subject: [Freeswitch-users] OT: tool recomendations
In-Reply-To: <u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
Message-ID: <9381E74F-DC70-4D38-ADBF-48519470956D@binarysignal.com>

I've been toying with FS for the last couple of weeks and am going to  
be using it on a couple of software/hardware projects.  In the process  
of evaluating FS, I needed a few things that would allow me to do some  
testing without writing much code.  One of my favorites (for OS X  
anyhow) is this simple yet powerful XML-RPC client:

http://ditchnet.org/xmlrpc/

(Note this guy also has similar tools, and HTTP client and a SOAP  
client).  I'm curious what other users use to play/prototype/test with?


From brian at freeswitch.org  Tue Apr 13 14:45:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Apr 2010 16:45:06 -0500
Subject: [Freeswitch-users] OT: tool recomendations
In-Reply-To: <9381E74F-DC70-4D38-ADBF-48519470956D@binarysignal.com>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
	<9381E74F-DC70-4D38-ADBF-48519470956D@binarysignal.com>
Message-ID: <DEE03C3C-4C6B-492D-9594-2AE58DF1B656@freeswitch.org>

Kurt,
	Thanks for the info that should go on the wiki also.  Also in the future can you please not hijack threads.  You clicked reply to the message with a subject of switch_ivr_action_t, changed the subject and deleted the body and sent your message.  This causes some things including the mailing list archive to thread improperly.

Thanks,
Brian

On Apr 13, 2010, at 4:11 PM, Kurt Ward wrote:

> I've been toying with FS for the last couple of weeks and am going to  
> be using it on a couple of software/hardware projects.  In the process  
> of evaluating FS, I needed a few things that would allow me to do some  
> testing without writing much code.  One of my favorites (for OS X  
> anyhow) is this simple yet powerful XML-RPC client:
> 
> http://ditchnet.org/xmlrpc/
> 
> (Note this guy also has similar tools, and HTTP client and a SOAP  
> client).  I'm curious what other users use to play/prototype/test with?

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From douga at cachecomm.com  Tue Apr 13 16:02:05 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Tue, 13 Apr 2010 17:02:05 -0600
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
Message-ID: <4BC4F7ED.50406@cachecomm.com>

Is anyone having success using 2B Channel Transfer with FreeSWITCH and 
Sangoma?

If so, what are the driver considerations and what does it take to make 
the transfer work?

Thanks,
Doug


From mrene_lists at avgs.ca  Tue Apr 13 16:33:53 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 13 Apr 2010 19:33:53 -0400
Subject: [Freeswitch-users] 100& LSB compliant FS init script
In-Reply-To: <7FE4080A46DF4F57820DF3587C71E550@MOBILEE1705>
References: <7FE4080A46DF4F57820DF3587C71E550@MOBILEE1705>
Message-ID: <F21937CD-C1C7-45B4-9CAE-B5D0FA64B9FE@avgs.ca>

Hi,

Thanks for the contribution. Please note that the proper place to post patches (or new files) and to report bugs is on http://jira.freeswitch.org/

Also, why are you starting freeswitch with -waste?
 
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-13, at 12:01 AM, Madovsky wrote:

> <freeswitch>



From marketing at cluecon.com  Tue Apr 13 16:24:24 2010
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 13 Apr 2010 16:24:24 -0700
Subject: [Freeswitch-users] ClueCon MMX - Update
Message-ID: <u2w87f2f3b91004131624j2f6eb63frc516afb4e402ee91@mail.gmail.com>

Greetings,

We would like to let everyone know that ClueCon MMX preparations are moving
forward. New sponsors have signed up and we are starting to fill up our
speaker slots. Stay tuned for more information.

In the meantime, if you have not already signed up for ClueCon then please
call 877.742.CLUE (2583) and get registered right away!

Looking forward to seeing you in Chicago,
The ClueCon Team
877.742.CLUE
http://www.cluecon.com
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From garrison at codefix.net  Tue Apr 13 18:00:06 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Tue, 13 Apr 2010 21:00:06 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
Message-ID: <4BC51396.70404@codefix.net>

Michael Collins wrote:

> FYI,the wiki had the correct syntax all along:
> http://wiki.freeswitch.org/wiki/IVR_Menu

That is *so* untrue! The wiki says:

   * menu-say-text - Speak a TTS prompt

Which is what gave the error in my original post.


From mike at jerris.com  Tue Apr 13 20:51:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Apr 2010 23:51:39 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC51396.70404@codefix.net>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
	<4BC51396.70404@codefix.net>
Message-ID: <0D363F7C-68B5-467C-87D4-9782ABD32D1A@jerris.com>

would you mind updating the wiki page with the correct information?

On Apr 13, 2010, at 9:00 PM, Garrison Hoffman wrote:

> Michael Collins wrote:
> 
>> FYI,the wiki had the correct syntax all along:
>> http://wiki.freeswitch.org/wiki/IVR_Menu
> 
> That is *so* untrue! The wiki says:
> 
>   * menu-say-text - Speak a TTS prompt
> 
> Which is what gave the error in my original post.
> 



From mike at jerris.com  Tue Apr 13 20:53:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Apr 2010 23:53:27 -0400
Subject: [Freeswitch-users] Reinvite time
In-Reply-To: <u2k809ad7ab1004131047z7f4ae7a0kdb353207bfdecc15@mail.gmail.com>
References: <u2k809ad7ab1004131047z7f4ae7a0kdb353207bfdecc15@mail.gmail.com>
Message-ID: <D9515702-F7F4-45C1-B30B-F6C19D3ACB2F@jerris.com>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout

On Apr 13, 2010, at 1:47 PM, pankaj anand wrote:

> hi, 
>   Freeswitch sends invite message after every 60 seconds as a keep alive signal. Can the time 60 seconds can be increased or removed completely. 
> 
> 
> any help would appreciated. 
> 
> with regards
> Pankaj anand

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From garrison at codefix.net  Tue Apr 13 22:03:52 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Wed, 14 Apr 2010 01:03:52 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <0D363F7C-68B5-467C-87D4-9782ABD32D1A@jerris.com>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
	<4BC51396.70404@codefix.net>
	<0D363F7C-68B5-467C-87D4-9782ABD32D1A@jerris.com>
Message-ID: <4BC54CB8.60809@codefix.net>

Michael Jerris wrote:
> would you mind updating the wiki page with the correct information?

No I don't mind, presumably I need to register a wiki account to do so? 
(OpenID would be nice)

I'm new around here so I'm still figuring out what's customary. 
Incidentally, the most pressing FS issue I have is described in a Dial 
Group question I posted on Monday, but it seems to have been overlooked, 
is it cool to repost it after a couple days?

-gh


From garrison at codefix.net  Tue Apr 13 22:40:12 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Wed, 14 Apr 2010 01:40:12 -0400
Subject: [Freeswitch-users] Dial Group & Handy-Tone 386
In-Reply-To: <4BC34D7B.7000606@codefix.net>
References: <4BC34D7B.7000606@codefix.net>
Message-ID: <4BC5553C.9090109@codefix.net>

Adding the following seems to have fixed my issue:

   <action application="answer"/>

But I'm still a bit foggy on the details. Specifically I'd like to know 
more about "answer" and if there is anything in the log which indicates 
a call being picked up in early media. (it seems counter intuitive)

Presumably it's not safe to always insert the above, otherwise would it 
not be the default behavior?

-gh


From garrison at codefix.net  Tue Apr 13 22:43:32 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Wed, 14 Apr 2010 01:43:32 -0400
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC54CB8.60809@codefix.net>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
	<4BC51396.70404@codefix.net>
	<0D363F7C-68B5-467C-87D4-9782ABD32D1A@jerris.com>
	<4BC54CB8.60809@codefix.net>
Message-ID: <4BC55604.1010004@codefix.net>

Garrison Hoffman wrote:
> Michael Jerris wrote:
>> would you mind updating the wiki page with the correct information?
> 
> No I don't mind, presumably I need to register a wiki account to do so? 
> (OpenID would be nice)
> 
> I'm new around here so I'm still figuring out what's customary. 
> Incidentally, the most pressing FS issue I have is described in a Dial 
> Group question I posted on Monday, but it seems to have been overlooked, 
> is it cool to repost it after a couple days?

How ironic that part of the answer to my other question was on that very 
page.  Or maybe knew that you did, Master Yoda  ;-)

-gh


From mike at jerris.com  Tue Apr 13 22:58:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 01:58:23 -0400
Subject: [Freeswitch-users] Dial Group & Handy-Tone 386
In-Reply-To: <4BC5553C.9090109@codefix.net>
References: <4BC34D7B.7000606@codefix.net> <4BC5553C.9090109@codefix.net>
Message-ID: <44FE0C3D-B508-4A55-8769-4EF66B6C41C7@jerris.com>

If this "fixed" your issue, it sounds like some issue getting the real answer indication from the answering phone.  I would enable sip trace and see if we are actually getting the 200 from the phone or not.

Mike

On Apr 14, 2010, at 1:40 AM, Garrison Hoffman wrote:

> Adding the following seems to have fixed my issue:
> 
>   <action application="answer"/>
> 
> But I'm still a bit foggy on the details. Specifically I'd like to know 
> more about "answer" and if there is anything in the log which indicates 
> a call being picked up in early media. (it seems counter intuitive)
> 
> Presumably it's not safe to always insert the above, otherwise would it 
> not be the default behavior?
> 
> -gh



From msc at freeswitch.org  Tue Apr 13 23:27:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Apr 2010 23:27:27 -0700
Subject: [Freeswitch-users] switch_ivr_action_t
In-Reply-To: <4BC51396.70404@codefix.net>
References: <4BC361BA.504@codefix.net> <4BC3B23F.5070700@codefix.net>
	<u2u87f2f3b91004131346h625105eahfeec4c96d5ed2296@mail.gmail.com>
	<4BC51396.70404@codefix.net>
Message-ID: <m2w87f2f3b91004132327rae1232b6pbfbb7d51bcc1911c@mail.gmail.com>

Mea culpa - I thought you were talking about playing a file not TTS. I see
you updated the wiki. Thanks! You did a good job on that edit. Feel free to
add more knowledge any time.
-MC

On Tue, Apr 13, 2010 at 6:00 PM, Garrison Hoffman <garrison at codefix.net>wrote:

> Michael Collins wrote:
>
> > FYI,the wiki had the correct syntax all along:
> > http://wiki.freeswitch.org/wiki/IVR_Menu
>
> That is *so* untrue! The wiki says:
>
>   * menu-say-text - Speak a TTS prompt
>
> Which is what gave the error in my original post.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 13 23:33:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Apr 2010 23:33:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call
Message-ID: <q2g87f2f3b91004132333r3d6a463cx94b4f6bafcc65ce@mail.gmail.com>

Hello all,

Our agenda for tomorrow is extremely light:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_14

No one was available for a formal discussion so I thought maybe we could
have an open discussion instead. If you have any agenda items please add
them.

Thanks!

-MC
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From jonas.gauffin at gmail.com  Tue Apr 13 23:45:50 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 08:45:50 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
Message-ID: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>

Hello,

I got a problem with Snom phones where they report a different port than the
one provisioned. I tried to talk with the snom support without success.

REGISTER sip:phone.gateon.se:5070 SIP/2.0
Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK-u1y0v7is9ix0;rport
From: "Helene XXX" <sip:u1000028 at phone.xxxx.se:5070>;tag=kjaa26j1qv
To: "Helene xxx" <sip:u1000028 at phone.xxxx.se:5070>
Call-ID: 3c267019ee2d-ni5vq3v88za6
CSeq: 133 REGISTER
Max-Forwards: 70
Contact: <sip:u1000028 at 217.31.xxx.xxx
:26062;line=yhkplirb>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d-ad61-2be3d1a88425>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom360/7.3.30
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.3.109
Expires: 3600
Content-Length: 0
The port that I provisioned (and I've doublechecked in the webinterface) is
5071 and not 26062.

Have anyone else got this problem and solved it?

Regards,
  Jonas
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From steveayre at gmail.com  Wed Apr 14 00:05:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 14 Apr 2010 08:05:59 +0100
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
Message-ID: <g2weafa772f1004140005ya45bc3f7p7cbd51a5e515b030@mail.gmail.com>

"How can I enable a rtp packet trace? It think I should check this too."

http://www.wireshark.org/


On 13 April 2010 18:03, Francisco Scaramanga <scaram at hotmail.de> wrote:
> I am running the last stable release 1.0.4 and could reproduce the failure.
> Sometimes it happens in a call of 4 minutes duration.
>
> I don't have a rtp packet trace but there is one more reason for
> MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls down to
> the winsock function recvfrom. It also could be a socket error.? If the
> recvfrom function returns -2 a MEDIA_TIMEOUT will happen. Maybe something
> was wrong with the network hardware cables, switch ..
> I changed my hardware. Test is running...
>
> How can I enable a rtp packet trace? It think I should check this too.
>
>
>> Date: Tue, 13 Apr 2010 16:23:44 +0100
>> From: steveayre at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
>>
>> Which version of FS are you running? From the line numbers it isn't
>> the latest, so you might try updating and see if you can still
>> reproduce.
>>
>> Given that the hangup appears to happen on mod_sofia.c:744, there is a
>> MEDIA_TIMEOUT near that line which is triggered if FS times out trying
>> to read a video frame from RTP. That will occur when the RTP stack
>> thinks it has missed too many RTP packets.
>>
>> Do you have an RTP packet trace for the call?
>>
>> -Steve
>>
>>
>>
>> On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
>> > hi , I have a problem with long duration calls. After 11 hours and 35
>> > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
>> > reproduceable.
>> > Any idea? I use SIP phone SIEMENS Gigaset S685.
>> >
>> > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
>> > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
>> > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
>> > [sofia/internal/1010 at 192.168.1.234] has been answered
>> > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
>> > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
>> > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
>> > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
>> > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
>> > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
>> > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [KILL]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
>> > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
>> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
>> > (sofia/internal/1010 at 192.168.1.234) State HANGUP
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
>> > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
>> > sofia/internal/1010 at 192.168.1.234
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
>> > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
>> > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
>> > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP ->
>> > CS_REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
>> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
>> > (sofia/internal/1010 at 192.168.1.234) State REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
>> > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause:
>> > MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
>> > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
>> > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING ->
>> > CS_DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3
>> > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
>> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3
>> > (sofia/internal/1010 at 192.168.1.234) Ended
>> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close
>> > Channel
>> > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore
>> > original
>> > codec.
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
>> > (sofia/internal/1010 at 192.168.1.234) State DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
>> > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
>> > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
>> > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
>> >
>> > ________________________________
>> > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From math.parent at gmail.com  Wed Apr 14 00:20:38 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Wed, 14 Apr 2010 09:20:38 +0200
Subject: [Freeswitch-users] Cisco 7911G
In-Reply-To: <73733B3332CD4C9B883A79E0A1A46CE5@bp1.ad.bp.com>
References: <73733B3332CD4C9B883A79E0A1A46CE5@bp1.ad.bp.com>
Message-ID: <p2y960738411004140020yab91c0aai54c2b88d4c01c881@mail.gmail.com>

On Tue, Apr 13, 2010 at 11:19 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
>
> Hi,
Hi,

>
> does anyone have a working config file to use a Cisco 7911G with FreeSWITCH please ?
>
> The SEP<MAC>.CNF.XML file is significantly different to the 7940 format and I can't get the phone to register with FreeSWITCH, if anyone has a working version, I'd appreciate a copy please ?

Are you using SIP or SCCP? Can you test mod_skinny (from trunk)?

I don't have 7911G. But as a general note, you can search on
<http://www.voip-info.org/>: plenty of examples for Asterisk can be
adapted for FS.

Regards
Mathieu Parent


From scaram at hotmail.de  Wed Apr 14 01:00:33 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Wed, 14 Apr 2010 10:00:33 +0200
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>,
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>,
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>,
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
Message-ID: <SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>



Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
I thought it is a release candidate, isn't it? Is it safe to use it in production?


BTW MEDIA_TIMEOUT didn't occur since 16 hours, so I think the failure was a defective contact..

 

Date: Tue, 13 Apr 2010 12:19:10 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call

The last stable release is actually 1.0.6


On Tue, Apr 13, 2010 at 12:03 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:






I am running the last stable release 1.0.4 and could reproduce the failure. Sometimes it happens in a call of 4 minutes duration.

I don't have a rtp packet trace but there is one more reason for MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls down to the winsock function recvfrom. It also could be a socket error.  If the recvfrom function returns -2 a MEDIA_TIMEOUT will happen. Maybe something was wrong with the network hardware cables, switch ..

I changed my hardware. Test is running...

How can I enable a rtp packet trace? It think I should check this too.


> Date: Tue, 13 Apr 2010 16:23:44 +0100
> From: steveayre at gmail.com

> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call

> 
> Which version of FS are you running? From the line numbers it isn't
> the latest, so you might try updating and see if you can still
> reproduce.
> 
> Given that the hangup appears to happen on mod_sofia.c:744, there is a

> MEDIA_TIMEOUT near that line which is triggered if FS times out trying
> to read a video frame from RTP. That will occur when the RTP stack
> thinks it has missed too many RTP packets.
> 
> Do you have an RTP packet trace for the call?

> 
> -Steve
> 
> 
> 
> On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
> > hi , I have a problem with long duration calls. After 11 hours and 35

> > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
> > reproduceable.
> > Any idea? I use SIP phone SIEMENS Gigaset S685.
> >
> > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel

> > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
> > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal

> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
> > [sofia/internal/1010 at 192.168.1.234] has been answered

> > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
> > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
> > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul

> > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
> > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
> > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup

> > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
> > sofia/internal/1010 at 192.168.1.234 [KILL]

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491

> > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > (sofia/internal/1010 at 192.168.1.234) State HANGUP
> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel

> > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
> > sofia/internal/1010 at 192.168.1.234

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
> > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434

> > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
> > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP -> CS_REPORTING

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
> > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398

> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > (sofia/internal/1010 at 192.168.1.234) State REPORTING

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
> > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause: MEDIA_TIMEOUT

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411

> > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING -> CS_DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3

> > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3

> > (sofia/internal/1010 at 192.168.1.234) Ended
> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close Channel
> > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]

> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore original
> > codec.
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > (sofia/internal/1010 at 192.168.1.234) State DESTROY

> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
> > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60

> > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep

> >
> > ________________________________
> > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> > _______________________________________________
> > FreeSWITCH-users mailing list

> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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

> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org

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

> http://www.freeswitch.org
 		 	   		  
Ihre Daten brauchen Platz?  SkyDrive gibt Ihnen 25 GB - gratis!


_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org




-- 
Anthony Minessale II

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

Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888 at conference.freeswitch.org

pstn:+19193869900
 		 	   		  
_________________________________________________________________
http://redirect.gimas.net/?n=M1004xNoSpam2
Angst vor Spam? Hotmail sch?tzt Sie mit modernster Technologie!
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From jonas.gauffin at gmail.com  Wed Apr 14 01:05:11 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 10:05:11 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
Message-ID: <u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>

I found the problem. The phone uses RPORT and switches to the port reported
by FS instead of using the one specified in the provisioning. :(

On Wed, Apr 14, 2010 at 8:45 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I got a problem with Snom phones where they report a different port than
> the one provisioned. I tried to talk with the snom support without success.
>
> REGISTER sip:phone.gateon.se:5070 SIP/2.0
> Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK-u1y0v7is9ix0;rport
> From: "Helene XXX" <sip:u1000028 at phone.xxxx.se:5070>;tag=kjaa26j1qv
> To: "Helene xxx" <sip:u1000028 at phone.xxxx.se:5070>
> Call-ID: 3c267019ee2d-ni5vq3v88za6
> CSeq: 133 REGISTER
> Max-Forwards: 70
> Contact: <sip:u1000028 at 217.31.xxx.xxx
> :26062;line=yhkplirb>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d-ad61-2be3d1a88425>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom360/7.3.30
> Supported: gruu
> Allow-Events: dialog
> X-Real-IP: 192.168.3.109
> Expires: 3600
> Content-Length: 0
> The port that I provisioned (and I've doublechecked in the webinterface) is
> 5071 and not 26062.
>
> Have anyone else got this problem and solved it?
>
> Regards,
>   Jonas
>
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From jonas.gauffin at gmail.com  Wed Apr 14 02:49:26 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 11:49:26 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
Message-ID: <n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>

A follow up question:

Can I force freeswitch to use a specific port (saved in user directory) when
the phone asks for "rport", instead of using the actual port that the
message was received from?


On Wed, Apr 14, 2010 at 10:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> I found the problem. The phone uses RPORT and switches to the port reported
> by FS instead of using the one specified in the provisioning. :(
>
>
> On Wed, Apr 14, 2010 at 8:45 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello,
>>
>> I got a problem with Snom phones where they report a different port than
>> the one provisioned. I tried to talk with the snom support without success.
>>
>> REGISTER sip:phone.gateon.se:5070 SIP/2.0
>> Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK-u1y0v7is9ix0;rport
>> From: "Helene XXX" <sip:u1000028 at phone.xxxx.se:5070>;tag=kjaa26j1qv
>> To: "Helene xxx" <sip:u1000028 at phone.xxxx.se:5070>
>> Call-ID: 3c267019ee2d-ni5vq3v88za6
>> CSeq: 133 REGISTER
>> Max-Forwards: 70
>> Contact: <sip:u1000028 at 217.31.xxx.xxx
>> :26062;line=yhkplirb>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d-ad61-2be3d1a88425>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
>> User-Agent: snom360/7.3.30
>> Supported: gruu
>> Allow-Events: dialog
>> X-Real-IP: 192.168.3.109
>> Expires: 3600
>> Content-Length: 0
>> The port that I provisioned (and I've doublechecked in the webinterface)
>> is 5071 and not 26062.
>>
>> Have anyone else got this problem and solved it?
>>
>> Regards,
>>   Jonas
>>
>
>
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From stevendt at primrosebank.net  Wed Apr 14 02:57:34 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 14 Apr 2010 10:57:34 +0100
Subject: [Freeswitch-users] Cisco 7911G
References: <73733B3332CD4C9B883A79E0A1A46CE5@bp1.ad.bp.com>
	<p2y960738411004140020yab91c0aai54c2b88d4c01c881@mail.gmail.com>
Message-ID: <D4A29DDA52D243D5BB963D477516BCBF@bp1.ad.bp.com>

Hi Mathieu,

The phone was SCCP but I've just converted it to SIP as I have not managed 
to get time to try FreeSWITCH SCCP support yet. Having picked up a couple of 
working config files off the internet, I have managed to get the 7911 
working but the config file is a mystery to me at the moment and I would 
like to get some more information on what the options in there actually do.

On the SCCP front, I do have another SCCP phone which can't be converted to 
SIP so I do want to try to get mod_skinny working. I have looked at it, but 
there was no Windows (Visual Studio) build file when I last looked, so I 
can't compile it at the moment - unless there is a Build File in there now ?

regards
Dave


----- Original Message ----- 
From: "Mathieu Parent" <math.parent at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, April 14, 2010 8:20 AM
Subject: Re: [Freeswitch-users] Cisco 7911G


> On Tue, Apr 13, 2010 at 11:19 PM, Dave Stevenson
> <stevendt at primrosebank.net> wrote:
>>
>> Hi,
> Hi,
>
>>
>> does anyone have a working config file to use a Cisco 7911G with 
>> FreeSWITCH please ?
>>
>> The SEP<MAC>.CNF.XML file is significantly different to the 7940 format 
>> and I can't get the phone to register with FreeSWITCH, if anyone has a 
>> working version, I'd appreciate a copy please ?
>
> Are you using SIP or SCCP? Can you test mod_skinny (from trunk)?
>
> I don't have 7911G. But as a general note, you can search on
> <http://www.voip-info.org/>: plenty of examples for Asterisk can be
> adapted for FS.
>
> Regards
> Mathieu Parent
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From b_ball_henry at hotmail.com  Wed Apr 14 03:11:00 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Wed, 14 Apr 2010 18:11:00 +0800
Subject: [Freeswitch-users] Freeswitch memory issue/question
Message-ID: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>

Dear developers:

Our product is close to sipping stage and we are doing stress tests on FS
servers. We have now hit a dead end with our test results and running out of
ideas. And the result is pointing to memory issues. We have ran our test on
both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests.

Here is the test senario:

Send 10,000,000 invites to our test server running FS 1.0.6 with fixed
amount of invites per second. The xml dialplan simply bridge the call to an
unknown/unregistered user like so (user/500001) and therefore FS would hang
up once it doesn't find the user. When running the test, FS is steady, but
the memory usage slowly piles up over time. The memory keeps adding up till
it hit the ceiling, then FS crash. I don't understand what could be holding
the memory and not releasing it since we are not doing anything special and
the calls are not being answered. And I forgot to mention, through out the
whole test, the CPU usage stays at about 35% .

Please help


Besides FS native test, we have created our own application module to run
the same test. The app is written in C and basically what we did is to point
the xml dialplan to the custom app. Inside our app , we are simply doing the
same kind of test as the xml dialplan would do. The only thing we did for
the stress test is the following code:

switch_core_session_execute_application(session, "bridge", "user/500001");

Like the xml dialplan test, we are bridging the call to a unregistered user,
so the call would hang up right away if system doesn't find the user. But
the bizarre thing is this - the memory usage is about 7 ~ 8 times as much as
we are running only xml dialplan. This I don't understand. Our app is
calling the native FS C function to bridge the call , why would it use more
memory than the xml dialplan...

your input would be very much appreciated.

thanks


-- 
Henry Huang
aka bbhenry

VoIP & Open Source software Consultant
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From jonas.gauffin at gmail.com  Wed Apr 14 03:12:14 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 12:12:14 +0200
Subject: [Freeswitch-users] Update user directory
Message-ID: <i2jd021d3be1004140312x6aeba47ds9825db82c8093e07@mail.gmail.com>

Hello,

I'm using curl to provide FS with the user directory.
Can I tell FS (through the event socket) that it should update the user
directory for a specific user?

Regards,
  Jonas
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From pankaja at wientech.com  Wed Apr 14 03:27:16 2010
From: pankaja at wientech.com (pankaj anand)
Date: Wed, 14 Apr 2010 15:57:16 +0530
Subject: [Freeswitch-users] Reinvite time
Message-ID: <h2m809ad7ab1004140327k772de639l4005afec2670161b@mail.gmail.com>

hi,
by default

<param name="session-timeout"  value="120"/>
<param name="enable-timer" value="false"/>

were not defined in my sofia.conf.xml.

I added both these values but still freeswitch sends the reinvite message
after every 60 seconds. BTW my client is behind the NAT.

with regards
Pankaj anand
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From mayamatakeshi at gmail.com  Wed Apr 14 03:42:27 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 14 Apr 2010 19:42:27 +0900
Subject: [Freeswitch-users] Reinvite time
In-Reply-To: <h2m809ad7ab1004140327k772de639l4005afec2670161b@mail.gmail.com>
References: <h2m809ad7ab1004140327k772de639l4005afec2670161b@mail.gmail.com>
Message-ID: <i2k15b9404e1004140342r6678f9d8j1a243840ba5e4f3c@mail.gmail.com>

On Wed, Apr 14, 2010 at 7:27 PM, pankaj anand <pankaja at wientech.com> wrote:

> hi,
> by default
>
> <param name="session-timeout"  value="120"/>
> <param name="enable-timer" value="false"/>
>
> were not defined in my sofia.conf.xml.
>
> I added both these values but still freeswitch sends the reinvite message
> after every 60 seconds. BTW my client is behind the NAT.
>
>
I believe these are supposed to be put in the sip profile files.
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From pankajanand18 at gmail.com  Wed Apr 14 03:55:26 2010
From: pankajanand18 at gmail.com (pankaj anand)
Date: Wed, 14 Apr 2010 16:25:26 +0530
Subject: [Freeswitch-users] Reinvite problem
Message-ID: <x2y809ad7ab1004140355if7450b17g3652b1b85a13258c@mail.gmail.com>

these points were mentioned in the Sofia configuration, i was pointed out by
someone on the list. that is why i tried that with sofia configuration file.

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout

Any ways i will try to enable those variables in the specific sip profiles,
but i doubt even that will work because "sofia.conf.xml" includes the XML
files which are located in sip_profile folder.

Let me check and get back 2 you on this.

Thanks for the information.

with regards
Pankaj


---------- Forwarded message ----------
From: mayamatakeshi <mayamatakeshi at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 14 Apr 2010 19:42:27 +0900
Subject: Re: [Freeswitch-users] Reinvite time


On Wed, Apr 14, 2010 at 7:27 PM, pankaj anand <pankaja at wientech.com> wrote:

> hi,
> by default
>
> <param name="session-timeout"  value="120"/>
> <param name="enable-timer" value="false"/>
>
> were not defined in my sofia.conf.xml.
>
> I added both these values but still freeswitch sends the reinvite message
> after every 60 seconds. BTW my client is behind the NAT.
>
>
I believe these are supposed to be put in the sip profile files.
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From steveayre at gmail.com  Wed Apr 14 04:23:22 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 14 Apr 2010 12:23:22 +0100
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
Message-ID: <i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>

We are no longer on SVN, we have now moved to GIT for the 1.0.6 release.

http://wiki.freeswitch.org/wiki/Download_FreeSWITCH

Note that the latest most stable version is always likely to be latest
GIT version not the latest tagged released, as there will often be
bugs present in the releases that are fixed in the latest GIT version.
Tagged releases have more to do with the introduction of new features
(such as G729 support in 1.0.6) than any guarantee of stability.

-Steve


On 14 April 2010 09:00, Francisco Scaramanga <scaram at hotmail.de> wrote:
>
> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
> I thought it is a release candidate, isn't it? Is it safe to use it in
> production?
>
>
> BTW MEDIA_TIMEOUT didn't occur since 16 hours, so I think the failure was a
> defective contact..
>
>
>
> ________________________________
> Date: Tue, 13 Apr 2010 12:19:10 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
>
> The last stable release is actually 1.0.6
>
>
> On Tue, Apr 13, 2010 at 12:03 PM, Francisco Scaramanga <scaram at hotmail.de>
> wrote:
>
> I am running the last stable release 1.0.4 and could reproduce the failure.
> Sometimes it happens in a call of 4 minutes duration.
>
> I don't have a rtp packet trace but there is one more reason for
> MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls down to
> the winsock function recvfrom. It also could be a socket error.? If the
> recvfrom function returns -2 a MEDIA_TIMEOUT will happen. Maybe something
> was wrong with the network hardware cables, switch ..
> I changed my hardware. Test is running...
>
> How can I enable a rtp packet trace? It think I should check this too.
>
>
>> Date: Tue, 13 Apr 2010 16:23:44 +0100
>> From: steveayre at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
>>
>> Which version of FS are you running? From the line numbers it isn't
>> the latest, so you might try updating and see if you can still
>> reproduce.
>>
>> Given that the hangup appears to happen on mod_sofia.c:744, there is a
>> MEDIA_TIMEOUT near that line which is triggered if FS times out trying
>> to read a video frame from RTP. That will occur when the RTP stack
>> thinks it has missed too many RTP packets.
>>
>> Do you have an RTP packet trace for the call?
>>
>> -Steve
>>
>>
>>
>> On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de> wrote:
>> > hi , I have a problem with long duration calls. After 11 hours and 35
>> > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
>> > reproduceable.
>> > Any idea? I use SIP phone SIEMENS Gigaset S685.
>> >
>> > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
>> > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
>> > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
>> > [sofia/internal/1010 at 192.168.1.234] has been answered
>> > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
>> > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
>> > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
>> > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
>> > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
>> > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
>> > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [KILL]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
>> > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
>> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
>> > (sofia/internal/1010 at 192.168.1.234) State HANGUP
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
>> > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
>> > sofia/internal/1010 at 192.168.1.234
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
>> > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
>> > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
>> > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP ->
>> > CS_REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal
>> > sofia/internal/1010 at 192.168.1.234 [BREAK]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
>> > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
>> > (sofia/internal/1010 at 192.168.1.234) State REPORTING
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
>> > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause:
>> > MEDIA_TIMEOUT
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
>> > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
>> > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING ->
>> > CS_DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3
>> > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external entities
>> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3
>> > (sofia/internal/1010 at 192.168.1.234) Ended
>> > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close
>> > Channel
>> > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore
>> > original
>> > codec.
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
>> > (sofia/internal/1010 at 192.168.1.234) State DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
>> > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
>> > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
>> > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
>> > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
>> >
>> > ________________________________
>> > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> ________________________________
> Angst vor Spam? Hotmail sch?tzt Sie mit modernster Technologie!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From math.parent at gmail.com  Wed Apr 14 04:30:12 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Wed, 14 Apr 2010 13:30:12 +0200
Subject: [Freeswitch-users] Cisco 7911G
In-Reply-To: <D4A29DDA52D243D5BB963D477516BCBF@bp1.ad.bp.com>
References: <73733B3332CD4C9B883A79E0A1A46CE5@bp1.ad.bp.com>
	<p2y960738411004140020yab91c0aai54c2b88d4c01c881@mail.gmail.com>
	<D4A29DDA52D243D5BB963D477516BCBF@bp1.ad.bp.com>
Message-ID: <i2n960738411004140430i64db75d3ydd611d2a73bfc038@mail.gmail.com>

On Wed, Apr 14, 2010 at 11:57 AM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Hi Mathieu,
Hi Dave,

> The phone was SCCP but I've just converted it to SIP as I have not managed
> to get time to try FreeSWITCH SCCP support yet. Having picked up a couple of
> working config files off the internet, I have managed to get the 7911
> working but the config file is a mystery to me at the moment and I would
> like to get some more information on what the options in there actually do.
>
> On the SCCP front, I do have another SCCP phone which can't be converted to
> SIP so I do want to try to get mod_skinny working. I have looked at it, but
> there was no Windows (Visual Studio) build file when I last looked, so I
> can't compile it at the moment - unless there is a Build File in there now ?

I don't develop on Windows, can anybody help on this side?

Mathieu


From maciej.aniserowicz at gmail.com  Wed Apr 14 04:48:45 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 14 Apr 2010 13:48:45 +0200
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>
References: <1269856895179-4817081.post@n2.nabble.com>	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>	<4BB1A30A.4060701@gmail.com>	<1270796330100-4875677.post@n2.nabble.com>	<u2w87f2f3b91004091001s1cb1e559n995216538a46ca42@mail.gmail.com>	<1270833275502-4878370.post@n2.nabble.com>
	<m2i87f2f3b91004091050kf813ab83r5a471c177d19d15d@mail.gmail.com>
Message-ID: <4BC5AB9D.9060405@gmail.com>

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From david.ponzone at gmail.com  Wed Apr 14 04:49:24 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 14 Apr 2010 13:49:24 +0200
Subject: [Freeswitch-users] Reinvite time
In-Reply-To: <h2m809ad7ab1004140327k772de639l4005afec2670161b@mail.gmail.com>
References: <h2m809ad7ab1004140327k772de639l4005afec2670161b@mail.gmail.com>
Message-ID: <5DC96151-C924-4217-BCDC-2543FA33339D@gmail.com>

Pankaj,

as someone already told you, that's supposed to go in the SIP profile.
Also, if you set enable-timer to false, it's no used to set session- 
timeout, as you can guess.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 14/04/2010 ? 12:27, pankaj anand a ?crit :

> hi,
> by default
>
> <param name="session-timeout"  value="120"/>
> <param name="enable-timer" value="false"/>
>
> were not defined in my sofia.conf.xml.
>
> I added both these values but still freeswitch sends the reinvite  
> message after every 60 seconds. BTW my client is behind the NAT.
>
> with regards
> Pankaj anand
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pankaja at wientech.com  Wed Apr 14 05:03:05 2010
From: pankaja at wientech.com (pankaj anand)
Date: Wed, 14 Apr 2010 17:33:05 +0530
Subject: [Freeswitch-users] Reinvite problem
Message-ID: <o2l809ad7ab1004140503s71a7310fu3828b12814cc338f@mail.gmail.com>

david,

 i made changes in my sip profile but still it doesn't show any difference
in timings of reinvite message. it still goes after every 60 seconds.

regards
Pankaj anand
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From mike at jerris.com  Wed Apr 14 05:25:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 08:25:18 -0400
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>,
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>,
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>,
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
Message-ID: <F1D4BA36-EBE0-4B65-A376-AE7D5006CFE1@jerris.com>

that's not 1.0.6


http://wiki.freeswitch.org/wiki/Installation_Guide

On Apr 14, 2010, at 4:00 AM, Francisco Scaramanga <scaram at hotmail.de>  
wrote:

>
> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
> I thought it is a release candidate, isn't it? Is it safe to use it  
> in production?
>
>
> BTW MEDIA_TIMEOUT didn't occur since 16 hours, so I think the  
> failure was a defective contact..
>
>
>
> Date: Tue, 13 Apr 2010 12:19:10 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
>
> The last stable release is actually 1.0.6
>
>
> On Tue, Apr 13, 2010 at 12:03 PM, Francisco Scaramanga <scaram at hotmail.de 
> > wrote:
> I am running the last stable release 1.0.4 and could reproduce the  
> failure. Sometimes it happens in a call of 4 minutes duration.
>
> I don't have a rtp packet trace but there is one more reason for  
> MEDIA_TIMEOUT. I had a look at the sourcecode and followed the calls  
> down to the winsock function recvfrom. It also could be a socket  
> error.  If the recvfrom function returns -2 a MEDIA_TIMEOUT will  
> happen. Maybe something was wrong with the network hardware cables,  
> switch ..
> I changed my hardware. Test is running...
>
> How can I enable a rtp packet trace? It think I should check this too.
>
>
> > Date: Tue, 13 Apr 2010 16:23:44 +0100
> > From: steveayre at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration  
> call
>
> >
> > Which version of FS are you running? From the line numbers it isn't
> > the latest, so you might try updating and see if you can still
> > reproduce.
> >
> > Given that the hangup appears to happen on mod_sofia.c:744, there  
> is a
> > MEDIA_TIMEOUT near that line which is triggered if FS times out  
> trying
> > to read a video frame from RTP. That will occur when the RTP stack
> > thinks it has missed too many RTP packets.
> >
> > Do you have an RTP packet trace for the call?
> >
> > -Steve
> >
> >
> >
> > On 13 April 2010 14:53, Francisco Scaramanga <scaram at hotmail.de>  
> wrote:
> > > hi , I have a problem with long duration calls. After 11 hours  
> and 35
> > > minutes the call ends because of MEDIA_TIMEOUT. It seems to be
> > > reproduceable.
> > > Any idea? I use SIP phone SIEMENS Gigaset S685.
> > >
> > > 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel
> > > sofia/internal/1010 at 192.168.1.234 entering state [completed][200]
> > > 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630  
> Send signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel
> > > [sofia/internal/1010 at 192.168.1.234] has been answered
> > > 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182
> > > sofia/internal/1010 at 192.168.1.234 receive message [AUDIO_SYNC]
> > > EXECUTE sofia/internal/1010 at 192.168.1.234 MyModul
> > > 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel
> > > sofia/internal/1010 at 192.168.1.234 entering state [ready][200]
> > > 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup
> > > sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send  
> signal
> > > sofia/internal/1010 at 192.168.1.234 [KILL]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932  
> Send signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491
> > > (sofia/internal/1010 at 192.168.1.234) State EXECUTE going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > > (sofia/internal/1010 at 192.168.1.234) Running State Change CS_HANGUP
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > > (sofia/internal/1010 at 192.168.1.234) State HANGUP
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel
> > > sofia/internal/1010 at 192.168.1.234 hanging up, cause: MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to
> > > sofia/internal/1010 at 192.168.1.234
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46
> > > sofia/internal/1010 at 192.168.1.234 Standard HANGUP, cause:  
> MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434
> > > (sofia/internal/1010 at 192.168.1.234) State HANGUP going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476
> > > (sofia/internal/1010 at 192.168.1.234) State Change CS_HANGUP ->  
> CS_REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932  
> Send signal
> > > sofia/internal/1010 at 192.168.1.234 [BREAK]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398
> > > (sofia/internal/1010 at 192.168.1.234) Running State Change  
> CS_REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > > (sofia/internal/1010 at 192.168.1.234) State REPORTING
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53
> > > sofia/internal/1010 at 192.168.1.234 Standard REPORTING, cause:  
> MEDIA_TIMEOUT
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612
> > > (sofia/internal/1010 at 192.168.1.234) State REPORTING going to sleep
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411
> > > (sofia/internal/1010 at 192.168.1.234) State Change CS_REPORTING ->  
> CS_DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068  
> Session 3
> > > (sofia/internal/1010 at 192.168.1.234) Locked, Waiting on external  
> entities
> > > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086  
> Session 3
> > > (sofia/internal/1010 at 192.168.1.234) Ended
> > > 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088  
> Close Channel
> > > sofia/internal/1010 at 192.168.1.234 [CS_DESTROY]
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122  
> Restore original
> > > codec.
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > > (sofia/internal/1010 at 192.168.1.234) State DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255
> > > sofia/internal/1010 at 192.168.1.234 SOFIA DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60
> > > sofia/internal/1010 at 192.168.1.234 Standard DESTROY
> > > 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564
> > > (sofia/internal/1010 at 192.168.1.234) State DESTROY going to sleep
> > >
> > > ________________________________
> > > Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
>
> Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> Angst vor Spam? Hotmail sch?tzt Sie mit modernster Technologie!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Apr 14 05:28:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 08:28:05 -0400
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
Message-ID: <00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>

I think your question here is can you force the nat device to pick a  
specific port to map?

On Apr 14, 2010, at 5:49 AM, Jonas Gauffin <jonas.gauffin at gmail.com>  
wrote:

> A follow up question:
>
> Can I force freeswitch to use a specific port (saved in user  
> directory) when the phone asks for "rport", instead of using the  
> actual port that the message was received from?
>
>
> On Wed, Apr 14, 2010 at 10:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com 
> > wrote:
> I found the problem. The phone uses RPORT and switches to the port  
> reported by FS instead of using the one specified in the  
> provisioning. :(
>
>
> On Wed, Apr 14, 2010 at 8:45 AM, Jonas Gauffin <jonas.gauffin at gmail.com 
> > wrote:
> Hello,
>
> I got a problem with Snom phones where they report a different port  
> than the one provisioned. I tried to talk with the snom support  
> without success.
>
> REGISTER sip:phone.gateon.se:5070 SIP/2.0
> Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK- 
> u1y0v7is9ix0;rport
> From: "Helene XXX" <sip:u1000028 at phone.xxxx.se:5070>;tag=kjaa26j1qv
> To: "Helene xxx" <sip:u1000028 at phone.xxxx.se:5070>
> Call-ID: 3c267019ee2d-ni5vq3v88za6
> CSeq: 133 REGISTER
> Max-Forwards: 70
> Contact: <sip:u1000028 at 217.31.xxx.xxx:26062;line=yhkplirb>;reg- 
> id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d- 
> ad61- 
> 2be3d1a88425> 
> ";a 
> udio; 
> mobility= 
> "fixed" 
> ;duplex= 
> "full" 
> ;description= 
> "snom360" 
> ;actor= 
> "principal" 
> ;events= 
> "dialog" 
> ;methods= 
> "INVITE, 
> ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom360/7.3.30
> Supported: gruu
> Allow-Events: dialog
> X-Real-IP: 192.168.3.109
> Expires: 3600
> Content-Length: 0
> The port that I provisioned (and I've doublechecked in the  
> webinterface) is 5071 and not 26062.
>
> Have anyone else got this problem and solved it?
>
> Regards,
>   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
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From mike at jerris.com  Wed Apr 14 05:32:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 08:32:03 -0400
Subject: [Freeswitch-users] Update user directory
In-Reply-To: <i2jd021d3be1004140312x6aeba47ds9825db82c8093e07@mail.gmail.com>
References: <i2jd021d3be1004140312x6aeba47ds9825db82c8093e07@mail.gmail.com>
Message-ID: <D8EDB4F3-D1AE-44C7-9C52-53A1C5944960@jerris.com>

It asks every time already.  what would it do diffferently if you told  
it?

On Apr 14, 2010, at 6:12 AM, Jonas Gauffin <jonas.gauffin at gmail.com>  
wrote:

> Hello,
>
> I'm using curl to provide FS with the user directory.
> Can I tell FS (through the event socket) that it should update the  
> user directory for a specific user?
>
> Regards,
>   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


From mike at jerris.com  Wed Apr 14 06:08:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 09:08:32 -0400
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
	<i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>
Message-ID: <9E1A4D19-A8C4-41A3-AD2B-E324EA31B43B@jerris.com>

That being said, for a change things have been somewhat quiet after the 1.0.6 release.  There have been some bug fixes that went in after, but not the usual frenzy of reports right after a release.

Mike

On Apr 14, 2010, at 7:23 AM, Steven Ayre wrote:

> We are no longer on SVN, we have now moved to GIT for the 1.0.6 release.
> 
> http://wiki.freeswitch.org/wiki/Download_FreeSWITCH
> 
> Note that the latest most stable version is always likely to be latest
> GIT version not the latest tagged released, as there will often be
> bugs present in the releases that are fixed in the latest GIT version.
> Tagged releases have more to do with the introduction of new features
> (such as G729 support in 1.0.6) than any guarantee of stability.
> 
> -Steve
> 
> 
> On 14 April 2010 09:00, Francisco Scaramanga <scaram at hotmail.de> wrote:
>> 
>> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
>> I thought it is a release candidate, isn't it? Is it safe to use it in
>> production?
>> 



From jonas.gauffin at gmail.com  Wed Apr 14 06:11:47 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 15:11:47 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
Message-ID: <k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>

Well, yes. But snom can only turn off rport through provisioning and only in
the latest firmware.

It would be great if it could be turned off in FS too, per user account.


On Wed, Apr 14, 2010 at 2:28 PM, Michael Jerris <mike at jerris.com> wrote:

> I think your question here is can you force the nat device to pick a
> specific port to map?
>
>
> On Apr 14, 2010, at 5:49 AM, Jonas Gauffin <jonas.gauffin at gmail.com>
> wrote:
>
> A follow up question:
>
> Can I force freeswitch to use a specific port (saved in user directory)
> when the phone asks for "rport", instead of using the actual port that the
> message was received from?
>
>
> On Wed, Apr 14, 2010 at 10:05 AM, Jonas Gauffin <<jonas.gauffin at gmail.com>
> jonas.gauffin at gmail.com> wrote:
>
>> I found the problem. The phone uses RPORT and switches to the port
>> reported by FS instead of using the one specified in the provisioning. :(
>>
>>
>> On Wed, Apr 14, 2010 at 8:45 AM, Jonas Gauffin <<jonas.gauffin at gmail.com>
>> jonas.gauffin at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I got a problem with Snom phones where they report a different port than
>>> the one provisioned. I tried to talk with the snom support without success.
>>>
>>> REGISTER sip:phone.gateon.se:5070 SIP/2.0
>>> Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK-u1y0v7is9ix0;rport
>>> From: "Helene XXX" <sip:u1000028 at phone.xxxx.se:5070>;tag=kjaa26j1qv
>>> To: "Helene xxx" <sip:u1000028 at phone.xxxx.se:5070>
>>> Call-ID: 3c267019ee2d-ni5vq3v88za6
>>> CSeq: 133 REGISTER
>>> Max-Forwards: 70
>>> Contact: <sip:u1000028 at 217.31.xxx.xxx
>>> :26062;line=yhkplirb>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d-ad61-2be3d1a88425>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
>>> User-Agent: snom360/7.3.30
>>> Supported: gruu
>>> Allow-Events: dialog
>>> X-Real-IP: 192.168.3.109
>>> Expires: 3600
>>> Content-Length: 0
>>> The port that I provisioned (and I've doublechecked in the webinterface)
>>> is 5071 and not 26062.
>>>
>>> Have anyone else got this problem and solved it?
>>>
>>> Regards,
>>>   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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Wed Apr 14 06:16:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 14 Apr 2010 15:16:21 +0200
Subject: [Freeswitch-users] Reinvite problem
In-Reply-To: <o2l809ad7ab1004140503s71a7310fu3828b12814cc338f@mail.gmail.com>
References: <o2l809ad7ab1004140503s71a7310fu3828b12814cc338f@mail.gmail.com>
Message-ID: <193D0917-818B-4708-8357-04518DA37C4F@gmail.com>

You rescaned the profile ?

sofia profile internal rescan reloadxml

Replace internal with your profile.

PS: can you check your mail client settings ? In a mailing-list, you  
are not supposed to reply to the mail sender directly. You are  
supposed to honor the Reply-To header.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 14/04/2010 ? 14:03, pankaj anand a ?crit :

> david,
>
>  i made changes in my sip profile but still it doesn't show any  
> difference in timings of reinvite message. it still goes after every  
> 60 seconds.
>
> regards
> Pankaj anand

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From pjintheusa at gmail.com  Wed Apr 14 06:19:28 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 14 Apr 2010 09:19:28 -0400
Subject: [Freeswitch-users] SVN read only mirror ???
Message-ID: <s2v367751821004140619k135dcd05i9af826b66b5ccef2@mail.gmail.com>

Hi there,

I read previously on this list that the devs would try and maintain a read
only mirror of FreeSWITCH on SVN. Was this possible?

 When I use the "Latest Build Version:" link from the front page I get:

"moose penis

NOT IT IS NOT HACKED... ITS CALLED A JOKE and some advice to use FreeSWITCH
1.0.6"


Nice!

So the question is, is the SVN mirror being maintained or are we using GIT
exclusively now?

Pj
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From brian at freeswitch.org  Wed Apr 14 06:53:19 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Apr 2010 08:53:19 -0500
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
	<k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
Message-ID: <CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>

You want rport otherwise NAT HELL will begin.

/b

On Apr 14, 2010, at 8:11 AM, Jonas Gauffin wrote:

> Well, yes. But snom can only turn off rport through provisioning and only in the latest firmware.
> 
> It would be great if it could be turned off in FS too, per user account.



From pankaja at wientech.com  Wed Apr 14 07:23:19 2010
From: pankaja at wientech.com (pankaj anand)
Date: Wed, 14 Apr 2010 19:53:19 +0530
Subject: [Freeswitch-users] Reinvite problem
Message-ID: <h2t809ad7ab1004140723n8f16a44fq598ca10141ff8522@mail.gmail.com>

hi,
  I tried
"sofia profile external rescan reloadxml"

but still its not working ..
 i was looking at some other posts and found the following one :

http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.html

has anybody tried this before ?
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From brian at freeswitch.org  Wed Apr 14 07:30:50 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Apr 2010 09:30:50 -0500
Subject: [Freeswitch-users] Reinvite problem
In-Reply-To: <h2t809ad7ab1004140723n8f16a44fq598ca10141ff8522@mail.gmail.com>
References: <h2t809ad7ab1004140723n8f16a44fq598ca10141ff8522@mail.gmail.com>
Message-ID: <2D18D430-CE8B-4F1B-B27B-38C6EBE4FDDF@freeswitch.org>

not all settings reload on rescan... restart the profile when in doubt.

/b

On Apr 14, 2010, at 9:23 AM, pankaj anand wrote:

> hi, 
>   I tried 
> "sofia profile external rescan reloadxml"
>  
> but still its not working .. 
>  i was looking at some other posts and found the following one :
> 
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.html
> 
> has anybody tried this before ?
> 

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From jonas.gauffin at gmail.com  Wed Apr 14 07:50:18 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 14 Apr 2010 16:50:18 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
	<k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
	<CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>
Message-ID: <v2rd021d3be1004140750refd7f5cdi7c95ca508bbed1e8@mail.gmail.com>

Most of my customers have forwarded ports in their firewalls/routers. Those
ports will not be used if rport is turned on.

I currently got a problem with a customer  who is using an old Windows 2000
server as firewall/router. It wont let packets through when using rport. It
would work if rport was turned off and all communication was done using the
configured port.

On Wed, Apr 14, 2010 at 3:53 PM, Brian West <brian at freeswitch.org> wrote:

> You want rport otherwise NAT HELL will begin.
>
> /b
>
> On Apr 14, 2010, at 8:11 AM, Jonas Gauffin wrote:
>
> > Well, yes. But snom can only turn off rport through provisioning and only
> in the latest firmware.
> >
> > It would be great if it could be turned off in FS too, per user account.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From linux4michelle at tamay-dogan.net  Wed Apr 14 07:53:41 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 14 Apr 2010 16:53:41 +0200
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <y2r191c3a031004130853u3bd2cafejeac02d85c17dbe80@mail.gmail.com>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
	<4BC44A68.9040209@telemaque.fr>
	<20100413105326.GY12939@tamay-dogan.net>
	<y2r191c3a031004130853u3bd2cafejeac02d85c17dbe80@mail.gmail.com>
Message-ID: <20100414145341.GC12939@tamay-dogan.net>

Hello Anthony Minessale,

Am 2010-04-13 10:53:48, hacktest Du folgendes herunter:
> You guys talk too much =D
> 
> The log file reached 2gb and even in 2010 in the age of multiple terabyte
> media on a 64 bit box it somehow still is a problem with the file size
> causing software to break.

Hehehe...

I run logrotate every hour to get rid of singel 18 GByte  logfiles  from
Apache, Courier and Spamassassin.  (I have in total arround 70 GByte  of
logs per day)

> The problem has been fixed.

What was it?

;-)

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
Gesch. Michelle Konzack          Gesch. Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz               Kinzigstra?e 17
67100 Strasbourg/France         77694 Kehl/Germany
Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From anthony.minessale at gmail.com  Wed Apr 14 07:54:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Apr 2010 09:54:32 -0500
Subject: [Freeswitch-users] Reinvite problem
In-Reply-To: <2D18D430-CE8B-4F1B-B27B-38C6EBE4FDDF@freeswitch.org>
References: <h2t809ad7ab1004140723n8f16a44fq598ca10141ff8522@mail.gmail.com>
	<2D18D430-CE8B-4F1B-B27B-38C6EBE4FDDF@freeswitch.org>
Message-ID: <k2o191c3a031004140754j482f0133i6369d17825746101@mail.gmail.com>

When you have nat detection enabled it forces the session timer to a low
value.
This is not configurable.


On Wed, Apr 14, 2010 at 9:30 AM, Brian West <brian at freeswitch.org> wrote:

> not all settings reload on rescan... restart the profile when in doubt.
>
> /b
>
> On Apr 14, 2010, at 9:23 AM, pankaj anand wrote:
>
> hi,
>   I tried
> "sofia profile external rescan reloadxml"
>
> but still its not working ..
>  i was looking at some other posts and found the following one :
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.html
>
> has anybody tried this before ?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From helmut.kuper at ewetel.de  Wed Apr 14 07:54:30 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 14 Apr 2010 16:54:30 +0200
Subject: [Freeswitch-users] Problem with SIP INFO (caller display update)
	after connect
Message-ID: <4BC5D726.5000508@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,


I have the following problem.
A call comes in the following way:
ISDN PRI->Sangoma->wanpipe->openzap->mod_openzap->freeswitch->B-Party

I use a national numbering plan, so caller's number e.g 044112345 is
transmitted as 4412345 on isdn level.
My dialplan adds two zeros to the effective_caller_id_name and -number.
So it is now 0044112345. The B-Party displays this correctly. But as
soon as B-Party picks up the caller's name switched back to 44112345
which is quite bad.

I tried to set caller's name and number via effective_caller_id_... ,
via origination_caller_id_... and via  origination_callee_id_... but
nothing worked.

It looks like the sip info process after connecting two sessions is not
effected by those channel variables mentioned above.

I use FS tunk (17073M)

Any ideas what is wrong here?

kind regards
helmut
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From bcxml at hotmail.com  Wed Apr 14 08:02:35 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Wed, 14 Apr 2010 11:02:35 -0400
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
References: <28146712.post@talk.nabble.com>,
	<20100406040606.78daf9fd@anubis.defcon1>,
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>,
	<1270823599661-4877398.post@n2.nabble.com>,
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
Message-ID: <SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>



Can anyone provide some asistance for this issue ?

 

Thanks

 

 

Brian


From: bcxml at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: RE: [Freeswitch-users] Problem with IVR app starting too soon
Date: Mon, 12 Apr 2010 15:08:30 -0400



I am resubmitting this just in case it got lost in the pile
 
Here is the link
 
http://pastebin.freeswitch.org/12675
 
Thanks
 
 
Brian Campbell
 


Date: Fri, 9 Apr 2010 10:03:38 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with IVR app starting too soon

Can you pastebin a full debug log of this call?
-MC


On Fri, Apr 9, 2010 at 7:33 AM, Brian Campbell <bcxml at hotmail.com> wrote:



I tried what you mentioned but it did not seem to have any effect. I am
still losing part of the openning message

Here is my inbound dialplan..

<extension name="Inbound_1234567890">
 <condition field='destination_number' expression='1234567890'>
   <action application="playback" data="silence_stream://1000"/>
   <action application="bridge"
data="{bypass_media=true}sofia/internal/1234567890 at 127.0.0.1:5060;transport=tcp"/>
 </condition>
</extension>


Brian
--
View this message in context: http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4877398.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




Live connected. Get Hotmail & Messenger for mobile. 		 	   		  
_________________________________________________________________
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From brian at freeswitch.org  Wed Apr 14 08:07:54 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Apr 2010 10:07:54 -0500
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>
References: <28146712.post@talk.nabble.com>,
	<20100406040606.78daf9fd@anubis.defcon1>,
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>,
	<1270823599661-4877398.post@n2.nabble.com>,
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
	<SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>
Message-ID: <84617B93-6D32-41CC-9E9F-5484413B46A0@freeswitch.org>

If the app is starting too soon then SLEEP or do playback silence_stream://2000 before the IVR so the remote side has time to get media up.  

/b

On Apr 14, 2010, at 10:02 AM, Brian Campbell wrote:

> 
> Can anyone provide some asistance for this issue ?
>  
> Thanks
>  
>  
> Brian

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From anthony.minessale at gmail.com  Wed Apr 14 08:10:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Apr 2010 10:10:12 -0500
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <20100414145341.GC12939@tamay-dogan.net>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
	<4BC44A68.9040209@telemaque.fr>
	<20100413105326.GY12939@tamay-dogan.net>
	<y2r191c3a031004130853u3bd2cafejeac02d85c17dbe80@mail.gmail.com>
	<20100414145341.GC12939@tamay-dogan.net>
Message-ID: <n2q191c3a031004140810u4e9d0569t1ac9381666726ea6@mail.gmail.com>

The problem was the mailing list software suffers from a 2gb log file
limitation and our log rotate cronjob
was paused somehow by accident during our last migration.


On Wed, Apr 14, 2010 at 9:53 AM, Michelle Konzack <
linux4michelle at tamay-dogan.net> wrote:

> Hello Anthony Minessale,
>
> Am 2010-04-13 10:53:48, hacktest Du folgendes herunter:
> > You guys talk too much =D
> >
> > The log file reached 2gb and even in 2010 in the age of multiple terabyte
> > media on a 64 bit box it somehow still is a problem with the file size
> > causing software to break.
>
> Hehehe...
>
> I run logrotate every hour to get rid of singel 18 GByte  logfiles  from
> Apache, Courier and Spamassassin.  (I have in total arround 70 GByte  of
> logs per day)
>
> > The problem has been fixed.
>
> What was it?
>
> ;-)
>
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>
> --
> ##################### Debian GNU/Linux Consultant ######################
>   Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
> Gesch. Michelle Konzack          Gesch. Michelle Konzack
>
> Apt. 917 (homeoffice)
> 50, rue de Soultz               Kinzigstra?e 17
> 67100 Strasbourg/France         77694 Kehl/Germany
> Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
> Tel: +33-9-52705884 fix
>
> <http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ    #328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 14 08:13:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Apr 2010 10:13:23 -0500
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>
References: <28146712.post@talk.nabble.com>
	<20100406040606.78daf9fd@anubis.defcon1>
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>
	<1270823599661-4877398.post@n2.nabble.com>
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>
	<SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>
Message-ID: <q2h191c3a031004140813u4020efdfrf77f2bc8c961b2f1@mail.gmail.com>

the playback of the silence stream should be on the ivr side, not before you
make the call.
you can play a file in your dialplan there to prove you have audio on that
leg, its the far end with the ivr
who is starting too soon.


On Wed, Apr 14, 2010 at 10:02 AM, Brian Campbell <bcxml at hotmail.com> wrote:

>
> Can anyone provide some asistance for this issue ?
>
> Thanks
>
>
> Brian
> ------------------------------
> From: bcxml at hotmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: RE: [Freeswitch-users] Problem with IVR app starting too soon
> Date: Mon, 12 Apr 2010 15:08:30 -0400
>
>
> I am resubmitting this just in case it got lost in the pile
>
> Here is the link
>
> http://pastebin.freeswitch.org/12675
>
> Thanks
>
>
> Brian Campbell
>
> ------------------------------
> Date: Fri, 9 Apr 2010 10:03:38 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Problem with IVR app starting too soon
>
>
> Can you pastebin a full debug log of this call?
> -MC
>
> On Fri, Apr 9, 2010 at 7:33 AM, Brian Campbell <bcxml at hotmail.com> wrote:
>
>
>
> I tried what you mentioned but it did not seem to have any effect. I am
> still losing part of the openning message
>
> Here is my inbound dialplan..
>
> <extension name="Inbound_1234567890">
>  <condition field='destination_number' expression='1234567890'>
>    <action application="playback" data="silence_stream://1000"/>
>    <action application="bridge"
> data="{bypass_media=true}sofia/internal/1234567890 at 127.0.0.1:5060
> ;transport=tcp"/>
>  </condition>
> </extension>
>
>
> Brian
> --
> View this message in context:
> http://n2.nabble.com/Problem-with-IVR-app-starting-too-soon-tp4856555p4877398.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
>
>
>
> ------------------------------
> Live connected. Get Hotmail & Messenger for mobile.<http://go.microsoft.com/?linkid=9724459>
> ------------------------------
> Hotmail & Messenger. Get them on your phone now.<http://go.microsoft.com/?linkid=9724456>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 14 08:19:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Apr 2010 10:19:11 -0500
Subject: [Freeswitch-users] Freeswitch memory issue/question
In-Reply-To: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
Message-ID: <h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>

We have a policy against load testing issues on this mailing list.

What you are doing is most likely doing something wrong and we don't have
time to debug it for you.
We get way too many requests like this and there is a 99% result in proving
user error and improper load testing.

You can consider paid support from FreeSWITCH Solutions LLC for $175/hr per
consultant and an ongoing support contract for $10k/yr to deal with business
related bugfixing and support.

The free help we give here is based on a community shared resource pool and
we simply do not have enough to go around helping people with your type of
issue.  Please feel free to report bugs if you can pinpoint one using a more
realistic use case that does not involve DDosing your box with 10 million
invites.


On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Dear developers:
>
> Our product is close to sipping stage and we are doing stress tests on FS
> servers. We have now hit a dead end with our test results and running out of
> ideas. And the result is pointing to memory issues. We have ran our test on
> both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests.
>
> Here is the test senario:
>
> Send 10,000,000 invites to our test server running FS 1.0.6 with fixed
> amount of invites per second. The xml dialplan simply bridge the call to an
> unknown/unregistered user like so (user/500001) and therefore FS would hang
> up once it doesn't find the user. When running the test, FS is steady, but
> the memory usage slowly piles up over time. The memory keeps adding up till
> it hit the ceiling, then FS crash. I don't understand what could be holding
> the memory and not releasing it since we are not doing anything special and
> the calls are not being answered. And I forgot to mention, through out the
> whole test, the CPU usage stays at about 35% .
>
> Please help
>
>
> Besides FS native test, we have created our own application module to run
> the same test. The app is written in C and basically what we did is to point
> the xml dialplan to the custom app. Inside our app , we are simply doing the
> same kind of test as the xml dialplan would do. The only thing we did for
> the stress test is the following code:
>
> switch_core_session_execute_application(session, "bridge", "user/500001");
>
> Like the xml dialplan test, we are bridging the call to a unregistered
> user, so the call would hang up right away if system doesn't find the user.
> But the bizarre thing is this - the memory usage is about 7 ~ 8 times as
> much as we are running only xml dialplan. This I don't understand. Our app
> is calling the native FS C function to bridge the call , why would it use
> more memory than the xml dialplan...
>
> your input would be very much appreciated.
>
> thanks
>
>
> --
> Henry Huang
> aka bbhenry
>
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From bcxml at hotmail.com  Wed Apr 14 08:21:51 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Wed, 14 Apr 2010 11:21:51 -0400
Subject: [Freeswitch-users] Problem with IVR app starting too soon
In-Reply-To: <84617B93-6D32-41CC-9E9F-5484413B46A0@freeswitch.org>
References: <28146712.post@talk.nabble.com>, ,
	<20100406040606.78daf9fd@anubis.defcon1>, ,
	<ED49B892-F4A2-450A-A14F-9558ED88C677@freeswitch.org>, ,
	<1270823599661-4877398.post@n2.nabble.com>, ,
	<n2q87f2f3b91004091003hf4c1aa9ftff9c8ee11cf7f9ed@mail.gmail.com>,
	<SNT135-w371665403B6A2D6B8DC38EB7100@phx.gbl>,
	<84617B93-6D32-41CC-9E9F-5484413B46A0@freeswitch.org>
Message-ID: <SNT135-w64B89F11F96AA63510A34B7100@phx.gbl>


Thank you Brian. Previously I had the setting as silence_stream://1000 which I guess did not give enough time

 

Setting to silence_stream://2000 seems to work better

 

Thanks for your help

 

 

Brian Campbell
 


From: brian at freeswitch.org
Date: Wed, 14 Apr 2010 10:07:54 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with IVR app starting too soon

If the app is starting too soon then SLEEP or do playback silence_stream://2000 before the IVR so the remote side has time to get media up.  


/b



On Apr 14, 2010, at 10:02 AM, Brian Campbell wrote:

Can anyone provide some asistance for this issue ?
 
Thanks
 
 
Brian
 		 	   		  
_________________________________________________________________
Hotmail & Messenger are available on your phone. Try now.
http://go.microsoft.com/?linkid=9724461
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From red.rain.seven at gmail.com  Wed Apr 14 08:41:14 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 14 Apr 2010 23:41:14 +0800
Subject: [Freeswitch-users] Freeswitch memory issue/question
In-Reply-To: <h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
Message-ID: <k2n59ad9ca11004140841kd198c5b6v79e63bc5bfafd60@mail.gmail.com>

Anthony:

yeh, I thought it could be something wrong with my testing setup too. And
you are right, these kind of question requires professional support.

Thanks for taking the time to reply me. But if anyone in the community would
be kind to point out what should be a more proper way of conducting a SIPp
test based on their experience, I am all ears.

Thanks again, and FS 1.0.6 is truly great. Jumping from 1.0.4 to 1.0.6, we
found it more robust and bug free.

Henry

On Wed, Apr 14, 2010 at 11:19 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We have a policy against load testing issues on this mailing list.
>
> What you are doing is most likely doing something wrong and we don't have
> time to debug it for you.
> We get way too many requests like this and there is a 99% result in proving
> user error and improper load testing.
>
> You can consider paid support from FreeSWITCH Solutions LLC for $175/hr per
> consultant and an ongoing support contract for $10k/yr to deal with business
> related bugfixing and support.
>
> The free help we give here is based on a community shared resource pool and
> we simply do not have enough to go around helping people with your type of
> issue.  Please feel free to report bugs if you can pinpoint one using a more
> realistic use case that does not involve DDosing your box with 10 million
> invites.
>
>
> On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Dear developers:
>>
>> Our product is close to sipping stage and we are doing stress tests on FS
>> servers. We have now hit a dead end with our test results and running out of
>> ideas. And the result is pointing to memory issues. We have ran our test on
>> both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests.
>>
>> Here is the test senario:
>>
>> Send 10,000,000 invites to our test server running FS 1.0.6 with fixed
>> amount of invites per second. The xml dialplan simply bridge the call to an
>> unknown/unregistered user like so (user/500001) and therefore FS would hang
>> up once it doesn't find the user. When running the test, FS is steady, but
>> the memory usage slowly piles up over time. The memory keeps adding up till
>> it hit the ceiling, then FS crash. I don't understand what could be holding
>> the memory and not releasing it since we are not doing anything special and
>> the calls are not being answered. And I forgot to mention, through out the
>> whole test, the CPU usage stays at about 35% .
>>
>> Please help
>>
>>
>> Besides FS native test, we have created our own application module to run
>> the same test. The app is written in C and basically what we did is to point
>> the xml dialplan to the custom app. Inside our app , we are simply doing the
>> same kind of test as the xml dialplan would do. The only thing we did for
>> the stress test is the following code:
>>
>> switch_core_session_execute_application(session, "bridge", "user/500001");
>>
>> Like the xml dialplan test, we are bridging the call to a unregistered
>> user, so the call would hang up right away if system doesn't find the user.
>> But the bizarre thing is this - the memory usage is about 7 ~ 8 times as
>> much as we are running only xml dialplan. This I don't understand. Our app
>> is calling the native FS C function to bridge the call , why would it use
>> more memory than the xml dialplan...
>>
>> your input would be very much appreciated.
>>
>> thanks
>>
>>
>> --
>> Henry Huang
>> aka bbhenry
>>
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
VoIP & Open Source software Consultant
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From linux4michelle at tamay-dogan.net  Wed Apr 14 09:23:35 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 14 Apr 2010 18:23:35 +0200
Subject: [Freeswitch-users] getting spammed from the list
In-Reply-To: <n2q191c3a031004140810u4e9d0569t1ac9381666726ea6@mail.gmail.com>
References: <4BC442FD.8090506@telemaque.fr>
	<20100413101616.GW12939@tamay-dogan.net>
	<4BC44A68.9040209@telemaque.fr>
	<20100413105326.GY12939@tamay-dogan.net>
	<y2r191c3a031004130853u3bd2cafejeac02d85c17dbe80@mail.gmail.com>
	<20100414145341.GC12939@tamay-dogan.net>
	<n2q191c3a031004140810u4e9d0569t1ac9381666726ea6@mail.gmail.com>
Message-ID: <20100414162335.GE12939@tamay-dogan.net>

Hello Anthony Minessale,

Am 2010-04-14 10:10:12, hacktest Du folgendes herunter:
> The problem was the mailing list software suffers from a 2gb log file
> limitation and our log rotate cronjob
> was paused somehow by accident during our last migration.

:-/   N.C.

It happen to me some times ago too...   And after 2 or 3 weeks I had the
problem how to split this 100 GByte munsters of logfiles...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France           itsystems at tdnet UG (haftungsbeschr?nkt)
Gesch. Michelle Konzack          Gesch. Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz               Kinzigstra?e 17
67100 Strasbourg/France         77694 Kehl/Germany
Tel: +33-6-61925193 mobil       Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From helmut.kuper at ewetel.de  Wed Apr 14 09:36:00 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 14 Apr 2010 18:36:00 +0200
Subject: [Freeswitch-users] Problem with SIP INFO (caller display
 update) after connect
In-Reply-To: <4BC5D726.5000508@ewetel.de>
References: <4BC5D726.5000508@ewetel.de>
Message-ID: <4BC5EEF0.7040207@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi again,

I found that the reason for this is maybe function send_display inf code
file switch_ivr_bridge.c.

At Line around 115 I replaced this:

	name = caller_profile->caller_id_name;
        number = caller_profile->caller_id_number;


with this:
        if (!(name = switch_channel_get_variable(caller_channel,
"effective_caller_id_name"))) {
            name = caller_profile->caller_id_name;
        }
        if (!(number = switch_channel_get_variable(caller_channel,
"effective_caller_id_number"))) {
            number = caller_profile->caller_id_number;
        }

Just for testing ... and it works as I wanted as long as I set
effective_caller_id_name and -number to the value I need.


Makes this sense to you? Or should openzap resp mod_openzap be enhanced
to detect "type of number" and add a 0 resp 00 to caller_id_number?


regards
helmut


On 14.04.2010 16:54, Helmut Kuper wrote:
> Hello,
> 
> 
> I have the following problem.
> A call comes in the following way:
> ISDN PRI->Sangoma->wanpipe->openzap->mod_openzap->freeswitch->B-Party
> 
> I use a national numbering plan, so caller's number e.g 044112345 is
> transmitted as 4412345 on isdn level.
> My dialplan adds two zeros to the effective_caller_id_name and -number.
> So it is now 0044112345. The B-Party displays this correctly. But as
> soon as B-Party picks up the caller's name switched back to 44112345
> which is quite bad.
> 
> I tried to set caller's name and number via effective_caller_id_... ,
> via origination_caller_id_... and via  origination_callee_id_... but
> nothing worked.
> 
> It looks like the sip info process after connecting two sessions is not
> effected by those channel variables mentioned above.
> 
> I use FS tunk (17073M)
> 
> Any ideas what is wrong here?
> 
> kind regards
> helmut
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EalE5W+o345Wc5n8LkW+BtY=
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From infos at madovsky.org  Wed Apr 14 08:35:40 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 14 Apr 2010 11:35:40 -0400
Subject: [Freeswitch-users] Freeswitch memory issue/question
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
Message-ID: <81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>

Henry Huang,

with 10 millions of invites I'm sure you have at least $175 to pay
to FS to help you !! :D
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, April 14, 2010 11:19 AM
  Subject: Re: [Freeswitch-users] Freeswitch memory issue/question


  We have a policy against load testing issues on this mailing list.

  What you are doing is most likely doing something wrong and we don't have time to debug it for you.
  We get way too many requests like this and there is a 99% result in proving user error and improper load testing.

  You can consider paid support from FreeSWITCH Solutions LLC for $175/hr per consultant and an ongoing support contract for $10k/yr to deal with business related bugfixing and support.

  The free help we give here is based on a community shared resource pool and we simply do not have enough to go around helping people with your type of issue.  Please feel free to report bugs if you can pinpoint one using a more realistic use case that does not involve DDosing your box with 10 million invites.



  On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com> wrote:

    Dear developers:

    Our product is close to sipping stage and we are doing stress tests on FS servers. We have now hit a dead end with our test results and running out of ideas. And the result is pointing to memory issues. We have ran our test on both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests. 

    Here is the test senario:

    Send 10,000,000 invites to our test server running FS 1.0.6 with fixed amount of invites per second. The xml dialplan simply bridge the call to an unknown/unregistered user like so (user/500001) and therefore FS would hang up once it doesn't find the user. When running the test, FS is steady, but the memory usage slowly piles up over time. The memory keeps adding up till it hit the ceiling, then FS crash. I don't understand what could be holding the memory and not releasing it since we are not doing anything special and the calls are not being answered. And I forgot to mention, through out the whole test, the CPU usage stays at about 35% . 

    Please help


    Besides FS native test, we have created our own application module to run the same test. The app is written in C and basically what we did is to point the xml dialplan to the custom app. Inside our app , we are simply doing the same kind of test as the xml dialplan would do. The only thing we did for the stress test is the following code: 

    switch_core_session_execute_application(session, "bridge", "user/500001");

    Like the xml dialplan test, we are bridging the call to a unregistered user, so the call would hang up right away if system doesn't find the user. But the bizarre thing is this - the memory usage is about 7 ~ 8 times as much as we are running only xml dialplan. This I don't understand. Our app is calling the native FS C function to bridge the call , why would it use more memory than the xml dialplan... 

    your input would be very much appreciated. 

    thanks 


    -- 
    Henry Huang
    aka bbhenry

    VoIP & Open Source software Consultant

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





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  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
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing 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 vhatz at kinetix.gr  Wed Apr 14 10:20:56 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Wed, 14 Apr 2010 20:20:56 +0300
Subject: [Freeswitch-users] Freeswitch memory issue/question
In-Reply-To: <81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
	<81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
Message-ID: <4BC5F978.5010607@kinetix.gr>

Henry's 10 millions of INVITEs are from a sipp test scenario, not from 
live commercial traffic.

On 14/4/10 6:35 ??, Madovsky wrote:
> Henry Huang,
> with 10 millions of invites I'm sure you have at least $175 to pay
> to FS to help you !! :D
>
>     ----- Original Message -----
>     *From:* Anthony Minessale <mailto:anthony.minessale at gmail.com>
>     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Wednesday, April 14, 2010 11:19 AM
>     *Subject:* Re: [Freeswitch-users] Freeswitch memory issue/question
>
>     We have a policy against load testing issues on this mailing list.
>
>     What you are doing is most likely doing something wrong and we
>     don't have time to debug it for you.
>     We get way too many requests like this and there is a 99% result
>     in proving user error and improper load testing.
>
>     You can consider paid support from FreeSWITCH Solutions LLC for
>     $175/hr per consultant and an ongoing support contract for $10k/yr
>     to deal with business related bugfixing and support.
>
>     The free help we give here is based on a community shared resource
>     pool and we simply do not have enough to go around helping people
>     with your type of issue.  Please feel free to report bugs if you
>     can pinpoint one using a more realistic use case that does not
>     involve DDosing your box with 10 million invites.
>
>
>     On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang
>     <b_ball_henry at hotmail.com <mailto:b_ball_henry at hotmail.com>> wrote:
>
>         Dear developers:
>
>         Our product is close to sipping stage and we are doing stress
>         tests on FS servers. We have now hit a dead end with our test
>         results and running out of ideas. And the result is pointing
>         to memory issues. We have ran our test on both FS 1.0.4 and
>         1.0.6 trunk with SIPP to do simply invite tests.
>
>         Here is the test senario:
>
>         Send 10,000,000 invites to our test server running FS 1.0.6
>         with fixed amount of invites per second. The xml dialplan
>         simply bridge the call to an unknown/unregistered user like so
>         (user/500001) and therefore FS would hang up once it doesn't
>         find the user. When running the test, FS is steady, but the
>         memory usage slowly piles up over time. The memory keeps
>         adding up till it hit the ceiling, then FS crash. I don't
>         understand what could be holding the memory and not releasing
>         it since we are not doing anything special and the calls are
>         not being answered. And I forgot to mention, through out the
>         whole test, the CPU usage stays at about 35% .
>
>         Please help
>
>
>         Besides FS native test, we have created our own application
>         module to run the same test. The app is written in C and
>         basically what we did is to point the xml dialplan to the
>         custom app. Inside our app , we are simply doing the same kind
>         of test as the xml dialplan would do. The only thing we did
>         for the stress test is the following code:
>
>         switch_core_session_execute_application(session, "bridge",
>         "user/500001");
>
>         Like the xml dialplan test, we are bridging the call to a
>         unregistered user, so the call would hang up right away if
>         system doesn't find the user. But the bizarre thing is this -
>         the memory usage is about 7 ~ 8 times as much as we are
>         running only xml dialplan. This I don't understand. Our app is
>         calling the native FS C function to bridge the call , why
>         would it use more memory than the xml dialplan...
>
>         your input would be very much appreciated.
>
>         thanks
>
>
>         -- 
>         Henry Huang
>         aka bbhenry
>
>         VoIP & Open Source software Consultant
>
>         _______________________________________________
>         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/
>     Twitter: http://twitter.com/FreeSWITCH_wire
>
>     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>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:+19193869900
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From pjintheusa at gmail.com  Wed Apr 14 10:26:34 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 14 Apr 2010 13:26:34 -0400
Subject: [Freeswitch-users] Freeswitch memory issue/question
In-Reply-To: <81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
	<81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
Message-ID: <v2l367751821004141026p7aeaff00vd6bc43e5250c6c88@mail.gmail.com>

Henry,

Realistically - what is your expected traffic? i.e. how big a community are
you building this system for? In your test - now many cps where you hitting
FS with?

On Wed, Apr 14, 2010 at 11:35 AM, Madovsky <infos at madovsky.org> wrote:

>  Henry Huang,
>
> with 10 millions of invites I'm sure you have at least $175 to pay
> to FS to help you !! :D
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, April 14, 2010 11:19 AM
> *Subject:* Re: [Freeswitch-users] Freeswitch memory issue/question
>
> We have a policy against load testing issues on this mailing list.
>
> What you are doing is most likely doing something wrong and we don't have
> time to debug it for you.
> We get way too many requests like this and there is a 99% result in proving
> user error and improper load testing.
>
> You can consider paid support from FreeSWITCH Solutions LLC for $175/hr per
> consultant and an ongoing support contract for $10k/yr to deal with business
> related bugfixing and support.
>
> The free help we give here is based on a community shared resource pool and
> we simply do not have enough to go around helping people with your type of
> issue.  Please feel free to report bugs if you can pinpoint one using a more
> realistic use case that does not involve DDosing your box with 10 million
> invites.
>
>
> On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Dear developers:
>>
>> Our product is close to sipping stage and we are doing stress tests on FS
>> servers. We have now hit a dead end with our test results and running out of
>> ideas. And the result is pointing to memory issues. We have ran our test on
>> both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests.
>>
>> Here is the test senario:
>>
>> Send 10,000,000 invites to our test server running FS 1.0.6 with fixed
>> amount of invites per second. The xml dialplan simply bridge the call to an
>> unknown/unregistered user like so (user/500001) and therefore FS would hang
>> up once it doesn't find the user. When running the test, FS is steady, but
>> the memory usage slowly piles up over time. The memory keeps adding up till
>> it hit the ceiling, then FS crash. I don't understand what could be holding
>> the memory and not releasing it since we are not doing anything special and
>> the calls are not being answered. And I forgot to mention, through out the
>> whole test, the CPU usage stays at about 35% .
>>
>> Please help
>>
>>
>> Besides FS native test, we have created our own application module to run
>> the same test. The app is written in C and basically what we did is to point
>> the xml dialplan to the custom app. Inside our app , we are simply doing the
>> same kind of test as the xml dialplan would do. The only thing we did for
>> the stress test is the following code:
>>
>> switch_core_session_execute_application(session, "bridge", "user/500001");
>>
>> Like the xml dialplan test, we are bridging the call to a unregistered
>> user, so the call would hang up right away if system doesn't find the user.
>> But the bizarre thing is this - the memory usage is about 7 ~ 8 times as
>> much as we are running only xml dialplan. This I don't understand. Our app
>> is calling the native FS C function to bridge the call , why would it use
>> more memory than the xml dialplan...
>>
>> your input would be very much appreciated.
>>
>> thanks
>>
>>
>> --
>> Henry Huang
>> aka bbhenry
>>
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Wed Apr 14 11:15:54 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 14 Apr 2010 11:15:54 -0700
Subject: [Freeswitch-users] Best low cost router/firewall
Message-ID: <ED788322-BA9C-4A8A-8CFE-9C9CF1082AA1@tlainvestments.com>

Any opinions?  I use pfSense, but need a simple consumer router for a client that works well with fs, sip, etc.

Thanks!
-Troy



From william.suffill at gmail.com  Wed Apr 14 11:27:09 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 14 Apr 2010 14:27:09 -0400
Subject: [Freeswitch-users] Best low cost router/firewall
In-Reply-To: <ED788322-BA9C-4A8A-8CFE-9C9CF1082AA1@tlainvestments.com>
References: <ED788322-BA9C-4A8A-8CFE-9C9CF1082AA1@tlainvestments.com>
Message-ID: <m2o6b65470d1004141127ia45c735ci5dff0ebe7b6484b9@mail.gmail.com>

I've had good luck with Linksys routers with 3rd party firmware. (OpenWRT or
DD-Wrt)

WRT54GS & WRT300N atm. Other routers are supported by the 3rd party
firmwares listed above as well.
The work nicely with the FS upnp firewall support.

-- W
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From m.sobkow at marketelsystems.com  Wed Apr 14 11:30:50 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 14 Apr 2010 12:30:50 -0600
Subject: [Freeswitch-users] origination_caller_id_name doesn't seem to be
	working
Message-ID: <4BC609DA.8050000@marketelsystems.com>

OriginateCmd is 
"{call_timeout=15,originate_timeout=15,origination_caller_id_name='Mark 
Test',origination_caller_id_number=3065551212}sofia/external/533 at rats.marketel 
'&erlang( pbx_called_cust pursuit at testsrv )'"

However, I'm not getting the origination caller id name displayed on the 
soft phone, even after answering the call.  Could this be a problem with 
the Asterisk gateway, or is there a problem with the implementation in 
Freeswitch?  Unfortunately I have to route all my calls through the 
Asterisk gateway, so I can't tell whether it's a problem with Freeswitch 
or not.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From anthony.minessale at gmail.com  Wed Apr 14 11:34:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Apr 2010 13:34:24 -0500
Subject: [Freeswitch-users] Freeswitch memory issue/question
In-Reply-To: <v2l367751821004141026p7aeaff00vd6bc43e5250c6c88@mail.gmail.com>
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>
	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com>
	<81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
	<v2l367751821004141026p7aeaff00vd6bc43e5250c6c88@mail.gmail.com>
Message-ID: <r2m191c3a031004141134zd8fb4a5fg8dc018f6ae7cee8e@mail.gmail.com>

1 hint,

you need enough ram to hold on to every dialog for several minutes even
after the call has ended per the RFC so you will see a large increase in
memory if you send millions of calls at the box.

On Wed, Apr 14, 2010 at 12:26 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Henry,
>
> Realistically - what is your expected traffic? i.e. how big a community are
> you building this system for? In your test - now many cps where you hitting
> FS with?
>
> On Wed, Apr 14, 2010 at 11:35 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  Henry Huang,
>>
>> with 10 millions of invites I'm sure you have at least $175 to pay
>> to FS to help you !! :D
>>
>> ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Wednesday, April 14, 2010 11:19 AM
>> *Subject:* Re: [Freeswitch-users] Freeswitch memory issue/question
>>
>> We have a policy against load testing issues on this mailing list.
>>
>> What you are doing is most likely doing something wrong and we don't have
>> time to debug it for you.
>> We get way too many requests like this and there is a 99% result in
>> proving user error and improper load testing.
>>
>> You can consider paid support from FreeSWITCH Solutions LLC for $175/hr
>> per consultant and an ongoing support contract for $10k/yr to deal with
>> business related bugfixing and support.
>>
>> The free help we give here is based on a community shared resource pool
>> and we simply do not have enough to go around helping people with your type
>> of issue.  Please feel free to report bugs if you can pinpoint one using a
>> more realistic use case that does not involve DDosing your box with 10
>> million invites.
>>
>>
>> On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>>
>>> Dear developers:
>>>
>>> Our product is close to sipping stage and we are doing stress tests on FS
>>> servers. We have now hit a dead end with our test results and running out of
>>> ideas. And the result is pointing to memory issues. We have ran our test on
>>> both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests.
>>>
>>> Here is the test senario:
>>>
>>> Send 10,000,000 invites to our test server running FS 1.0.6 with fixed
>>> amount of invites per second. The xml dialplan simply bridge the call to an
>>> unknown/unregistered user like so (user/500001) and therefore FS would hang
>>> up once it doesn't find the user. When running the test, FS is steady, but
>>> the memory usage slowly piles up over time. The memory keeps adding up till
>>> it hit the ceiling, then FS crash. I don't understand what could be holding
>>> the memory and not releasing it since we are not doing anything special and
>>> the calls are not being answered. And I forgot to mention, through out the
>>> whole test, the CPU usage stays at about 35% .
>>>
>>> Please help
>>>
>>>
>>> Besides FS native test, we have created our own application module to run
>>> the same test. The app is written in C and basically what we did is to point
>>> the xml dialplan to the custom app. Inside our app , we are simply doing the
>>> same kind of test as the xml dialplan would do. The only thing we did for
>>> the stress test is the following code:
>>>
>>> switch_core_session_execute_application(session, "bridge",
>>> "user/500001");
>>>
>>> Like the xml dialplan test, we are bridging the call to a unregistered
>>> user, so the call would hang up right away if system doesn't find the user.
>>> But the bizarre thing is this - the memory usage is about 7 ~ 8 times as
>>> much as we are running only xml dialplan. This I don't understand. Our app
>>> is calling the native FS C function to bridge the call , why would it use
>>> more memory than the xml dialplan...
>>>
>>> your input would be very much appreciated.
>>>
>>> thanks
>>>
>>>
>>> --
>>> Henry Huang
>>> aka bbhenry
>>>
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Wed Apr 14 11:35:05 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Apr 2010 13:35:05 -0500
Subject: [Freeswitch-users] origination_caller_id_name doesn't seem to
	be working
In-Reply-To: <4BC609DA.8050000@marketelsystems.com>
References: <4BC609DA.8050000@marketelsystems.com>
Message-ID: <7617BDDD-09AA-4C8A-BC55-51BFB87813EC@freeswitch.org>

chances are your provider sucks and doesn't like the CID in the RPID or PAID

/b

On Apr 14, 2010, at 1:30 PM, Mark Sobkow wrote:

> 
> However, I'm not getting the origination caller id name displayed on the 
> soft phone, even after answering the call.  Could this be a problem with 
> the Asterisk gateway, or is there a problem with the implementation in 
> Freeswitch?  Unfortunately I have to route all my calls through the 
> Asterisk gateway, so I can't tell whether it's a problem with Freeswitch 
> or not.



From infos at madovsky.org  Wed Apr 14 10:30:27 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 14 Apr 2010 13:30:27 -0400
Subject: [Freeswitch-users] Freeswitch memory issue/question
References: <i2q59ad9ca11004140311wf8aafe9amd8d97613af93911e@mail.gmail.com>	<h2r191c3a031004140819sd29c3c08lffc61bbd34c20f10@mail.gmail.com><81B35E81588E44B2A9A13EE81EBEAA47@MOBILEE1705>
	<4BC5F978.5010607@kinetix.gr>
Message-ID: <9104B253C6B44A7E89C9B3A87AF5508B@MOBILEE1705>

ho yes ? I didn't know ;)
  ----- Original Message ----- 
  From: Vlasis Hatzistavrou (KTI) 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, April 14, 2010 1:20 PM
  Subject: Re: [Freeswitch-users] Freeswitch memory issue/question


  Henry's 10 millions of INVITEs are from a sipp test scenario, not from live commercial traffic.

  On 14/4/10 6:35 ??, Madovsky wrote: 
    Henry Huang,

    with 10 millions of invites I'm sure you have at least $175 to pay
    to FS to help you !! :D
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, April 14, 2010 11:19 AM
      Subject: Re: [Freeswitch-users] Freeswitch memory issue/question


      We have a policy against load testing issues on this mailing list.

      What you are doing is most likely doing something wrong and we don't have time to debug it for you.
      We get way too many requests like this and there is a 99% result in proving user error and improper load testing.

      You can consider paid support from FreeSWITCH Solutions LLC for $175/hr per consultant and an ongoing support contract for $10k/yr to deal with business related bugfixing and support.

      The free help we give here is based on a community shared resource pool and we simply do not have enough to go around helping people with your type of issue.  Please feel free to report bugs if you can pinpoint one using a more realistic use case that does not involve DDosing your box with 10 million invites.



      On Wed, Apr 14, 2010 at 5:11 AM, Henry Huang <b_ball_henry at hotmail.com> wrote:

        Dear developers:

        Our product is close to sipping stage and we are doing stress tests on FS servers. We have now hit a dead end with our test results and running out of ideas. And the result is pointing to memory issues. We have ran our test on both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests. 

        Here is the test senario:

        Send 10,000,000 invites to our test server running FS 1.0.6 with fixed amount of invites per second. The xml dialplan simply bridge the call to an unknown/unregistered user like so (user/500001) and therefore FS would hang up once it doesn't find the user. When running the test, FS is steady, but the memory usage slowly piles up over time. The memory keeps adding up till it hit the ceiling, then FS crash. I don't understand what could be holding the memory and not releasing it since we are not doing anything special and the calls are not being answered. And I forgot to mention, through out the whole test, the CPU usage stays at about 35% . 

        Please help


        Besides FS native test, we have created our own application module to run the same test. The app is written in C and basically what we did is to point the xml dialplan to the custom app. Inside our app , we are simply doing the same kind of test as the xml dialplan would do. The only thing we did for the stress test is the following code: 

        switch_core_session_execute_application(session, "bridge", "user/500001");

        Like the xml dialplan test, we are bridging the call to a unregistered user, so the call would hang up right away if system doesn't find the user. But the bizarre thing is this - the memory usage is about 7 ~ 8 times as much as we are running only xml dialplan. This I don't understand. Our app is calling the native FS C function to bridge the call , why would it use more memory than the xml dialplan... 

        your input would be very much appreciated. 

        thanks 


        -- 
        Henry Huang
        aka bbhenry

        VoIP & Open Source software Consultant

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





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      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
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------
      _______________________________________________
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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


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

------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing 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 david.ponzone at gmail.com  Wed Apr 14 12:19:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 14 Apr 2010 21:19:16 +0200
Subject: [Freeswitch-users] origination_caller_id_name doesn't seem to
	be working
In-Reply-To: <4BC609DA.8050000@marketelsystems.com>
References: <4BC609DA.8050000@marketelsystems.com>
Message-ID: <8DB80E6B-5060-43CF-9A4C-83A6665BA7A0@gmail.com>

Try playing wth sip_cid_type.
	
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/04/2010 ? 20:30, Mark Sobkow a ?crit :

> OriginateCmd is
> "{call_timeout 
> =15,originate_timeout=15,origination_caller_id_name='Mark
> Test',origination_caller_id_number=3065551212}sofia/external/533 at rats.marketel
> '&erlang( pbx_called_cust pursuit at testsrv )'"
>
> However, I'm not getting the origination caller id name displayed on  
> the
> soft phone, even after answering the call.  Could this be a problem  
> with
> the Asterisk gateway, or is there a problem with the implementation in
> Freeswitch?  Unfortunately I have to route all my calls through the
> Asterisk gateway, so I can't tell whether it's a problem with  
> Freeswitch
> or not.
>
> -- 
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.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 mike at jerris.com  Wed Apr 14 12:36:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 15:36:07 -0400
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <v2rd021d3be1004140750refd7f5cdi7c95ca508bbed1e8@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
	<k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
	<CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>
	<v2rd021d3be1004140750refd7f5cdi7c95ca508bbed1e8@mail.gmail.com>
Message-ID: <058900EC-80E9-4312-B302-07DC750EFB32@jerris.com>

This does not make any sense.  It limits outbound connections based on source port?  rport would make us tell the guy the port he DID come from, so the nat mapping is in place and working because we got the packet from that port.

On Apr 14, 2010, at 10:50 AM, Jonas Gauffin wrote:

> Most of my customers have forwarded ports in their firewalls/routers. Those ports will not be used if rport is turned on.
> 
> I currently got a problem with a customer  who is using an old Windows 2000 server as firewall/router. It wont let packets through when using rport. It would work if rport was turned off and all communication was done using the configured port.
> 
> On Wed, Apr 14, 2010 at 3:53 PM, Brian West <brian at freeswitch.org> wrote:
> You want rport otherwise NAT HELL will begin.
> 
> /b
> 
> On Apr 14, 2010, at 8:11 AM, Jonas Gauffin wrote:
> 
> > Well, yes. But snom can only turn off rport through provisioning and only in the latest firmware.
> >
> > It would be great if it could be turned off in FS too, per user account.
> 

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From mike at jerris.com  Wed Apr 14 12:38:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 15:38:23 -0400
Subject: [Freeswitch-users] SVN read only mirror ???
In-Reply-To: <s2v367751821004140619k135dcd05i9af826b66b5ccef2@mail.gmail.com>
References: <s2v367751821004140619k135dcd05i9af826b66b5ccef2@mail.gmail.com>
Message-ID: <6E511D4A-8E86-419D-BE61-E0690398CE99@jerris.com>

I ran into an issue with the mirror, it is being maintained but a bit behind at the moment.  Should be back up to date shortly.

Mike

On Apr 14, 2010, at 9:19 AM, Phillip Jones wrote:

> Hi there,
> 
> I read previously on this list that the devs would try and maintain a read only mirror of FreeSWITCH on SVN. Was this possible?
> 
>  When I use the "Latest Build Version:" link from the front page I get:
> 
> "moose penis
> 
> NOT IT IS NOT HACKED... ITS CALLED A JOKE and some advice to use FreeSWITCH 1.0.6"
> 
> 
> Nice! 
> 
> So the question is, is the SVN mirror being maintained or are we using GIT exclusively now?
> 
> Pj


From nandy1925 at gmail.com  Wed Apr 14 16:01:53 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 15 Apr 2010 07:01:53 +0800
Subject: [Freeswitch-users] Best low cost router/firewall
In-Reply-To: <m2o6b65470d1004141127ia45c735ci5dff0ebe7b6484b9@mail.gmail.com>
References: <ED788322-BA9C-4A8A-8CFE-9C9CF1082AA1@tlainvestments.com>
	<m2o6b65470d1004141127ia45c735ci5dff0ebe7b6484b9@mail.gmail.com>
Message-ID: <y2z7d0bfd8c1004141601n29d6f2c9kf3436d5302d5fb18@mail.gmail.com>

the asus wlg-520gu also works nicely with DD-WRT. one thing nice w/ the unit
- it has USB port for printer sharing and i was able to use LAN ports 2, 3,
and 4 as a VLAN switch.


On Thu, Apr 15, 2010 at 2:27 AM, William Suffill
<william.suffill at gmail.com>wrote:

> I've had good luck with Linksys routers with 3rd party firmware. (OpenWRT
> or DD-Wrt)
>
> WRT54GS & WRT300N atm. Other routers are supported by the 3rd party
> firmwares listed above as well.
> The work nicely with the FS upnp firewall support.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vfclists at googlemail.com  Wed Apr 14 17:32:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Thu, 15 Apr 2010 01:32:36 +0100
Subject: [Freeswitch-users] Most appropriate mapping between Asterisk CDR
	and XML CDR
Message-ID: <r2r7abab2411004141732x25c501fbqa97738db2b8ef547@mail.gmail.com>

I am trying to work out the most appropriate mapping between the Asterisk
CDR and Freeswitch xml_cdr as an app of mine is based on the Asterisk CDR.

The xml cdr is quite large and some of the values are repeated in the
different sections.
This is the nearest I have come to, I have omitted the full paths to the
values, but would like to use the right paths if they are the most
meaningful.

The example displays t a call from extension 1001 to an external number
08451234567


  calldate <start_stamp>2010-04-14%2023%3A08%3A10</start_stamp> or
<start_epoch>1271282890</start_epoch> or
<created_time>1271282890796875</created_time>
  clid <caller_id_name>Booth1</caller_id_name> +
<caller_id_number>1001</caller_id_number>
  src <ani>1001</ani> or <username>1001</username>
  dst <destination_number>08451234567</destination_number>
  dcontext <context>default</context>
  channel
<bridge_channel>sofia/internal/1001%40192.168.1.133</bridge_channel>
or<chan_name>sofia/internal/1001 at 192.168.1.133</chan_name>
  dstchannel <chan_name>sofia/external/08451234567</chan_name>
  lastapp <application app_name="bridge" app_data="sofia/gateway/
provider.co.uk/08451234567"></application>
  lastdata <chan_name>sofia/external/08451234567</chan_name>
  duration <duration>46</duration>
  billsec   <billsec>18</billsec>
  disposition <endpoint_disposition>ANSWER</endpoint_disposition>?
  amaflags bigint,
  accountcode  -  <accountcode>1001</accountcode>
  uniqueid character uuid=ce234b00-157a-456f-97be-62d15a53b0c9

Is there some guide to which values are more appropriate? I checked the
templates for cdr_csv but the look rather simple to me, being as ever a
sucker for complexity.

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From brian at freeswitch.org  Wed Apr 14 17:37:26 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Apr 2010 19:37:26 -0500
Subject: [Freeswitch-users] Most appropriate mapping between Asterisk
	CDR and XML CDR
In-Reply-To: <r2r7abab2411004141732x25c501fbqa97738db2b8ef547@mail.gmail.com>
References: <r2r7abab2411004141732x25c501fbqa97738db2b8ef547@mail.gmail.com>
Message-ID: <6DA20671-2F30-427F-8102-8C76AF77AE0C@freeswitch.org>

Their already exists a default on cdr.conf.xml that shows you the exact ones that map as close as possible.

<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>


Its already in cdr_csv.conf.xml

/b

On Apr 14, 2010, at 7:32 PM, Frank Church wrote:

> I am trying to work out the most appropriate mapping between the Asterisk CDR and Freeswitch xml_cdr as an app of mine is based on the Asterisk CDR.
> 
> The xml cdr is quite large and some of the values are repeated in the different sections.
> This is the nearest I have come to, I have omitted the full paths to the values, but would like to use the right paths if they are the most meaningful.
> 
> The example displays t a call from extension 1001 to an external number 08451234567
> 
> 
>   calldate <start_stamp>2010-04-14%2023%3A08%3A10</start_stamp> or <start_epoch>1271282890</start_epoch> or  <created_time>1271282890796875</created_time>
>   clid <caller_id_name>Booth1</caller_id_name> + <caller_id_number>1001</caller_id_number>
>   src <ani>1001</ani> or <username>1001</username>
>   dst <destination_number>08451234567</destination_number>
>   dcontext <context>default</context>
>   channel     <bridge_channel>sofia/internal/1001%40192.168.1.133</bridge_channel> or<chan_name>sofia/internal/1001 at 192.168.1.133</chan_name> 
>   dstchannel <chan_name>sofia/external/08451234567</chan_name>
>   lastapp <application app_name="bridge" app_data="sofia/gateway/provider.co.uk/08451234567"></application>
>   lastdata <chan_name>sofia/external/08451234567</chan_name>
>   duration <duration>46</duration>
>   billsec   <billsec>18</billsec>
>   disposition <endpoint_disposition>ANSWER</endpoint_disposition>?
>   amaflags bigint,
>   accountcode  -  <accountcode>1001</accountcode>
>   uniqueid character uuid=ce234b00-157a-456f-97be-62d15a53b0c9
>   
> Is there some guide to which values are more appropriate? I checked the templates for cdr_csv but the look rather simple to me, being as ever a sucker for complexity.
> 
> -- 
> Frank Church
> 
> =======================
> http://devblog.brahmancreations.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 mike at jerris.com  Wed Apr 14 20:04:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Apr 2010 23:04:12 -0400
Subject: [Freeswitch-users] Problem with SIP INFO (caller display
	update) after connect
In-Reply-To: <4BC5EEF0.7040207@ewetel.de>
References: <4BC5D726.5000508@ewetel.de> <4BC5EEF0.7040207@ewetel.de>
Message-ID: <6355C3CD-F5C6-46A2-B8E3-7DA835F16D97@jerris.com>

I think we really need better freeswitch handling of ton and number translation up in freeswitch somewhere, maybe a new app?  I have never come to terms personally of where exactly the best place is for this.

Mike

On Apr 14, 2010, at 12:36 PM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi again,
> 
> I found that the reason for this is maybe function send_display inf code
> file switch_ivr_bridge.c.
> 
> At Line around 115 I replaced this:
> 
> 	name = caller_profile->caller_id_name;
>        number = caller_profile->caller_id_number;
> 
> 
> with this:
>        if (!(name = switch_channel_get_variable(caller_channel,
> "effective_caller_id_name"))) {
>            name = caller_profile->caller_id_name;
>        }
>        if (!(number = switch_channel_get_variable(caller_channel,
> "effective_caller_id_number"))) {
>            number = caller_profile->caller_id_number;
>        }
> 
> Just for testing ... and it works as I wanted as long as I set
> effective_caller_id_name and -number to the value I need.
> 
> 
> Makes this sense to you? Or should openzap resp mod_openzap be enhanced
> to detect "type of number" and add a 0 resp 00 to caller_id_number?
> 
> 
> regards
> helmut
> 
> 
> On 14.04.2010 16:54, Helmut Kuper wrote:
>> Hello,
>> 
>> 
>> I have the following problem.
>> A call comes in the following way:
>> ISDN PRI->Sangoma->wanpipe->openzap->mod_openzap->freeswitch->B-Party
>> 
>> I use a national numbering plan, so caller's number e.g 044112345 is
>> transmitted as 4412345 on isdn level.
>> My dialplan adds two zeros to the effective_caller_id_name and -number.
>> So it is now 0044112345. The B-Party displays this correctly. But as
>> soon as B-Party picks up the caller's name switched back to 44112345
>> which is quite bad.
>> 
>> I tried to set caller's name and number via effective_caller_id_... ,
>> via origination_caller_id_... and via  origination_callee_id_... but
>> nothing worked.
>> 
>> It looks like the sip info process after connecting two sessions is not
>> effected by those channel variables mentioned above.
>> 
>> I use FS tunk (17073M)
>> 
>> Any ideas what is wrong here?
>> 
>> kind regards
>> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> 
> iD8DBQFLxe7w4tZeNddg3dwRAiPYAJ9t4hyOF0kukgFfnUzHsLHbas0QfACfR2zN
> EalE5W+o345Wc5n8LkW+BtY=
> =e3G6
> -----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 talk2ram at gmail.com  Wed Apr 14 20:45:17 2010
From: talk2ram at gmail.com (ram)
Date: Thu, 15 Apr 2010 09:15:17 +0530
Subject: [Freeswitch-users] Multi-tenanting
In-Reply-To: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>
References: <l2l1c9955401004091542t7a533c37ye73cd5d8aab49c50@mail.gmail.com>
Message-ID: <o2vb74751491004142045p8f4baf8fkf2e9d6d39e11949f@mail.gmail.com>

Iam running test bed of Multi-tenant System

but each tenant use different instance..in single Server

Ram

On Sat, Apr 10, 2010 at 4:12 AM, Kenneth Noisewater <noisewaterphd at gmail.com
> wrote:

> Hi All,
>
> I'm just a few days into my FreeSwitch investigation, and so far I have to
> say, it seems almost too good to be true! Kudos to everyone involved. I've
> got  quite a bit of experience with other systems, ranging from Asterisk to
> Mitel/Shoretel type systems, and FreeSwitch is really looking good.
>
> Anyway...
>
> So pouring over configs, it seems it would be really simple to set up a
> multi tenant system with the whole 'domain' concept. Does this domain model
> maintain good seperation throughout the system? Is there anything to be
> aware of in doing a multi tenant setup?
>
> Thanks,
>
> Kenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Wed Apr 14 23:21:51 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 15 Apr 2010 10:51:51 +0430
Subject: [Freeswitch-users] mod_managed
Message-ID: <p2g1fdf92721004142321ib6dabcbbv17195cbe9e9b77b@mail.gmail.com>

Hi
can someone please explain, step by step, how a I can develop a module in
.net?
Thanks
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From jonas.gauffin at gmail.com  Thu Apr 15 00:15:40 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 15 Apr 2010 09:15:40 +0200
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <058900EC-80E9-4312-B302-07DC750EFB32@jerris.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
	<k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
	<CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>
	<v2rd021d3be1004140750refd7f5cdi7c95ca508bbed1e8@mail.gmail.com>
	<058900EC-80E9-4312-B302-07DC750EFB32@jerris.com>
Message-ID: <i2rd021d3be1004150015m6859336bi39d754829334019b@mail.gmail.com>

It doesnt make sense. But the fact is that the FW/router assigns another
port than the requested one.

I found a setting in latest firmware for snom which disables rport. But when
turned off, it stops using the external IP that stun found. Turning off
RPORT should not turn of external ip detection, right?

My FS server is on a public IP. Can I tell FS to always send replies to
received-ip instead of the one specified in the request (if the specified ip
is a RFC1918 ip?)

On Wed, Apr 14, 2010 at 9:36 PM, Michael Jerris <mike at jerris.com> wrote:

> This does not make any sense.  It limits outbound connections based on
> source port?  rport would make us tell the guy the port he DID come from, so
> the nat mapping is in place and working because we got the packet from that
> port.
>
> On Apr 14, 2010, at 10:50 AM, Jonas Gauffin wrote:
>
> Most of my customers have forwarded ports in their firewalls/routers. Those
> ports will not be used if rport is turned on.
>
> I currently got a problem with a customer  who is using an old Windows 2000
> server as firewall/router. It wont let packets through when using rport. It
> would work if rport was turned off and all communication was done using the
> configured port.
>
> On Wed, Apr 14, 2010 at 3:53 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You want rport otherwise NAT HELL will begin.
>>
>> /b
>>
>> On Apr 14, 2010, at 8:11 AM, Jonas Gauffin wrote:
>>
>> > Well, yes. But snom can only turn off rport through provisioning and
>> only in the latest firmware.
>> >
>> > It would be great if it could be turned off in FS too, per user account.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Thu Apr 15 01:40:29 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 15 Apr 2010 10:40:29 +0200
Subject: [Freeswitch-users] [INFO] User-Agent in mod_sofia, and git-builds.
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557767736A@cooper>

I just wanted to share an experience I had with a git-build and Avaya Communications Manager PBX. It seems it can't handle the new User-Agent values correctly, and this started to occur after doing git builds. In the end I  found that the ":" in the timestamp was what broke the INVITE.

After overriding the user-agent in the sofia configuration it started to work again.

Probably it's a Avaya problem - but I just wanted to inform the other people on the list about this.

This is how my User-Agent looked like: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700

Regards,

Peter Olsson
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From lakindia89 at gmail.com  Thu Apr 15 04:11:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 15 Apr 2010 16:41:29 +0530
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
Message-ID: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>

Hi all,
I have downloaded the latest freeswitch 1.0.6.
I configured my sangoma card with libpri as told in
                http://wiki.freeswitch.org/wiki/Openzap.sangoma.libpri
I use wanpipe-3.5.10.10 and libpri-1.4.10.2.

openzap.conf
[span wanpipe PRI_1]
name => OpenZAP
number => 1
trunk_type => e1
b-channel => 1:1-15
b-channel => 1:17-31
[span wanpipe PRI_2]
name => OpenZAP
number => 2
trunk_type => e1
b-channel => 2:1-15
b-channel => 2:17-31

openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="1"/>
</settings>
<libpri_spans>
<span name="PRI_1">
<param name="node" value="cpe"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="l1" value="alaw"/>
</span>
<span name="PRI_2">
<param name="node" value="net"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="l1" value="alaw"/>
</span>
</libpri_spans>
</configuration>

I started wanrouter and after that I started freeswitch.
I got core dumped immediately after saying auto-loaded libpri.
Here is the freeswitch log.

http://pastebin.freeswitch.org/12712
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From lakindia89 at gmail.com  Thu Apr 15 04:23:06 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 15 Apr 2010 16:53:06 +0530
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
Message-ID: <w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>

I tried to analyze the core dump and got the following.

gdb ./freeswitch core.28775

warning: Can't read pathname for load map: Input/output error.
Reading symbols from /lib/i686/cmov/libm.so.6...done.
Loaded symbols for /lib/i686/cmov/libm.so.6
Reading symbols from /usr/local/freeswitch/lib/libfreeswitch.so.1...done.
Loaded symbols for /usr/local/freeswitch/lib/libfreeswitch.so.1
Reading symbols from /lib/i686/cmov/librt.so.1...done.
Loaded symbols for /lib/i686/cmov/librt.so.1
Reading symbols from /lib/i686/cmov/libdl.so.2...done.
Loaded symbols for /lib/i686/cmov/libdl.so.2
Reading symbols from /lib/i686/cmov/libcrypt.so.1...done.
Loaded symbols for /lib/i686/cmov/libcrypt.so.1
Reading symbols from /lib/i686/cmov/libpthread.so.0...done.
Loaded symbols for /lib/i686/cmov/libpthread.so.0
Reading symbols from /lib/libncurses.so.5...done.
Loaded symbols for /lib/libncurses.so.5
Reading symbols from /lib/i686/cmov/libc.so.6...done.
Loaded symbols for /lib/i686/cmov/libc.so.6
Reading symbols from /lib/ld-linux.so.2...done.
Loaded symbols for /lib/ld-linux.so.2
Reading symbols from /usr/lib/libstdc++.so.6...done.
Loaded symbols for /usr/lib/libstdc++.so.6
Reading symbols from /lib/libgcc_s.so.1...done.
Loaded symbols for /lib/libgcc_s.so.1
Reading symbols from /usr/local/freeswitch/mod/mod_console.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_console.so
Reading symbols from /usr/local/freeswitch/mod/mod_logfile.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_logfile.so
Reading symbols from /usr/local/freeswitch/mod/mod_enum.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_enum.so
Reading symbols from /usr/local/freeswitch/mod/mod_cdr_csv.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_cdr_csv.so
Reading symbols from /usr/local/freeswitch/mod/mod_event_socket.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_event_socket.so
Reading symbols from /usr/local/freeswitch/mod/mod_sofia.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_sofia.so
Reading symbols from /usr/local/freeswitch/mod/mod_loopback.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_loopback.so
Reading symbols from /usr/local/freeswitch/mod/mod_openzap.so...done.
Loaded symbols for /usr/local/freeswitch/mod/mod_openzap.so
Reading symbols from /usr/local/freeswitch/lib/libopenzap.so.1...done.
Loaded symbols for /usr/local/freeswitch/lib/libopenzap.so.1
Reading symbols from /usr/local/freeswitch/mod/ozmod_wanpipe.so...done.
Loaded symbols for /usr/local/freeswitch/mod/ozmod_wanpipe.so
Reading symbols from /usr/lib/libsangoma.so.3...done.
Loaded symbols for /usr/lib/libsangoma.so.3
Reading symbols from /usr/local/freeswitch/mod/ozmod_libpri.so...done.
Loaded symbols for /usr/local/freeswitch/mod/ozmod_libpri.so
Reading symbols from /usr/lib/libpri.so.1.4...done.
Loaded symbols for /usr/lib/libpri.so.1.4
Core was generated by `./freeswitch'.
Program terminated with signal 11, Segmentation fault.
[New process 28795]
[New process 28775]
[New process 28794]
[New process 28793]
[New process 28792]
[New process 28791]
[New process 28790]
[New process 28789]
[New process 28788]
[New process 28787]
[New process 28786]
[New process 28785]
[New process 28784]
[New process 28781]
[New process 28780]
[New process 28779]
[New process 28778]
[New process 28777]
[New process 28776]
#0  0xb6f08ee5 in lpwrap_init_pri (spri=0x9fdb8a0, span=0x9e7cb38,
dchan=0x0, swtype=5,
    node=2, debug=0) at src/ozmod/ozmod_libpri/lpwrap_pri.c:180
180             if ((spri->pri = pri_new_cb(spri->dchan->sockfd, node,
swtype, __pri_lpwrap_read, __pri_lpwrap_write, spri))){

On Thu, Apr 15, 2010 at 4:41 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> I have downloaded the latest freeswitch 1.0.6.
> I configured my sangoma card with libpri as told in
>                 http://wiki.freeswitch.org/wiki/Openzap.sangoma.libpri
> I use wanpipe-3.5.10.10 and libpri-1.4.10.2.
>
> openzap.conf
> [span wanpipe PRI_1]
> name => OpenZAP
> number => 1
> trunk_type => e1
> b-channel => 1:1-15
> b-channel => 1:17-31
> [span wanpipe PRI_2]
> name => OpenZAP
> number => 2
> trunk_type => e1
> b-channel => 2:1-15
> b-channel => 2:17-31
>
> openzap.conf.xml:
> <configuration name="openzap.conf" description="OpenZAP Configuration">
> <settings>
> <param name="debug" value="1"/>
> </settings>
> <libpri_spans>
> <span name="PRI_1">
> <param name="node" value="cpe"/>
> <param name="switch" value="euroisdn"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> <param name="l1" value="alaw"/>
> </span>
> <span name="PRI_2">
> <param name="node" value="net"/>
> <param name="switch" value="euroisdn"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> <param name="l1" value="alaw"/>
> </span>
> </libpri_spans>
> </configuration>
>
> I started wanrouter and after that I started freeswitch.
> I got core dumped immediately after saying auto-loaded libpri.
> Here is the freeswitch log.
>
> http://pastebin.freeswitch.org/12712
>
>
>
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From null at invalid.name  Thu Apr 15 05:04:47 2010
From: null at invalid.name (Dan Lane)
Date: Thu, 15 Apr 2010 13:04:47 +0100
Subject: [Freeswitch-users] DTMF issues
Message-ID: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>

I've got an issue where DTMF has stopped working with outbound calls
(DTMF works on inbound calls from them) via one of my service
providers (still works fine with several other providers) so I assumed
it was a change the termination provider had made and duly reported it
to them.

Their response was:

"We have analysed rtp from xlite to freeswitch to us and xlite to us
1. volume is being made much quieter by freeswitch, i.e. standard dtmf
volume 10 is being re-written to 7
2. dtmf is shorter, i.e messages from phones are being re-written to a
really short version
3. timestamp discontinuity  Audio to RTP conversion point. timestamps
are indicating missing packets received via freeswitch"

The service provider is running a mixture of Cisco AS 5300's and
5350xm's, I don't think it's a bug in Freeswitch (I have a number of
Freeswitch boxes that are all affected running different versions from
1.0.4pre7 to r17188) as DTMF was working once upon a time... can
anyone think of anything obvious that either I or the service provider
could be missing?

Thanks.


From scaram at hotmail.de  Thu Apr 15 05:08:11 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Thu, 15 Apr 2010 14:08:11 +0200
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <9E1A4D19-A8C4-41A3-AD2B-E324EA31B43B@jerris.com>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>,
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>,
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>,
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>,
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>,
	<i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>,
	<9E1A4D19-A8C4-41A3-AD2B-E324EA31B43B@jerris.com>
Message-ID: <SNT133-w5393F1951FBF79FB8E145EA00F0@phx.gbl>


Now I am using stable release 1.0.6 and have a MEDIA_TIMEOUT again, because function rtp_common_read counts missed packets.
During a long session of several hours the amount of missed packets exceeds the limit. 

Now I changed my .\conf\sip_profiles\internal.xml and set the rtp timeout to 0, which means no timeout.
  
  set <param name="rtp-timeout-sec" value="0"/>

My usecase is to have long duration calls. Can I do this without cross-effects, or is not wise to set rtp timout to 0? 

> From: mike at jerris.com
> Date: Wed, 14 Apr 2010 09:08:32 -0400
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
> 
> That being said, for a change things have been somewhat quiet after the 1.0.6 release.  There have been some bug fixes that went in after, but not the usual frenzy of reports right after a release.
> 
> Mike
> 
> On Apr 14, 2010, at 7:23 AM, Steven Ayre wrote:
> 
> > We are no longer on SVN, we have now moved to GIT for the 1.0.6 release.
> > 
> > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH
> > 
> > Note that the latest most stable version is always likely to be latest
> > GIT version not the latest tagged released, as there will often be
> > bugs present in the releases that are fixed in the latest GIT version.
> > Tagged releases have more to do with the introduction of new features
> > (such as G729 support in 1.0.6) than any guarantee of stability.
> > 
> > -Steve
> > 
> > 
> > On 14 April 2010 09:00, Francisco Scaramanga <scaram at hotmail.de> wrote:
> >> 
> >> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
> >> I thought it is a release candidate, isn't it? Is it safe to use it in
> >> production?
> >> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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://redirect.gimas.net/?n=M1004xjajah2
?ber Messenger g?nstiger telefonieren? Sagen Sie "Ja" zu JAJAH!
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From yehavi.bourvine at gmail.com  Thu Apr 15 06:25:15 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 15 Apr 2010 16:25:15 +0300
Subject: [Freeswitch-users] DTMF issues
In-Reply-To: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
References: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
Message-ID: <y2ld361ca7b1004150625hcf2e195ap1cdfdc5ef13b842b@mail.gmail.com>

we have a Cisco-2,811 gateway and also have problems with DTMF. Turnning on
debug on Cisco shows that it recgonise the DTMF but do not pass them along.
Our solution was to set the profile toward that Cisco to INFO instead of
RFC-2833 and this solved this issue.

                   __Yehavi:

2010/4/15 Dan Lane <null at invalid.name>

> I've got an issue where DTMF has stopped working with outbound calls
> (DTMF works on inbound calls from them) via one of my service
> providers (still works fine with several other providers) so I assumed
> it was a change the termination provider had made and duly reported it
> to them.
>
> Their response was:
>
> "We have analysed rtp from xlite to freeswitch to us and xlite to us
> 1. volume is being made much quieter by freeswitch, i.e. standard dtmf
> volume 10 is being re-written to 7
> 2. dtmf is shorter, i.e messages from phones are being re-written to a
> really short version
> 3. timestamp discontinuity  Audio to RTP conversion point. timestamps
> are indicating missing packets received via freeswitch"
>
> The service provider is running a mixture of Cisco AS 5300's and
> 5350xm's, I don't think it's a bug in Freeswitch (I have a number of
> Freeswitch boxes that are all affected running different versions from
> 1.0.4pre7 to r17188) as DTMF was working once upon a time... can
> anyone think of anything obvious that either I or the service provider
> could be missing?
>
> 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 pjintheusa at gmail.com  Thu Apr 15 06:30:16 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 15 Apr 2010 09:30:16 -0400
Subject: [Freeswitch-users] mod_managed
In-Reply-To: <p2g1fdf92721004142321ib6dabcbbv17195cbe9e9b77b@mail.gmail.com>
References: <p2g1fdf92721004142321ib6dabcbbv17195cbe9e9b77b@mail.gmail.com>
Message-ID: <i2k367751821004150630l6606c153t16c86f25d360ad76@mail.gmail.com>

How far along are you?

Have to followed http://wiki.freeswitch.org/wiki/Mod_managed and are you
loading the mod? Have you tried a simple hello world app?

Let me know where you are stuck I will be glad to help.

On Thu, Apr 15, 2010 at 2:21 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> can someone please explain, step by step, how a I can develop a module in
> .net?
> 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  Thu Apr 15 06:39:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 09:39:41 -0400
Subject: [Freeswitch-users] OT: Snom phones reporting incorrect port in
	Contact/Via
In-Reply-To: <i2rd021d3be1004150015m6859336bi39d754829334019b@mail.gmail.com>
References: <h2od021d3be1004132345l9ec3c1d6keb303a3a8958fee@mail.gmail.com>
	<u2qd021d3be1004140105rfcfc56eer67c5a3cd52e40536@mail.gmail.com>
	<n2yd021d3be1004140249v81c9fcffn1c38f448758f472c@mail.gmail.com>
	<00C84A31-1CC7-4CBF-8259-F621A0D6D39E@jerris.com>
	<k2td021d3be1004140611h80b01570t6105e4a8cb2c56ad@mail.gmail.com>
	<CE953CDF-3F4C-484F-A29F-95602E1448F3@freeswitch.org>
	<v2rd021d3be1004140750refd7f5cdi7c95ca508bbed1e8@mail.gmail.com>
	<058900EC-80E9-4312-B302-07DC750EFB32@jerris.com>
	<i2rd021d3be1004150015m6859336bi39d754829334019b@mail.gmail.com>
Message-ID: <044D7ECA-88BA-48FD-8AFC-6E5299F54C03@jerris.com>

The one requested?  you seem to be indicating that the firewall uses different nat mappings for inbound vs outbound.  When using rport, it doesn't matter what the local port on the phone is.  The phone make a connection to the outside world, the firewall maps it to some port, we don't care what.  When we respond, we respond to the port we got it from, and we tell the guy that port.  it doesn't matter what port the client or anyone else thought it was, it only matters what the firewall chose.

Mike

On Apr 15, 2010, at 3:15 AM, Jonas Gauffin wrote:

> It doesnt make sense. But the fact is that the FW/router assigns another port than the requested one.
> 
> I found a setting in latest firmware for snom which disables rport. But when turned off, it stops using the external IP that stun found. Turning off RPORT should not turn of external ip detection, right?
> 
> My FS server is on a public IP. Can I tell FS to always send replies to received-ip instead of the one specified in the request (if the specified ip is a RFC1918 ip?)
> 
> On Wed, Apr 14, 2010 at 9:36 PM, Michael Jerris <mike at jerris.com> wrote:
> This does not make any sense.  It limits outbound connections based on source port?  rport would make us tell the guy the port he DID come from, so the nat mapping is in place and working because we got the packet from that port.
> 
> On Apr 14, 2010, at 10:50 AM, Jonas Gauffin wrote:
> 
>> Most of my customers have forwarded ports in their firewalls/routers. Those ports will not be used if rport is turned on.
>> 
>> I currently got a problem with a customer  who is using an old Windows 2000 server as firewall/router. It wont let packets through when using rport. It would work if rport was turned off and all communication was done using the configured port.
>> 
>> On Wed, Apr 14, 2010 at 3:53 PM, Brian West <brian at freeswitch.org> wrote:
>> You want rport otherwise NAT HELL will begin.
>> 
>> /b
>> 
>> On Apr 14, 2010, at 8:11 AM, Jonas Gauffin wrote:
>> 
>> > Well, yes. But snom can only turn off rport through provisioning and only in the latest firmware.
>> >
>> > It would be great if it could be turned off in FS too, per user account.
>> 
> 

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From mike at jerris.com  Thu Apr 15 06:41:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 09:41:22 -0400
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
Message-ID: <8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>

please post this and all other bugs to http://jira.freeswitch.org/ using the bug reporting guidelines found http://wiki.freeswitch.org/wiki/Reporting_Bugs

Mike

On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:

> I tried to analyze the core dump and got the following.
> 
> gdb ./freeswitch core.28775
> 

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From mike at jerris.com  Thu Apr 15 06:42:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 09:42:29 -0400
Subject: [Freeswitch-users] [INFO] User-Agent in mod_sofia,
	and git-builds.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557767736A@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557767736A@cooper>
Message-ID: <2047C82C-4B36-44D1-91DF-5682991F105E@jerris.com>

please assign a bug to me on http://jira.freeswitch.org and I will take care of this, thank you.

On Apr 15, 2010, at 4:40 AM, Peter Olsson wrote:

> I just wanted to share an experience I had with a git-build and Avaya Communications Manager PBX. It seems it can?t handle the new User-Agent values correctly, and this started to occur after doing git builds. In the end I  found that the ?:? in the timestamp was what broke the INVITE.
>  
> After overriding the user-agent in the sofia configuration it started to work again.
>  
> Probably it?s a Avaya problem ? but I just wanted to inform the other people on the list about this.
>  
> This is how my User-Agent looked like: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700
>  
> Regards,
>  
> Peter Olsson

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From mike at jerris.com  Thu Apr 15 06:44:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 09:44:18 -0400
Subject: [Freeswitch-users] DTMF issues
In-Reply-To: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
References: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
Message-ID: <A1DB602D-A6EE-424E-82C0-3FF202CBCC42@jerris.com>

Look at the sdp, are they responding with a different pt for telephone-event than in our initial invite?

Mike

On Apr 15, 2010, at 8:04 AM, Dan Lane wrote:

> I've got an issue where DTMF has stopped working with outbound calls
> (DTMF works on inbound calls from them) via one of my service
> providers (still works fine with several other providers) so I assumed
> it was a change the termination provider had made and duly reported it
> to them.
> 
> Their response was:
> 
> "We have analysed rtp from xlite to freeswitch to us and xlite to us
> 1. volume is being made much quieter by freeswitch, i.e. standard dtmf
> volume 10 is being re-written to 7
> 2. dtmf is shorter, i.e messages from phones are being re-written to a
> really short version
> 3. timestamp discontinuity  Audio to RTP conversion point. timestamps
> are indicating missing packets received via freeswitch"
> 
> The service provider is running a mixture of Cisco AS 5300's and
> 5350xm's, I don't think it's a bug in Freeswitch (I have a number of
> Freeswitch boxes that are all affected running different versions from
> 1.0.4pre7 to r17188) as DTMF was working once upon a time... can
> anyone think of anything obvious that either I or the service provider
> could be missing?
> 
> Thanks.



From mike at jerris.com  Thu Apr 15 06:46:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 09:46:17 -0400
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <SNT133-w5393F1951FBF79FB8E145EA00F0@phx.gbl>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>,
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>,
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>,
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>,
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>,
	<i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>,
	<9E1A4D19-A8C4-41A3-AD2B-E324EA31B43B@jerris.com>
	<SNT133-w5393F1951FBF79FB8E145EA00F0@phx.gbl>
Message-ID: <FBD6F65A-E5E9-4230-A9B7-0347BB235859@jerris.com>

The threshold for rtp timeout is supposed to be consecutive packets lost, not over the duration of the call.  Please double check that there is not some long run of silence or something causing an actual rtp timeout, and if you can confirm that, please open a bug on jira for this issue.

Mike

On Apr 15, 2010, at 8:08 AM, Francisco Scaramanga wrote:

> Now I am using stable release 1.0.6 and have a MEDIA_TIMEOUT again, because function rtp_common_read counts missed packets.
> During a long session of several hours the amount of missed packets exceeds the limit. 
> 
> Now I changed my .\conf\sip_profiles\internal.xml and set the rtp timeout to 0, which means no timeout.
>   
>   set <param name="rtp-timeout-sec" value="0"/>
> 
> My usecase is to have long duration calls. Can I do this without cross-effects, or is not wise to set rtp timout to 0? 
> 
> > From: mike at jerris.com
> > Date: Wed, 14 Apr 2010 09:08:32 -0400
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
> > 
> > That being said, for a change things have been somewhat quiet after the 1.0.6 release. There have been some bug fixes that went in after, but not the usual frenzy of reports right after a release.
> > 
> > Mike
> > 
> > On Apr 14, 2010, at 7:23 AM, Steven Ayre wrote:
> > 
> > > We are no longer on SVN, we have now moved to GIT for the 1.0.6 release.
> > > 
> > > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH
> > > 
> > > Note that the latest most stable version is always likely to be latest
> > > GIT version not the latest tagged released, as there will often be
> > > bugs present in the releases that are fixed in the latest GIT version.
> > > Tagged releases have more to do with the introduction of new features
> > > (such as G729 support in 1.0.6) than any guarantee of stability.
> > > 
> > > -Steve
> > > 
> > > 
> > > On 14 April 2010 09:00, Francisco Scaramanga <scaram at hotmail.de> wrote:
> > >> 
> > >> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
> > >> I thought it is a release candidate, isn't it? Is it safe to use it in
> > >> production?
> > >> 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> ?ber Messenger g?nstiger telefonieren? Sagen Sie "Ja" zu JAJAH! _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fanta at me.com  Thu Apr 15 06:44:05 2010
From: fanta at me.com (Christian van der Leeden)
Date: Thu, 15 Apr 2010 15:44:05 +0200
Subject: [Freeswitch-users] Music On Hold for SNOM phones with freeswitch
Message-ID: <E8A07429-A8DA-4017-A417-46CE9B637720@me.com>

Hi,

	I'd like to configure freeswitch so it acts as music on hold server (replying to a
SIP Invite with music) for a SNOM phone. The SNOM phone is registered on my
Patton SmartNode. I've tried to point it to 9999 at 192.168.140.7 (ip of my server)
but freeswitch responds with 
just installed freeswitch.org on my linux box. I'd like to use it play music
on hold to a SNOM VoIP phone (it is registered on my patton SmartNode, not
with the freeswitch server).

I receive a 407 Proxy authentication required.

This is on a out of the box "quick install" from the git trunk.

What would I have to do, so freeswitch would answer to the invite?

Christian
P.S.: attached the session log

INVITE sip:9999 at 192.168.140.7;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:central at 192.168.140.42:1024>;reg-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.3.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 452

v=0
o=root 30256492 30256492 IN IP4 192.168.140.42
s=call
c=IN IP4 192.168.140.42
t=0 0
m=audio 54834 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:TpHIcA93pqErUArxCuWRC1tcd7mp8LTs73clcwnE
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:647 (362 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700
Content-Length: 0


Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:802 (853 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024
From: "Zentrale" <sip:central at patton.logicunited.com>;tag=d3h27cphyo
To: <sip:9999 at 192.168.140.7;user=phone>;tag=4jg7ajj84BB7K
Call-ID: 3c28f4234780-u9oqvzvds56e
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="patton.logicunited.com", nonce="82f5fa24-ae3a-4def-a67f-b097f599bc34", algorithm=MD5, qop="auth"
Content-Length: 0



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From brian at freeswitch.org  Thu Apr 15 06:50:19 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 15 Apr 2010 08:50:19 -0500
Subject: [Freeswitch-users] Music On Hold for SNOM phones with freeswitch
In-Reply-To: <E8A07429-A8DA-4017-A417-46CE9B637720@me.com>
References: <E8A07429-A8DA-4017-A417-46CE9B637720@me.com>
Message-ID: <E43D84EB-3015-4D23-A3AD-3B7B68D384B4@freeswitch.org>

Chance are you'll need to turn off authentication on the profile in question.

/b

On Apr 15, 2010, at 8:44 AM, Christian van der Leeden wrote:

> I receive a 407 Proxy authentication required.
> 



From anthony.minessale at gmail.com  Thu Apr 15 07:51:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 09:51:33 -0500
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
Message-ID: <u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>

you did not specify the d-chan, it should probably not crash in this case
but that is the cause.


On Thu, Apr 15, 2010 at 8:41 AM, Michael Jerris <mike at jerris.com> wrote:

> please post this and all other bugs to http://jira.freeswitch.org/ using
> the bug reporting guidelines found
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> <http://wiki.freeswitch.org/wiki/Reporting_Bugs>Mike
>
> On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:
>
> I tried to analyze the core dump and got the following.
>
> gdb ./freeswitch core.28775
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Apr 15 08:37:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 10:37:01 -0500
Subject: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
In-Reply-To: <FBD6F65A-E5E9-4230-A9B7-0347BB235859@jerris.com>
References: <SNT133-w1423810B903B2713B72BF0A0110@phx.gbl>
	<u2heafa772f1004130823n3079dccem25aaafc9120cb335@mail.gmail.com>
	<SNT133-w30AAA5D55B1E8E3674B850A0110@phx.gbl>
	<x2y191c3a031004131019ua3069053nddb4932937bb04a8@mail.gmail.com>
	<SNT133-w55A8772E1AC9694505F90AA0100@phx.gbl>
	<i2neafa772f1004140423v46a44ffy6b11debc9d122dc5@mail.gmail.com>
	<9E1A4D19-A8C4-41A3-AD2B-E324EA31B43B@jerris.com>
	<SNT133-w5393F1951FBF79FB8E145EA00F0@phx.gbl>
	<FBD6F65A-E5E9-4230-A9B7-0347BB235859@jerris.com>
Message-ID: <h2r191c3a031004150837m8f0a045au85727b734f81e93c@mail.gmail.com>

turning off media timeout is perfectly ok, its not even valid to use media
timeouts, we have it as a courtesy.
sip session timers is the valid method.

On Thu, Apr 15, 2010 at 8:46 AM, Michael Jerris <mike at jerris.com> wrote:

> The threshold for rtp timeout is supposed to be consecutive packets lost,
> not over the duration of the call.  Please double check that there is not
> some long run of silence or something causing an actual rtp timeout, and if
> you can confirm that, please open a bug on jira for this issue.
>
> Mike
>
> On Apr 15, 2010, at 8:08 AM, Francisco Scaramanga wrote:
>
> Now I am using stable release 1.0.6 and have a MEDIA_TIMEOUT again, because
> function rtp_common_read counts missed packets.
> During a long session of several hours the amount of missed packets exceeds
> the limit.
>
> Now I changed my .\conf\sip_profiles\internal.xml and set the rtp timeout
> to 0, which means no timeout.
>
>   set <param name="rtp-timeout-sec" value="0"/>
>
> My usecase is to have long duration calls. Can I do this without
> cross-effects, or is not wise to set rtp timout to 0?
>
> > From: mike at jerris.com
> > Date: Wed, 14 Apr 2010 09:08:32 -0400
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] MEDIA_TIMEOUT after long duration call
> >
> > That being said, for a change things have been somewhat quiet after the
> 1.0.6 release. There have been some bug fixes that went in after, but not
> the usual frenzy of reports right after a release.
> >
> > Mike
> >
> > On Apr 14, 2010, at 7:23 AM, Steven Ayre wrote:
> >
> > > We are no longer on SVN, we have now moved to GIT for the 1.0.6
> release.
> > >
> > > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH
> > >
> > > Note that the latest most stable version is always likely to be latest
> > > GIT version not the latest tagged released, as there will often be
> > > bugs present in the releases that are fixed in the latest GIT version.
> > > Tagged releases have more to do with the introduction of new features
> > > (such as G729 support in 1.0.6) than any guarantee of stability.
> > >
> > > -Steve
> > >
> > >
> > > On 14 April 2010 09:00, Francisco Scaramanga <scaram at hotmail.de>
> wrote:
> > >>
> > >> Ok, I see http://svn.freeswitch.org/svn/freeswitch/tags/1.0.rc6
> > >> I thought it is a release candidate, isn't it? Is it safe to use it in
> > >> production?
> > >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> ?ber Messenger g?nstiger telefonieren? Sagen Sie "Ja" zu JAJAH!<http://redirect.gimas.net/?n=M1004jajah>
>  _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From stevendt at primrosebank.net  Thu Apr 15 09:32:43 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Thu, 15 Apr 2010 17:32:43 +0100
Subject: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
Message-ID: <4B281EC072E8405A9B2ACF2925901D65@bp1.ad.bp.com>

Hi,

I'm trying to use mod_skinny under windows but can't get the Visual Studio 2008 Express build configuration working. 

I know that Mathieu does not run Windows, but can anyone else modify the VS build solution in source to include building mod_skinny please ?

regards
Dave
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From peter.olsson at visionutveckling.se  Thu Apr 15 09:49:58 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 15 Apr 2010 18:49:58 +0200
Subject: [Freeswitch-users] [INFO] User-Agent in mod_sofia,
 and git-builds.
In-Reply-To: <2047C82C-4B36-44D1-91DF-5682991F105E@jerris.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557767736A@cooper>,
	<2047C82C-4B36-44D1-91DF-5682991F105E@jerris.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4C84@cooper>

Added to Jira: MODSOFIA-71

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Jerris [mike at jerris.com]
Skickat: den 15 april 2010 15:42
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] [INFO] User-Agent in mod_sofia,    and git-builds.

please assign a bug to me on http://jira.freeswitch.org and I will take care of this, thank you.

On Apr 15, 2010, at 4:40 AM, Peter Olsson wrote:

I just wanted to share an experience I had with a git-build and Avaya Communications Manager PBX. It seems it can?t handle the new User-Agent values correctly, and this started to occur after doing git builds. In the end I  found that the ?:? in the timestamp was what broke the INVITE.

After overriding the user-agent in the sofia configuration it started to work again.

Probably it?s a Avaya problem ? but I just wanted to inform the other people on the list about this.

This is how my User-Agent looked like: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700

Regards,

Peter Olsson

!DSPAM:4bc719f432932804212586!


From fanta at me.com  Thu Apr 15 13:36:48 2010
From: fanta at me.com (Christian van der Leeden)
Date: Thu, 15 Apr 2010 22:36:48 +0200
Subject: [Freeswitch-users] Music On Hold for SNOM phones with freeswitch
In-Reply-To: <E43D84EB-3015-4D23-A3AD-3B7B68D384B4@freeswitch.org>
References: <E8A07429-A8DA-4017-A417-46CE9B637720@me.com>
	<E43D84EB-3015-4D23-A3AD-3B7B68D384B4@freeswitch.org>
Message-ID: <310A2E99-C2BA-47C0-BE5D-D72CF31D3086@me.com>

Hi,

	thanks for the hint, I was addressing the internal profile, which needs authentication,
so I am now going to port 5080 (profile external, dialplan public) and added the 
XML for the hold music extension to the public.xml dialplan.

>From SIP Trace this works fine, have to try it out tomorrow when in the office to see
if the phone does play music on hold :-)

Best Regards

Christian

On 15.04.2010, at 15:50, Brian West wrote:

> Chance are you'll need to turn off authentication on the profile in question.
> 
> /b
> 
> On Apr 15, 2010, at 8:44 AM, Christian van der Leeden wrote:
> 
>> I receive a 407 Proxy authentication required.
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lon at kickasspixels.com  Thu Apr 15 13:56:55 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 15 Apr 2010 13:56:55 -0700
Subject: [Freeswitch-users] Load balancing question
Message-ID: <5952013E-61F4-4149-BA1F-CBC52B381473@kickasspixels.com>

Hi gang,

I am tasked with load balancing calls to a cluster of Freeswitch servers, all running different "applications". Some will be handling conferencing others regular call routing.

We are using OpenSIPs to proxy to the clusters, but I'm trying to figure out a way to "know" which server is handling a particular conference for example. So we route all participants into the correct server and conference room.

Any ideas on doing this?

Thanks for any ideas.

Lon

From anthony.minessale at gmail.com  Thu Apr 15 14:06:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 16:06:49 -0500
Subject: [Freeswitch-users] Load balancing question
In-Reply-To: <5952013E-61F4-4149-BA1F-CBC52B381473@kickasspixels.com>
References: <5952013E-61F4-4149-BA1F-CBC52B381473@kickasspixels.com>
Message-ID: <h2y191c3a031004151406w1673657ew5f13e1e0ee34f871@mail.gmail.com>

you could write a plugin for opensips that uses the esl library so it can
get events from all the FS boxes and know what is going on in each box
perhaps.

Or you could use the special variables from your dialplan to send X-headers
back to opensips in the 200ok indicating info to track.


On Thu, Apr 15, 2010 at 3:56 PM, Lon Baker <lon at kickasspixels.com> wrote:

> Hi gang,
>
> I am tasked with load balancing calls to a cluster of Freeswitch servers,
> all running different "applications". Some will be handling conferencing
> others regular call routing.
>
> We are using OpenSIPs to proxy to the clusters, but I'm trying to figure
> out a way to "know" which server is handling a particular conference for
> example. So we route all participants into the correct server and conference
> room.
>
> Any ideas on doing this?
>
> Thanks for any ideas.
>
> Lon
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From null at invalid.name  Thu Apr 15 14:12:39 2010
From: null at invalid.name (Dan Lane)
Date: Thu, 15 Apr 2010 22:12:39 +0100
Subject: [Freeswitch-users] DTMF issues
In-Reply-To: <A1DB602D-A6EE-424E-82C0-3FF202CBCC42@jerris.com>
References: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
	<A1DB602D-A6EE-424E-82C0-3FF202CBCC42@jerris.com>
Message-ID: <n2gb042b23b1004151412k99cbb0d7id5c0ea872fc620bd@mail.gmail.com>

On Thu, Apr 15, 2010 at 2:44 PM, Michael Jerris <mike at jerris.com> wrote:
> Look at the sdp, are they responding with a different pt for telephone-event than in our initial invite?

Thanks for the reply Mike,

I know that AQL have that issue but with this other provider both the
invite and the 183 progress have the same values and nothing looks out
of the ordinary. What's weird is that the provider claims that
something in FS is rewriting the DTMF length even though I've tried
manually changing it from 100ms to 300ms (to match the Cisco's
default).

Unfortunately Yehavi's suggestion isn't achievable either as the
provider doesn't support sip-info :(

Everything to do with RTP/DTMF is default and I've tried all the usual
tricks mention here: http://wiki.freeswitch.org/wiki/RTP_Issues.

Anyone else having these issues with Cisco kit?


From lon at kickasspixels.com  Thu Apr 15 14:18:14 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 15 Apr 2010 14:18:14 -0700
Subject: [Freeswitch-users] Load balancing question
In-Reply-To: <h2y191c3a031004151406w1673657ew5f13e1e0ee34f871@mail.gmail.com>
References: <5952013E-61F4-4149-BA1F-CBC52B381473@kickasspixels.com>
	<h2y191c3a031004151406w1673657ew5f13e1e0ee34f871@mail.gmail.com>
Message-ID: <E034B144-C35E-4D75-89B4-8BDAC327307C@kickasspixels.com>

Thank you! I will look into those options.

I had already scoped out using the ESL to track answered calls and hangups to update a MySQL table  and have OpenSIPs use that to route calls.

--
Lon Baker

On Apr 15, 2010, at 2:06 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> you could write a plugin for opensips that uses the esl library so it can get events from all the FS boxes and know what is going on in each box perhaps.
> 
> Or you could use the special variables from your dialplan to send X-headers back to opensips in the 200ok indicating info to track.
> 
> 
> On Thu, Apr 15, 2010 at 3:56 PM, Lon Baker <lon at kickasspixels.com> wrote:
> Hi gang,
> 
> I am tasked with load balancing calls to a cluster of Freeswitch servers, all running different "applications". Some will be handling conferencing others regular call routing.
> 
> We are using OpenSIPs to proxy to the clusters, but I'm trying to figure out a way to "know" which server is handling a particular conference for example. So we route all participants into the correct server and conference room.
> 
> Any ideas on doing this?
> 
> Thanks for any ideas.
> 
> Lon
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 15 14:59:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 17:59:52 -0400
Subject: [Freeswitch-users] DTMF issues
In-Reply-To: <n2gb042b23b1004151412k99cbb0d7id5c0ea872fc620bd@mail.gmail.com>
References: <u2kb042b23b1004150504i5eb540fax215fb497d3b458b8@mail.gmail.com>
	<A1DB602D-A6EE-424E-82C0-3FF202CBCC42@jerris.com>
	<n2gb042b23b1004151412k99cbb0d7id5c0ea872fc620bd@mail.gmail.com>
Message-ID: <58672AE8-02C5-4BDE-B995-49A79B61B26B@jerris.com>

if you can hit me up on IM I can help you out sorting this one.

Mike

On Apr 15, 2010, at 5:12 PM, Dan Lane wrote:

> On Thu, Apr 15, 2010 at 2:44 PM, Michael Jerris <mike at jerris.com> wrote:
>> Look at the sdp, are they responding with a different pt for telephone-event than in our initial invite?
> 
> Thanks for the reply Mike,
> 
> I know that AQL have that issue but with this other provider both the
> invite and the 183 progress have the same values and nothing looks out
> of the ordinary. What's weird is that the provider claims that
> something in FS is rewriting the DTMF length even though I've tried
> manually changing it from 100ms to 300ms (to match the Cisco's
> default).
> 
> Unfortunately Yehavi's suggestion isn't achievable either as the
> provider doesn't support sip-info :(
> 
> Everything to do with RTP/DTMF is default and I've tried all the usual
> tricks mention here: http://wiki.freeswitch.org/wiki/RTP_Issues.
> 
> Anyone else having these issues with Cisco kit?
> 


From garrison at codefix.net  Thu Apr 15 21:51:30 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Fri, 16 Apr 2010 00:51:30 -0400
Subject: [Freeswitch-users] Dial Group & Handy-Tone 386
In-Reply-To: <44FE0C3D-B508-4A55-8769-4EF66B6C41C7@jerris.com>
References: <4BC34D7B.7000606@codefix.net> <4BC5553C.9090109@codefix.net>
	<44FE0C3D-B508-4A55-8769-4EF66B6C41C7@jerris.com>
Message-ID: <4BC7ECD2.9030506@codefix.net>

Michael Jerris wrote:
> If this "fixed" your issue, it sounds like some issue getting the
> real answer indication from the answering phone.  I would enable sip
> trace and see if we are actually getting the 200 from the phone or
> not.

Actually it seems worse, but I've not had any time to look carefully. 
This weekend I'll read through all the debugging info on the wiki, looks 
like SIP trace is just the beginning of available diagnostics.

Thanks for the tips.

-gh



From bruce at nani.ca  Fri Apr 16 00:11:37 2010
From: bruce at nani.ca (Bruce Fletcher)
Date: Fri, 16 Apr 2010 00:11:37 -0700
Subject: [Freeswitch-users] mod_pocketsphinx link error on Mac OS X 10.6
Message-ID: <18B39E9E-4212-45D0-9F24-34DCEF3CAF45@nani.ca>

I'm having trouble with mod_pocketsphinx on Mac OS X 10.6 using git head from Wed Apr 14 21:36:41 2010 -0400.

My freeswitch.log says:

2010-04-15 22:50:30.418812 [NOTICE] switch_loadable_module.c:272 Adding API Function 'luarun'
2010-04-15 22:50:30.418863 [NOTICE] switch_loadable_module.c:272 Adding API Function 'lua'
2010-04-15 22:50:30.419302 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so
**dlopen(/usr/local/freeswitch/mod/mod_pocketsphinx.so, 6): Symbol not found: _iconv
 Referenced from: /usr/local/freeswitch/mod/mod_pocketsphinx.so
 Expected in: flat namespace
in /usr/local/freeswitch/mod/mod_pocketsphinx.so**

From my freeswitch.la I see:

# Libraries that this one depends upon.
dependency_libs=' -L/Users/admin/src/freeswitch/libs/apr-util/xml/expat/lib /Users/admin/src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la /Users/admin/src/freeswitch/libs/apr/libapr-1.la -liconv -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -lncurses'

That looks OK to my unskilled eyes, although I do have libiconv in two places, the default /usr/lib:

lrwxr-xr-x  1 root  wheel       16 Sep 10  2009 /usr/lib/libiconv.2.4.0.dylib -> libiconv.2.dylib
-r-xr-xr-x  1 root  wheel  3205760 Nov 19 18:00 /usr/lib/libiconv.2.dylib
lrwxr-xr-x  1 root  wheel       20 Sep 10  2009 /usr/lib/libiconv.dylib -> libiconv.2.4.0.dylib

and a copy from MacPorts, which came for the ride when I installed git:

-rw-r--r--  2 root  admin  1084720 Apr  2 12:40 /opt/local/lib/libiconv.2.dylib
-rw-r--r--  2 root  admin  1115488 Apr  2 12:40 /opt/local/lib/libiconv.a
lrw-r--r--  1 root  admin       16 Apr  2 12:40 /opt/local/lib/libiconv.dylib -> libiconv.2.dylib
-rw-r--r--  2 root  admin      918 Apr  2 12:40 /opt/local/lib/libiconv.la

Does anyone have any thoughts on how I can debug this?

Thanks,
- Bruce



From lakindia89 at gmail.com  Fri Apr 16 02:46:52 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 16 Apr 2010 15:16:52 +0530
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
	<u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
Message-ID: <t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>

Dear Anthony,
I have also tried it by specifying the D-Channel as follows:

[span wanpipe PRI_1]

name => OpenZAP

number => 1

trunk_type => e1

b-channel => 1:1-15

d-channel => 1:16

b-channel => 1:17-31

[span wanpipe PRI_2]

name => OpenZAP

number => 2

trunk_type => e1

b-channel => 2:1-15

d-channel => 2:16

b-channel => 2:17-31
In this case also I'm gtting the segmentation fult with core dumped.
But for both the D-Channels I got the following error message:



2010-04-16 13:57:26.317548 [INFO] ozmod_wanpipe.c:337 configuring device
s1c15 as OpenZAP device 1:15 fd:54 DTMF: software

2010-04-16 13:57:26.318046 [ERR] ozmod_wanpipe.c:235 Failed to open wanpipe
device span 1 channel 16

TDM API: CMD: 36

2010-04-16 13:57:26.342996 [INFO] ozmod_wanpipe.c:337 configuring device
s2c15 as OpenZAP device 2:15 fd:84 DTMF: software

2010-04-16 13:57:26.343460 [ERR] ozmod_wanpipe.c:235 Failed to open wanpipe
device span 2 channel 16

TDM API: CMD: 36

2010-04-16 13:57:26.355616 [INFO] zap_io.c:2777 Configured 60 channel(s)

2010-04-16 13:57:26.357252 [INFO] zap_io.c:2853 Loading IO from
/usr/local/freeswitch/mod/ozmod_libpri.so [libpri]

2010-04-16 13:57:26.357721 [INFO] zap_io.c:2870 Loading SIG from
/usr/local/freeswitch/mod/ozmod_libpri.so

2010-04-16 13:57:26.357938 [INFO] zap_io.c:2986 auto-loaded 'libpri'

2010-04-16 13:57:26.358153 [NOTICE] ozmod_libpri.c:1267 Setting default
Layer 1 to ALAW since this is an E1 trunk

Segmentation fault (core dumped)
Did I miss anything!!!.
I cheched the status of the wnrouter. It shows the channel is connected.
Do you have any guess when it fails to open D channel??


On 4/15/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> you did not specify the d-chan, it should probably not crash in this case
> but that is the cause.
>
>
>  On Thu, Apr 15, 2010 at 8:41 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>  please post this and all other bugs to http://jira.freeswitch.org/ using
>> the bug reporting guidelines found
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>>
>>  <http://wiki.freeswitch.org/wiki/Reporting_Bugs>Mike
>>
>>  On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:
>>
>> I tried to analyze the core dump and got the following.
>>
>> gdb ./freeswitch core.28775
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Sat Apr 17 04:55:58 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 17 Apr 2010 13:55:58 +0200
Subject: [Freeswitch-users] GIT Clone error
Message-ID: <4BC9A1CE.4080603@gmail.com>

hi all
i have error while using git in windows:

git.exe clone --progress -v  "git://git.freeswitch.org/freeswitch.git" 
"C:\SDK\Projects\freeswitch\freeswitch"

Initialized empty Git repository in 
C:/SDK/Projects/freeswitch/freeswitch/.git/
Server supports multi_ack_detailed
Server supports side-band-64k
Server supports ofs-delta
want 67504b97ef1b344926493167dc507e3dfed8951c (refs/heads/gitbuild)
want 00b262235007cb6407c986b355d3a49881401419 (refs/heads/master)
want 417171a8f352e398c700e1669e10590994614494 (refs/heads/rupa)
want 43bdf5747ec26b98e392af7e438140f686be1769 
(refs/tags/git2svn-syncpoint-master)
want 6ccecbeba028e684f001f823a20108d471ba7141 (refs/tags/v1.0-beta1)
want 6b5eefd3427ecc7dab5b48b1919837809c6cf1d5 (refs/tags/v1.0-beta2)
want 735423328eed3ac58d60b1c89bff593f393fc3d8 (refs/tags/v1.0-rc1)
want 93efeb8cef9b336e99265794f8e113199b97743e (refs/tags/v1.0-rc2)
want 58822f65cd506d4aef93081ab4d4f8ba12ccec3a (refs/tags/v1.0-rc3)
want ef58f0ecbb3bdca4b03dcacbb768f32512fd6006 (refs/tags/v1.0-rc4)
want cec85f1f54dc78b29cdfb89955688d369a4d8d8f (refs/tags/v1.0-rc5)
want 80ee68912b378f3d32c34a13ce688d97d8b46ddb (refs/tags/v1.0-rc6)
want 261c08751e93161f3d578e9994d3bc2c67e128d4 (refs/tags/v1.0.0)
want 8bb9531e16b2844546ed6874754c4e6e7ebd01ec (refs/tags/v1.0.1)
want afd011177d9b8773996328f7ea4c19c4848e27e8 (refs/tags/v1.0.2)
want 537cfcf5abb40febd67970d8d1b5cf10d4ad9ee9 (refs/tags/v1.0.3)
want ca55f6175805b6b247e2bcd6fd5d7a6c4acd6d1c (refs/tags/v1.0.4)
want e29ae074a94e7ae4fecc53366d5a71fe65f9c248 (refs/tags/v1.0.5.14226d2)
want c1dc8fb1468e58b6b6711ed912ec7af92a6edbc1 (refs/tags/v1.0.6)
done
fatal: The remote end hung up unexpectedly
fatal: early EOF

any help is welcome


From mike at jerris.com  Fri Apr 16 06:32:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Apr 2010 09:32:18 -0400
Subject: [Freeswitch-users] GIT Clone error
In-Reply-To: <4BC9A1CE.4080603@gmail.com>
References: <4BC9A1CE.4080603@gmail.com>
Message-ID: <E78D3A1E-0A5A-4AE7-AAAD-6417447C8E1D@jerris.com>

At a glance in looks like a connectivity problem, does it get part way and die each time if you try to fetch from that point or does it always die there?

Mike

On Apr 17, 2010, at 7:55 AM, Meftah Tayeb wrote:

> hi all
> i have error while using git in windows:
> 
> git.exe clone --progress -v  "git://git.freeswitch.org/freeswitch.git" 
> "C:\SDK\Projects\freeswitch\freeswitch"
> 
> Initialized empty Git repository in 
> C:/SDK/Projects/freeswitch/freeswitch/.git/
> Server supports multi_ack_detailed
> Server supports side-band-64k
> Server supports ofs-delta
> want 67504b97ef1b344926493167dc507e3dfed8951c (refs/heads/gitbuild)
> want 00b262235007cb6407c986b355d3a49881401419 (refs/heads/master)
> want 417171a8f352e398c700e1669e10590994614494 (refs/heads/rupa)
> want 43bdf5747ec26b98e392af7e438140f686be1769 
> (refs/tags/git2svn-syncpoint-master)
> want 6ccecbeba028e684f001f823a20108d471ba7141 (refs/tags/v1.0-beta1)
> want 6b5eefd3427ecc7dab5b48b1919837809c6cf1d5 (refs/tags/v1.0-beta2)
> want 735423328eed3ac58d60b1c89bff593f393fc3d8 (refs/tags/v1.0-rc1)
> want 93efeb8cef9b336e99265794f8e113199b97743e (refs/tags/v1.0-rc2)
> want 58822f65cd506d4aef93081ab4d4f8ba12ccec3a (refs/tags/v1.0-rc3)
> want ef58f0ecbb3bdca4b03dcacbb768f32512fd6006 (refs/tags/v1.0-rc4)
> want cec85f1f54dc78b29cdfb89955688d369a4d8d8f (refs/tags/v1.0-rc5)
> want 80ee68912b378f3d32c34a13ce688d97d8b46ddb (refs/tags/v1.0-rc6)
> want 261c08751e93161f3d578e9994d3bc2c67e128d4 (refs/tags/v1.0.0)
> want 8bb9531e16b2844546ed6874754c4e6e7ebd01ec (refs/tags/v1.0.1)
> want afd011177d9b8773996328f7ea4c19c4848e27e8 (refs/tags/v1.0.2)
> want 537cfcf5abb40febd67970d8d1b5cf10d4ad9ee9 (refs/tags/v1.0.3)
> want ca55f6175805b6b247e2bcd6fd5d7a6c4acd6d1c (refs/tags/v1.0.4)
> want e29ae074a94e7ae4fecc53366d5a71fe65f9c248 (refs/tags/v1.0.5.14226d2)
> want c1dc8fb1468e58b6b6711ed912ec7af92a6edbc1 (refs/tags/v1.0.6)
> done
> fatal: The remote end hung up unexpectedly
> fatal: early EOF
> 
> any help is welcome
> 


From anthony.minessale at gmail.com  Fri Apr 16 06:57:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Apr 2010 08:57:26 -0500
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
	<u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
	<t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>
Message-ID: <i2v191c3a031004160657o2926bfc2pd382b1396ad7d544@mail.gmail.com>

I think you must be.
Was the bt the same this time?
Maybe if you hop on irc we can have someone look at it for you.


On Fri, Apr 16, 2010 at 4:46 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear Anthony,
> I have also tried it by specifying the D-Channel as follows:
>
> [span wanpipe PRI_1]
>
> name => OpenZAP
>
> number => 1
>
> trunk_type => e1
>
> b-channel => 1:1-15
>
> d-channel => 1:16
>
> b-channel => 1:17-31
>
> [span wanpipe PRI_2]
>
> name => OpenZAP
>
> number => 2
>
> trunk_type => e1
>
> b-channel => 2:1-15
>
> d-channel => 2:16
>
> b-channel => 2:17-31
> In this case also I'm gtting the segmentation fult with core dumped.
> But for both the D-Channels I got the following error message:
>
>
>
> 2010-04-16 13:57:26.317548 [INFO] ozmod_wanpipe.c:337 configuring device
> s1c15 as OpenZAP device 1:15 fd:54 DTMF: software
>
> 2010-04-16 13:57:26.318046 [ERR] ozmod_wanpipe.c:235 Failed to open wanpipe
> device span 1 channel 16
>
> TDM API: CMD: 36
>
> 2010-04-16 13:57:26.342996 [INFO] ozmod_wanpipe.c:337 configuring device
> s2c15 as OpenZAP device 2:15 fd:84 DTMF: software
>
> 2010-04-16 13:57:26.343460 [ERR] ozmod_wanpipe.c:235 Failed to open wanpipe
> device span 2 channel 16
>
> TDM API: CMD: 36
>
> 2010-04-16 13:57:26.355616 [INFO] zap_io.c:2777 Configured 60 channel(s)
>
> 2010-04-16 13:57:26.357252 [INFO] zap_io.c:2853 Loading IO from
> /usr/local/freeswitch/mod/ozmod_libpri.so [libpri]
>
> 2010-04-16 13:57:26.357721 [INFO] zap_io.c:2870 Loading SIG from
> /usr/local/freeswitch/mod/ozmod_libpri.so
>
> 2010-04-16 13:57:26.357938 [INFO] zap_io.c:2986 auto-loaded 'libpri'
>
> 2010-04-16 13:57:26.358153 [NOTICE] ozmod_libpri.c:1267 Setting default
> Layer 1 to ALAW since this is an E1 trunk
>
> Segmentation fault (core dumped)
> Did I miss anything!!!.
> I cheched the status of the wnrouter. It shows the channel is connected.
> Do you have any guess when it fails to open D channel??
>
>
> On 4/15/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>
>> you did not specify the d-chan, it should probably not crash in this case
>> but that is the cause.
>>
>>
>>  On Thu, Apr 15, 2010 at 8:41 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>>  please post this and all other bugs to http://jira.freeswitch.org/using the bug reporting guidelines found
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>>
>>>  <http://wiki.freeswitch.org/wiki/Reporting_Bugs>Mike
>>>
>>>  On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:
>>>
>>> I tried to analyze the core dump and got the following.
>>>
>>> gdb ./freeswitch core.28775
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From moises.silva at gmail.com  Fri Apr 16 07:15:26 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 16 Apr 2010 10:15:26 -0400
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <i2v191c3a031004160657o2926bfc2pd382b1396ad7d544@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
	<u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
	<t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>
	<i2v191c3a031004160657o2926bfc2pd382b1396ad7d544@mail.gmail.com>
Message-ID: <n2lc4d05cbe1004160715nfa67a4d4uae99e0edbcb07fb6@mail.gmail.com>

May be you have still running sangoma_prid ( which opens the d-channel ) and
then FreeSWITCH fails to open it (just a guess).

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Fri, Apr 16, 2010 at 9:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I think you must be.
> Was the bt the same this time?
> Maybe if you hop on irc we can have someone look at it for you.
>
>
> On Fri, Apr 16, 2010 at 4:46 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear Anthony,
>> I have also tried it by specifying the D-Channel as follows:
>>
>> [span wanpipe PRI_1]
>>
>> name => OpenZAP
>>
>> number => 1
>>
>> trunk_type => e1
>>
>> b-channel => 1:1-15
>>
>> d-channel => 1:16
>>
>> b-channel => 1:17-31
>>
>> [span wanpipe PRI_2]
>>
>> name => OpenZAP
>>
>> number => 2
>>
>> trunk_type => e1
>>
>> b-channel => 2:1-15
>>
>> d-channel => 2:16
>>
>> b-channel => 2:17-31
>> In this case also I'm gtting the segmentation fult with core dumped.
>> But for both the D-Channels I got the following error message:
>>
>>
>>
>> 2010-04-16 13:57:26.317548 [INFO] ozmod_wanpipe.c:337 configuring device
>> s1c15 as OpenZAP device 1:15 fd:54 DTMF: software
>>
>> 2010-04-16 13:57:26.318046 [ERR] ozmod_wanpipe.c:235 Failed to open
>> wanpipe device span 1 channel 16
>>
>> TDM API: CMD: 36
>>
>> 2010-04-16 13:57:26.342996 [INFO] ozmod_wanpipe.c:337 configuring device
>> s2c15 as OpenZAP device 2:15 fd:84 DTMF: software
>>
>> 2010-04-16 13:57:26.343460 [ERR] ozmod_wanpipe.c:235 Failed to open
>> wanpipe device span 2 channel 16
>>
>> TDM API: CMD: 36
>>
>> 2010-04-16 13:57:26.355616 [INFO] zap_io.c:2777 Configured 60 channel(s)
>>
>> 2010-04-16 13:57:26.357252 [INFO] zap_io.c:2853 Loading IO from
>> /usr/local/freeswitch/mod/ozmod_libpri.so [libpri]
>>
>> 2010-04-16 13:57:26.357721 [INFO] zap_io.c:2870 Loading SIG from
>> /usr/local/freeswitch/mod/ozmod_libpri.so
>>
>> 2010-04-16 13:57:26.357938 [INFO] zap_io.c:2986 auto-loaded 'libpri'
>>
>> 2010-04-16 13:57:26.358153 [NOTICE] ozmod_libpri.c:1267 Setting default
>> Layer 1 to ALAW since this is an E1 trunk
>>
>> Segmentation fault (core dumped)
>> Did I miss anything!!!.
>> I cheched the status of the wnrouter. It shows the channel is connected.
>> Do you have any guess when it fails to open D channel??
>>
>>
>> On 4/15/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>
>>> you did not specify the d-chan, it should probably not crash in this case
>>> but that is the cause.
>>>
>>>
>>>  On Thu, Apr 15, 2010 at 8:41 AM, Michael Jerris <mike at jerris.com>wrote:
>>>
>>>>  please post this and all other bugs to http://jira.freeswitch.org/using the bug reporting guidelines found
>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>
>>>>
>>>>  <http://wiki.freeswitch.org/wiki/Reporting_Bugs>Mike
>>>>
>>>>  On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:
>>>>
>>>> I tried to analyze the core dump and got the following.
>>>>
>>>> gdb ./freeswitch core.28775
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Sat Apr 17 14:16:18 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 17 Apr 2010 23:16:18 +0200
Subject: [Freeswitch-users] GIT Clone error
In-Reply-To: <E78D3A1E-0A5A-4AE7-AAAD-6417447C8E1D@jerris.com>
References: <4BC9A1CE.4080603@gmail.com>
	<E78D3A1E-0A5A-4AE7-AAAD-6417447C8E1D@jerris.com>
Message-ID: <4BCA2522.4020605@gmail.com>

hi mike,
agis doing it all time from today
uin there is the trace:

git.exe clone --progress -v  "git://git.freeswitch.org/freeswitch.git" 
"C:\SDK\Projects\freeswitch\freeswitch"

Initialized empty Git repository in 
C:/SDK/Projects/freeswitch/freeswitch/.git/
Server supports multi_ack_detailed
Server supports side-band-64k
Server supports ofs-delta
want 67504b97ef1b344926493167dc507e3dfed8951c (refs/heads/gitbuild)
want 9a689a45fc19ca0c2107d333f2f3ac5e0495776f (refs/heads/master)
want 417171a8f352e398c700e1669e10590994614494 (refs/heads/rupa)
want 43bdf5747ec26b98e392af7e438140f686be1769 
(refs/tags/git2svn-syncpoint-master)
want 6ccecbeba028e684f001f823a20108d471ba7141 (refs/tags/v1.0-beta1)
want 6b5eefd3427ecc7dab5b48b1919837809c6cf1d5 (refs/tags/v1.0-beta2)
want 735423328eed3ac58d60b1c89bff593f393fc3d8 (refs/tags/v1.0-rc1)
want 93efeb8cef9b336e99265794f8e113199b97743e (refs/tags/v1.0-rc2)
want 58822f65cd506d4aef93081ab4d4f8ba12ccec3a (refs/tags/v1.0-rc3)
want ef58f0ecbb3bdca4b03dcacbb768f32512fd6006 (refs/tags/v1.0-rc4)
want cec85f1f54dc78b29cdfb89955688d369a4d8d8f (refs/tags/v1.0-rc5)
want 80ee68912b378f3d32c34a13ce688d97d8b46ddb (refs/tags/v1.0-rc6)
want 261c08751e93161f3d578e9994d3bc2c67e128d4 (refs/tags/v1.0.0)
want 8bb9531e16b2844546ed6874754c4e6e7ebd01ec (refs/tags/v1.0.1)
want afd011177d9b8773996328f7ea4c19c4848e27e8 (refs/tags/v1.0.2)
want 537cfcf5abb40febd67970d8d1b5cf10d4ad9ee9 (refs/tags/v1.0.3)
want ca55f6175805b6b247e2bcd6fd5d7a6c4acd6d1c (refs/tags/v1.0.4)
want e29ae074a94e7ae4fecc53366d5a71fe65f9c248 (refs/tags/v1.0.5.14226d2)
want c1dc8fb1468e58b6b6711ed912ec7af92a6edbc1 (refs/tags/v1.0.6)
done
fatal: The remote end hung up unexpectedly
fatal: early EOF
with msisgit
and tortwase GIT
thanks
Le 16/04/2010 15:32, Michael Jerris a ?crit :
> At a glance in looks like a connectivity problem, does it get part way and die each time if you try to fetch from that point or does it always die there?
>
> Mike
>
> On Apr 17, 2010, at 7:55 AM, Meftah Tayeb wrote:
>
>    
>> hi all
>> i have error while using git in windows:
>>
>> git.exe clone --progress -v  "git://git.freeswitch.org/freeswitch.git"
>> "C:\SDK\Projects\freeswitch\freeswitch"
>>
>> Initialized empty Git repository in
>> C:/SDK/Projects/freeswitch/freeswitch/.git/
>> Server supports multi_ack_detailed
>> Server supports side-band-64k
>> Server supports ofs-delta
>> want 67504b97ef1b344926493167dc507e3dfed8951c (refs/heads/gitbuild)
>> want 00b262235007cb6407c986b355d3a49881401419 (refs/heads/master)
>> want 417171a8f352e398c700e1669e10590994614494 (refs/heads/rupa)
>> want 43bdf5747ec26b98e392af7e438140f686be1769
>> (refs/tags/git2svn-syncpoint-master)
>> want 6ccecbeba028e684f001f823a20108d471ba7141 (refs/tags/v1.0-beta1)
>> want 6b5eefd3427ecc7dab5b48b1919837809c6cf1d5 (refs/tags/v1.0-beta2)
>> want 735423328eed3ac58d60b1c89bff593f393fc3d8 (refs/tags/v1.0-rc1)
>> want 93efeb8cef9b336e99265794f8e113199b97743e (refs/tags/v1.0-rc2)
>> want 58822f65cd506d4aef93081ab4d4f8ba12ccec3a (refs/tags/v1.0-rc3)
>> want ef58f0ecbb3bdca4b03dcacbb768f32512fd6006 (refs/tags/v1.0-rc4)
>> want cec85f1f54dc78b29cdfb89955688d369a4d8d8f (refs/tags/v1.0-rc5)
>> want 80ee68912b378f3d32c34a13ce688d97d8b46ddb (refs/tags/v1.0-rc6)
>> want 261c08751e93161f3d578e9994d3bc2c67e128d4 (refs/tags/v1.0.0)
>> want 8bb9531e16b2844546ed6874754c4e6e7ebd01ec (refs/tags/v1.0.1)
>> want afd011177d9b8773996328f7ea4c19c4848e27e8 (refs/tags/v1.0.2)
>> want 537cfcf5abb40febd67970d8d1b5cf10d4ad9ee9 (refs/tags/v1.0.3)
>> want ca55f6175805b6b247e2bcd6fd5d7a6c4acd6d1c (refs/tags/v1.0.4)
>> want e29ae074a94e7ae4fecc53366d5a71fe65f9c248 (refs/tags/v1.0.5.14226d2)
>> want c1dc8fb1468e58b6b6711ed912ec7af92a6edbc1 (refs/tags/v1.0.6)
>> done
>> fatal: The remote end hung up unexpectedly
>> fatal: early EOF
>>
>> any help is 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 d at d-man.org  Thu Apr 15 11:26:53 2010
From: d at d-man.org (Darren Schreiber)
Date: Thu, 15 Apr 2010 11:26:53 -0700
Subject: [Freeswitch-users] REMINDER: FreeSWITCH Users Group / Installfest
 in San Francisco next week!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F17818D5A78@EXVMBX020-3.exch020.serverdata.net>

Hi everyone,
    I posted about this a while ago but just wanted to remind those who may live in the Bay Area of an upcoming FreeSWITCH Users Group meetup and installfest.

    The location I've picked is easy to get to by car, train, Muni, walking or even on a Segway. It's Borders books, located right near the CalTrain station in San Francisco. We'll be meeting in the cafe. If you need help finding your way, just let me know, but I suspect Google Maps has got that covered for you.

    If you're interested, PLEASE BRING A LAPTOP WITH A LINUX VIRTUAL MACHINE RUNNING IF YOU WANT TO PARTICIPATE IN THE INSTALL-FEST.

    For more details, see the full listing: http://www.meetup.com/fsusers/calendar/13009468/

Please note that this isn't a FreeSWITCH sponsored event (or Bandwidth or anyone else), it's just a casual organized meetup to meet others and learn about FreeSWITCH. There is no fee, sales pitch or evil corporate agenda backing this.

When: Sunday, April 25, 2010 4:00 PM

Where: 
Borders Book Stores (in the Cafe)
200 King Street 
San Francisco, CA 94107
415-357-9931

Feel free to email me if you have questions at d at d-man.org

I look forward to seeing all the locals ;-)

Thanks much,
    Darren Schreiber


From fs-list at communicatefreely.net  Fri Apr 16 17:48:00 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 16 Apr 2010 20:48:00 -0400
Subject: [Freeswitch-users] performance comparison between centos and
 freebsd
In-Reply-To: <w2l191c3a031004120958h34cf45dfiafa9acf29f2e4f1f@mail.gmail.com>
References: <j2ra44688a1004100714id991bf65o1a4da77e48e7ceb3@mail.gmail.com>
	<w2l191c3a031004120958h34cf45dfiafa9acf29f2e4f1f@mail.gmail.com>
Message-ID: <4BC90540.7020505@communicatefreely.net>

I appreciate the "pains" taken.

We are a FreeBSD shop here, and it was no trouble at all to get FS up and running.  We are running
FreeBSD 8.0 on 64bit Intel and so far haven't had any problems at all.

-Tim

Anthony Minessale wrote:
> Either this will help:
> 
> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Threading_Library
> 
> or nothing will.
> 
> FreeBSD is always a pain to support because it's like a spoiled actor in
> a bad movie.
> It makes many demands on everyone, it should be called DivaOS.  Still we
> try to support it and if it does not
> work as well we really can't do much about it.
> 
> 
> 
> On Sat, Apr 10, 2010 at 9:14 AM, Woody Dickson <woodydickson at gmail.com
> <mailto:woodydickson at gmail.com>> wrote:
> 
>     Hi,
> 
>     I have tested with running freeswitch1.0.5 on freebsd and centos 5.4.
> 
>     The expectation is that freeswitch 1.0.5 acting as media proxy would
>     perform better in freebsd, but I found that freebsd can only sustain
>     half of the total concurrent calls as in centos 5.4 (120 vs 60).
> 
>     The test is run on both ATOM CPU and VIA c7 and the result is
>     relatively the same.
> 
>     Does anyone know why?  Is this some sort of setting issues in freebsd
>     kernel?  I have tried with pure freebsd and pfsense and the result is
>     the same.
> 
> 
> 
> 
>     Woody
> 
>     _______________________________________________
>     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/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From d at d-man.org  Thu Apr 15 11:26:53 2010
From: d at d-man.org (Darren Schreiber)
Date: Thu, 15 Apr 2010 11:26:53 -0700
Subject: [Freeswitch-users] REMINDER: FreeSWITCH Users Group / Installfest
 in San Francisco next week!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F17818D5A78@EXVMBX020-3.exch020.serverdata.net>

Hi everyone,
    I posted about this a while ago but just wanted to remind those who may live in the Bay Area of an upcoming FreeSWITCH Users Group meetup and installfest.

    The location I've picked is easy to get to by car, train, Muni, walking or even on a Segway. It's Borders books, located right near the CalTrain station in San Francisco. We'll be meeting in the cafe. If you need help finding your way, just let me know, but I suspect Google Maps has got that covered for you.

    If you're interested, PLEASE BRING A LAPTOP WITH A LINUX VIRTUAL MACHINE RUNNING IF YOU WANT TO PARTICIPATE IN THE INSTALL-FEST.

    For more details, see the full listing: http://www.meetup.com/fsusers/calendar/13009468/

Please note that this isn't a FreeSWITCH sponsored event (or Bandwidth or anyone else), it's just a casual organized meetup to meet others and learn about FreeSWITCH. There is no fee, sales pitch or evil corporate agenda backing this.

When: Sunday, April 25, 2010 4:00 PM

Where: 
Borders Book Stores (in the Cafe)
200 King Street 
San Francisco, CA 94107
415-357-9931

Feel free to email me if you have questions at d at d-man.org

I look forward to seeing all the locals ;-)

Thanks much,
    Darren Schreiber


From msc at freeswitch.org  Fri Apr 16 19:26:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Apr 2010 19:26:30 -0700
Subject: [Freeswitch-users] Wiki info: fs_ivrd
Message-ID: <u2j87f2f3b91004161926neaed410cza34a1315fa202921@mail.gmail.com>

Gang,

I just finished some awesome documentation on using fs_ivrd and the Perl
ESL::IVR module:

http://wiki.freeswitch.org/wiki/Ivrd

If anyone else is using fs_ivrd and has some examples then by all means
please add to this page. Also, feel free to check the page's accuracy.

Thanks!
-MC
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From lakindia89 at gmail.com  Fri Apr 16 22:10:21 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 17 Apr 2010 10:40:21 +0530
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <n2lc4d05cbe1004160715nfa67a4d4uae99e0edbcb07fb6@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
	<u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
	<t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>
	<i2v191c3a031004160657o2926bfc2pd382b1396ad7d544@mail.gmail.com>
	<n2lc4d05cbe1004160715nfa67a4d4uae99e0edbcb07fb6@mail.gmail.com>
Message-ID: <q2m7d79b3931004162210q10db9fa7m9c08b9837746081a@mail.gmail.com>

Moises,
You are right. sangoma_prid is running on my system.
I gave smg_ctrl stop.
Then I started freeswitch.
Now it didn't get core-dumped.
Thanks for your help, updated the wiki.

On Fri, Apr 16, 2010 at 7:45 PM, Moises Silva <moises.silva at gmail.com>wrote:

> May be you have still running sangoma_prid ( which opens the d-channel )
> and then FreeSWITCH fails to open it (just a guess).
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
>
> On Fri, Apr 16, 2010 at 9:57 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I think you must be.
>> Was the bt the same this time?
>> Maybe if you hop on irc we can have someone look at it for you.
>>
>>
>> On Fri, Apr 16, 2010 at 4:46 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Dear Anthony,
>>> I have also tried it by specifying the D-Channel as follows:
>>>
>>> [span wanpipe PRI_1]
>>>
>>> name => OpenZAP
>>>
>>> number => 1
>>>
>>> trunk_type => e1
>>>
>>> b-channel => 1:1-15
>>>
>>> d-channel => 1:16
>>>
>>> b-channel => 1:17-31
>>>
>>> [span wanpipe PRI_2]
>>>
>>> name => OpenZAP
>>>
>>> number => 2
>>>
>>> trunk_type => e1
>>>
>>> b-channel => 2:1-15
>>>
>>> d-channel => 2:16
>>>
>>> b-channel => 2:17-31
>>> In this case also I'm gtting the segmentation fult with core dumped.
>>> But for both the D-Channels I got the following error message:
>>>
>>>
>>>
>>> 2010-04-16 13:57:26.317548 [INFO] ozmod_wanpipe.c:337 configuring device
>>> s1c15 as OpenZAP device 1:15 fd:54 DTMF: software
>>>
>>> 2010-04-16 13:57:26.318046 [ERR] ozmod_wanpipe.c:235 Failed to open
>>> wanpipe device span 1 channel 16
>>>
>>> TDM API: CMD: 36
>>>
>>> 2010-04-16 13:57:26.342996 [INFO] ozmod_wanpipe.c:337 configuring device
>>> s2c15 as OpenZAP device 2:15 fd:84 DTMF: software
>>>
>>> 2010-04-16 13:57:26.343460 [ERR] ozmod_wanpipe.c:235 Failed to open
>>> wanpipe device span 2 channel 16
>>>
>>> TDM API: CMD: 36
>>>
>>> 2010-04-16 13:57:26.355616 [INFO] zap_io.c:2777 Configured 60 channel(s)
>>>
>>> 2010-04-16 13:57:26.357252 [INFO] zap_io.c:2853 Loading IO from
>>> /usr/local/freeswitch/mod/ozmod_libpri.so [libpri]
>>>
>>> 2010-04-16 13:57:26.357721 [INFO] zap_io.c:2870 Loading SIG from
>>> /usr/local/freeswitch/mod/ozmod_libpri.so
>>>
>>> 2010-04-16 13:57:26.357938 [INFO] zap_io.c:2986 auto-loaded 'libpri'
>>>
>>> 2010-04-16 13:57:26.358153 [NOTICE] ozmod_libpri.c:1267 Setting default
>>> Layer 1 to ALAW since this is an E1 trunk
>>>
>>> Segmentation fault (core dumped)
>>> Did I miss anything!!!.
>>> I cheched the status of the wnrouter. It shows the channel is connected.
>>> Do you have any guess when it fails to open D channel??
>>>
>>>
>>> On 4/15/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>>
>>>> you did not specify the d-chan, it should probably not crash in this
>>>> case but that is the cause.
>>>>
>>>>
>>>>  On Thu, Apr 15, 2010 at 8:41 AM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>>  please post this and all other bugs to http://jira.freeswitch.org/using the bug reporting guidelines found
>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>
>>>>>
>>>>>  <http://wiki.freeswitch.org/wiki/Reporting_Bugs>Mike
>>>>>
>>>>>  On Apr 15, 2010, at 7:23 AM, lakshmanan ganapathy wrote:
>>>>>
>>>>> I tried to analyze the core dump and got the following.
>>>>>
>>>>> gdb ./freeswitch core.28775
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Fri Apr 16 22:30:21 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 17 Apr 2010 10:00:21 +0430
Subject: [Freeswitch-users] How to find out a sip extension is busy?
Message-ID: <v2m1fdf92721004162230o492905fene3d427bcbf5032e0@mail.gmail.com>

Hi
Is there any way to check if a sip extension is busy or not? or find an idle
sip extension?
Thanks
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From yehavi.bourvine at gmail.com  Fri Apr 16 22:55:52 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 17 Apr 2010 08:55:52 +0300
Subject: [Freeswitch-users] How to find out a sip extension is busy?
In-Reply-To: <v2m1fdf92721004162230o492905fene3d427bcbf5032e0@mail.gmail.com>
References: <v2m1fdf92721004162230o492905fene3d427bcbf5032e0@mail.gmail.com>
Message-ID: <r2td361ca7b1004162255hf56c63aza6e2678065c5950e@mail.gmail.com>

You can query the sip_dialogs table from core DB. We use it to find whether
an extension is busy and the act upon user's prefference (waiting call, play
busy or send to voicemail).

If you need more help please contact me off the list.

                     __Yehavi:

 2010/4/17 babak yakhchali <babak.freeswitch at gmail.com>

> Hi
> Is there any way to check if a sip extension is busy or not? or find an
> idle sip extension?
> 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 lakindia89 at gmail.com  Sat Apr 17 02:12:16 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 17 Apr 2010 14:42:16 +0530
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <n2t87f2f3b91004082241m1262877fsb2b596671c3d5f2@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
	<4BBB9DEB.2040702@gmail.com>
	<v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
	<4BBCBD73.7020206@gmail.com>
	<x2n7d79b3931004082138y6573df06hcfd093103a3f944b@mail.gmail.com>
	<n2t87f2f3b91004082241m1262877fsb2b596671c3d5f2@mail.gmail.com>
Message-ID: <y2n7d79b3931004170212g4e302cadz9b952003db556c78@mail.gmail.com>

Hi all,
         I've contacted the sangoma support, and they are looking into the
issue.
They also told that the problem is in the PRI STACK.
So in the mean time, I planned to change the stack and check. So I switched
to libpri.
I've configured the things and both Incoming and outgoing works on span1.
On span2, from the extension, If I dial, I got

2010-04-17 10:51:15.218100 [WARNING] ozmod_libpri.c:760 --Duplicate Ring on
channel 2:-1 (ignored)

The freeswitch log is:
     http://pastebin.freeswitch.org/12738
I also went to see what is the problem. It seems to me that, when SETUP
comes, ozmod_libpri.c is checking for the B-channel number.
Since it is not there, it print Duplicate ring on channel.

I also came to know that, first SETUP will request the telco (B-chan
identification won't be there). Then the telco responds with the B-channel
to use in SETUP_ACK.
The further communication will happen with the B-chan identification.

Am  I correct!!. Please give inputs to me to solve this issue.


On Fri, Apr 9, 2010 at 11:11 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Apr 8, 2010 at 9:38 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Right now I can't provide you the ssh access to the box, since the company
>> policy doesn't allow to do so.
>
>
> You might want to ask the company if they can lift that policy for the sake
> of saving everyone's time and energy. I know we have some professionals who
> would be willing to sign a reasonable NDA or whatnot to ensure security.
> -MC
>
>
>> But I can explain you more clearly about
>> the setup and my need.
>>
>> The following is the setup.
>> >From a Telephone Exchange, a line will be connected to the FS-BOX,  in
>> span1.
>> We have an internal Hard PBX with some 4 extensions.
>> We have connected the Hard PBX to FS-BOX in span2.
>>
>> I've configured span1 as PRI_CPE and span2 as PRI_NET.
>> span1 don't have any problems. Both incoming and outgoing works fine.
>> In span2, I was able to make incoming call. It rings the extension.
>> But when I make outgoing call from those extension, I got NO CIRCUIT OR
>> CHANNEL AVAILABLE.
>>
>> Right now in sangoma_mgd log, I got the following when I make outgoing
>> call.
>>
>> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Opening
>> /var/log/sangoma_pri/dchan_2.log
>> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
>> 3:134716232 id:65535]
>> Apr  8 10:20:21 FMS-FreeSwitch sangoma_prid: Rx Tsoft [s2c0 7:StatusIn
>> 4:134716308 id:65535]
>>
>> Moreover I also have an Asterisk Box with Digim card. It works fine there.
>> When i dial from extension, it reaches asterisk and it responds with
>> SETUP_ACK.
>>
>> please guide me.
>>
>>
>> On Wed, Apr 7, 2010 at 10:44 PM, David Yat Sin <david.yatsin at gmail.com>
>> wrote:
>> >
>> > Can you provide me with SSH access to that box and a phone number I can
>> use to trigger an incoming call on span 2 to that box?
>> >
>> > You can email me at: dyatsin at sangoma.com
>> >
>> > David
>> >
>> > On 4/7/2010 4:48 AM, lakshmanan ganapathy wrote:
>> >
>> > Hi,
>> > I have set verbose=4 in smg_pri.conf.
>> > I started the wanrouter.
>> > Here is the sangoma_mgd.log
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: ================System
>> restart=============
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>> Stack Daemon       =
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Version: 1.63
>>               =
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Date: Feb 26 2010
>>               =
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>> wanpipe-3.5.8.6       =
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 15607
>>                =
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid:
>> ===========================================
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Number of spans:2
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Verbosity set to:4
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Debug disabled
>> (local:2)
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: Log Boost disabled
>> (local:6)
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:1 pri_cpe
>> euroisdn dChan:16
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: TSoft:span:2 pri_net
>> euroisdn dChan:16
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Down
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(P) Local:
>> 127.0.0.66:53001 Remote:127.0.0.65:53001
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:(N) Local:
>> 127.0.0.66:53000 Remote:127.0.0.65:53000
>> > Apr  7 13:58:25 FMS-FreeSwitch sangoma_prid: BST:Version:103
>> > Apr  7 13:58:27 FMS-FreeSwitch sangoma_prid: s2:Status:Up prot:Up
>> > Apr  7 13:58:31 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Down
>> > Apr  7 13:58:32 FMS-FreeSwitch sangoma_prid: s1:Status:Up prot:Up
>> > Apr  7 13:58:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:58:37 last:13:58:26 grace:0)
>> > Apr  7 13:58:43 FMS-FreeSwitch sangoma_prid: Opening
>> /var/log/sangoma_pri/dchan_1.log
>> > Apr  7 13:58:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:58:47 last:13:58:37 grace:0)
>> > Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:58:57 last:13:58:47 grace:0)
>> > Apr  7 13:58:57 FMS-FreeSwitch sangoma_prid: Assuming application is
>> dead
>> > Apr  7 13:59:07 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:07 last:13:58:57 grace:0)
>> > Apr  7 13:59:17 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:17 last:13:59:07 grace:0)
>> > Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:27 last:13:59:17 grace:0)
>> > Apr  7 13:59:27 FMS-FreeSwitch sangoma_prid: Assuming application is
>> dead
>> > Apr  7 13:59:37 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:37 last:13:59:27 grace:0)
>> > Apr  7 13:59:47 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:47 last:13:59:37 grace:0)
>> > Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: HeartBeat timeout
>> (current:13:59:57 last:13:59:47 grace:0)
>> > Apr  7 13:59:57 FMS-FreeSwitch sangoma_prid: Assuming application is
>> dead
>> >
>> > >From freeswitch cli, if I say
>> > originate openzap/1/a/39114603 at g2 &park(), I got the following in the
>> sangoma_mgd.log.
>> >
>> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Outgoing call (Smg-ID:2)
>> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2:Outgoing call ChanRq:1
>> Called-Nb[39114603] Calling-Nb[Unknown] (Smg-ID:2)
>> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c1:Remote
>> released-Unallocated (unassigned) number(1)
>> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: s2c0:Call was already
>> cleared (TSOFT-ID:4)
>> > Apr  7 14:13:50 FMS-FreeSwitch sangoma_prid: g2:Call cleared (SMG-ID:2)
>> >
>> > By seeing this, I confirmed that span2 is working. Leave out the
>> Unallocated number, that's my internal problem.
>> >
>> > >>From the extension, if I dial 0, nothing is printing in
>> sangoma_mgd.log
>> > But in dhcan2.log I got the No/Circuit or channel available as I
>> mentioned earlier.
>> >
>> > In freeswitch mod_openzap is loaded properly.
>> > Any help!!!
>> >
>> >
>> > On Wed, Apr 7, 2010 at 2:17 AM, David Yat Sin <david.yatsin at gmail.com>
>> wrote:
>> >>
>> >> Hi Lakshmanan,
>> >> If you do not have anything printing in /var/log/sangoma_mgd.log, and
>> you have:
>> >> verbose=4           //(or higher)
>> >>
>> >> in /etc/wanpipe/smg_pri.conf, check that you have these lines in
>> /etc/syslog.conf (or /etc/rsyslog.conf):
>> >>
>> >> local2.*                /var/log/sangoma_mgd.log
>> >>
>> >> and restart your syslog.
>> >>
>> >>
>> >> my first guess is that you do not have openzap loaded so sangoma_prid
>> is rejecting all incoming calls, but I would need logs in
>> /var/log/sangoma_mgd.log to confirm.
>> >>
>> >> If openzap is not loaded, you can type:
>> >> load mod_openzap
>> >>
>> >> from the freeswitch CLI to load it.
>> >>
>> >> --
>> >>
>> >> David Yat Sin, BEng, Software Developer
>> >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
>> L3R 9T3 Canada
>> >> t. 1 905 474 1990 x 119 | e. dyatsin at sangoma.com
>> >>
>> >>
>> >>
>> >> On 4/6/2010 12:24 AM, lakshmanan ganapathy wrote:
>> >>
>> >> Hi all,
>> >> In my office we have a Hard PBX, with some 4 extensions.
>> >> We also have sangoma A102 card.
>> >> >From the Hard PBX, if 0 is pressed, it is setup in a way that it will
>> go to outside world.
>> >> I've connected that line to span2 of the card.
>> >> The span2 in the A102 card, is configured as PRI_NET.
>> >>
>> >> wanrouter status shows connected for the span2.
>> >>
>> >> But if I dial from the extension, I got the following in the
>> sangoma_dchan log.
>> >> 2010-04-03 12:49:49
>> >> INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34
>> 33 39 31 31 34 36 30 30 7d 02 91 81 ]
>> >>   Call Ref:0164
>> >>   Type:Setup (0x5)
>> >>   Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0)
>> TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3)
>> >>   Calling Party Number:4439114600(l:10) plan:isdn(1)
>> type:unknown(0)scr:user, passed(1) pres:allowed(0)
>> >>   High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ]
>> >>
>> >> 2010-04-03 12:49:49
>> >> OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ]
>> >>   Call Ref:0164
>> >>   Type:Release Compl (0x5a)
>> >>   Cause:coding:ITU-T(0) location:Public network, local user(2) val:No
>> Circuit/Channel Available(34)
>> >>
>> >> The call in not reaching freeswitch ( I enabled debug log. But nothing
>> is printing in it ).
>> >> Can someone suggest how to make this work.
>> >>
>> >> Please ask me if you need more information?, since I don't know what to
>> give 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Sat Apr 17 09:01:43 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 17 Apr 2010 09:01:43 -0700
Subject: [Freeswitch-users] Audio quality
Message-ID: <C7EF2977.887E%sean@obscuradigital.com>

Hello, 

Wanted to say thanks for all the great work the freeswitch dev team is
doing.  Also thanks for the support on this list.

My question is:

I have a polycom 321 phone and I?m having a audio issue when using the
speaker.  The inbound quality is really good, but the outgoing audio sounds
digitized or broken.  I?m curious is there a setting I can adjust to solve
this issue.   Outgoing quality is good when using the handset.  I have the
most recent svn checkout.

Has anyone ever experienced this issue?

Thanks in advance
Sean
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From chris at fowler.cc  Sat Apr 17 09:23:51 2010
From: chris at fowler.cc (Chris Fowler)
Date: Sat, 17 Apr 2010 12:23:51 -0400
Subject: [Freeswitch-users] Audio quality
In-Reply-To: <C7EF2977.887E%sean@obscuradigital.com>
References: <C7EF2977.887E%sean@obscuradigital.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C7DA80E6A@VMBX113.ihostexchange.net>

Hi Sean,

We have about 20 of the 321's running; the rest are 450's - total ~100 phones hooked into FS.

No similar audio problems - running with the standard 3.2.3 SPIP firmware from Polycom.
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html

Chris.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
Sent: Saturday, April 17, 2010 9:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Audio quality

Hello,

Wanted to say thanks for all the great work the freeswitch dev team is doing.  Also thanks for the support on this list.

My question is:

I have a polycom 321 phone and I'm having a audio issue when using the speaker.  The inbound quality is really good, but the outgoing audio sounds digitized or broken.  I'm curious is there a setting I can adjust to solve this issue.   Outgoing quality is good when using the handset.  I have the most recent svn checkout.

Has anyone ever experienced this issue?

Thanks in advance
Sean
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From sean at obscuradigital.com  Sat Apr 17 10:10:43 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 17 Apr 2010 10:10:43 -0700
Subject: [Freeswitch-users] Audio quality
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C7DA80E6A@VMBX113.ihostexchange.net>
Message-ID: <C7EF39A3.8883%sean@obscuradigital.com>

Yea I?m running the most current Polycom firmware release.  Not sure if it?s
an isolated incident.  Not sure the best way to troubleshoot this issue
short of the standard siptrace tool and wireshark.

What codec are you using?

Sean

On 4/17/10 9:23 AM, "Chris Fowler" <chris at fowler.cc> wrote:

> Hi Sean,
>  
> We have about 20 of the 321?s running; the rest are 450?s ? total ~100 phones
> hooked into FS.
>  
> No similar audio problems ? running with the standard 3.2.3 SPIP firmware from
> Polycom.
> http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
>  
> Chris.
>  
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
> Sent: Saturday, April 17, 2010 9:02 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Audio quality
>  
> Hello, 
> 
> Wanted to say thanks for all the great work the freeswitch dev team is doing.
> Also thanks for the support on this list.
> 
> My question is:
> 
> I have a polycom 321 phone and I?m having a audio issue when using the
> speaker.  The inbound quality is really good, but the outgoing audio sounds
> digitized or broken.  I?m curious is there a setting I can adjust to solve
> this issue.   Outgoing quality is good when using the handset.  I have the
> most recent svn checkout.
> 
> Has anyone ever experienced this issue?
> 
> Thanks in advance
> Sean 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Apr 17 10:44:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Apr 2010 12:44:38 -0500
Subject: [Freeswitch-users] How to setup a span as PRI_NET
In-Reply-To: <h2y191c3a031004171043h38c7a1e8yddf5265a1307d31b@mail.gmail.com>
References: <y2s7d79b3931004052124gdff94381h85b1789ca1fa2300@mail.gmail.com>
	<4BBB9DEB.2040702@gmail.com>
	<v2t7d79b3931004070148w3cc64e67s22defeb353e0af14@mail.gmail.com>
	<4BBCBD73.7020206@gmail.com>
	<x2n7d79b3931004082138y6573df06hcfd093103a3f944b@mail.gmail.com>
	<n2t87f2f3b91004082241m1262877fsb2b596671c3d5f2@mail.gmail.com>
	<y2n7d79b3931004170212g4e302cadz9b952003db556c78@mail.gmail.com>
	<h2y191c3a031004171043h38c7a1e8yddf5265a1307d31b@mail.gmail.com>
Message-ID: <x2s191c3a031004171044k8d7b984et1f3bf619ea73a550@mail.gmail.com>

You just took a step backwards fyi.
You *were* talking to the sangoma engineers.
The guys who actually wrote it.

Now you are just talking to tier 1 support.

On Apr 17, 2010 4:18 AM, "lakshmanan ganapathy" <lakindia89 at gmail.com>
wrote:

Hi all,
         I've contacted the sangoma support, and they are looking into the
issue.
They also told that the problem is in the PRI STACK.
So in the mean time, I planned to change the stack and check. So I switched
to libpri.
I've configured the things and both Incoming and outgoing works on span1.
On span2, from the extension, If I dial, I got

2010-04-17 10:51:15.218100 [WARNING] ozmod_libpri.c:760 --Duplicate Ring on
channel 2:-1 (ignored)

The freeswitch log is:
     http://pastebin.freeswitch.org/12738
I also went to see what is the problem. It seems to me that, when SETUP
comes, ozmod_libpri.c is checking for the B-channel number.
Since it is not there, it print Duplicate ring on channel.

I also came to know that, first SETUP will request the telco (B-chan
identification won't be there). Then the telco responds with the B-channel
to use in SETUP_ACK.
The further communication will happen with the B-chan identification.

Am  I correct!!. Please give inputs to me to solve this issue.




On Fri, Apr 9, 2010 at 11:11 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
>
> On Thu, Apr ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From sos at sokhapkin.dyndns.org  Sat Apr 17 06:26:24 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 17 Apr 2010 09:26:24 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC core?
Message-ID: <201004170926.24979.sos@sokhapkin.dyndns.org>

I want to utilize nibblebill features for prepaid support, but as soon as I 
enable nibble billing with

set nibble_rate=XXX
set nibble_account=NNNNNN
set enable_heartbeat_events=60

in dialplan, FS process begins to eat more and more memory, RSS grows from 23M 
to 100M and more after processing few thousands calls and continues to grow. 
If I comment out those 3 lines from dialplan, then FS RSS grows from initial 
23M to 50-60M (depending on the number of concurrent calls) and stays at this 
value, no memory leaks.

Do anybody use mod_nibblebill?

I'm experiencing this problem with all svn versions I tried including latest 
git version.


From mike at jerris.com  Sat Apr 17 11:34:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Apr 2010 14:34:05 -0400
Subject: [Freeswitch-users] FreeSwitch 1.0.6 - Core Dumped with libpri
In-Reply-To: <q2m7d79b3931004162210q10db9fa7m9c08b9837746081a@mail.gmail.com>
References: <q2t7d79b3931004150411g3fed85b2t82e142564eb0e1b8@mail.gmail.com>
	<w2y7d79b3931004150423g7b75a91dm67993bb89706489c@mail.gmail.com>
	<8DF0B10E-AD0B-4088-B08A-DC5AE303F055@jerris.com>
	<u2p191c3a031004150751j858ddb47ud97fe13636ece039@mail.gmail.com>
	<t2j7d79b3931004160246s41aa251as46646e0465178b97@mail.gmail.com>
	<i2v191c3a031004160657o2926bfc2pd382b1396ad7d544@mail.gmail.com>
	<n2lc4d05cbe1004160715nfa67a4d4uae99e0edbcb07fb6@mail.gmail.com>
	<q2m7d79b3931004162210q10db9fa7m9c08b9837746081a@mail.gmail.com>
Message-ID: <710F7335-ED1C-4F64-8EE5-0B9159BC85A8@jerris.com>

If this core dump still exists in the current git head, please file a bug on jira for it so we can make sure to fix the core dump.

Mike

On Apr 17, 2010, at 1:10 AM, lakshmanan ganapathy wrote:

> Moises,
> You are right. sangoma_prid is running on my system.
> I gave smg_ctrl stop.
> Then I started freeswitch.
> Now it didn't get core-dumped.
> Thanks for your help, updated the wiki.
> 
> On Fri, Apr 16, 2010 at 7:45 PM, Moises Silva <moises.silva at gmail.com> wrote:
> May be you have still running sangoma_prid ( which opens the d-channel ) and then FreeSWITCH fails to open it (just a guess).
> 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> 
> 
> 
> On Fri, Apr 16, 2010 at 9:57 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> I think you must be.
> Was the bt the same this time?
> Maybe if you hop on irc we can have someone look at it for you.
> 
> 

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From mike at jerris.com  Sat Apr 17 12:18:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Apr 2010 15:18:47 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004170926.24979.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
Message-ID: <683E9FF7-1D74-4C7F-A3EF-92F972936BC2@jerris.com>

100M may just be the memory it is really using for what it is doing and not a leak.  Have you seen anything to tell you this is actually a leak?   Have you run under valgrind or some similar tool to confirm?

Mike

On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:

> I want to utilize nibblebill features for prepaid support, but as soon as I 
> enable nibble billing with
> 
> set nibble_rate=XXX
> set nibble_account=NNNNNN
> set enable_heartbeat_events=60
> 
> in dialplan, FS process begins to eat more and more memory, RSS grows from 23M 
> to 100M and more after processing few thousands calls and continues to grow. 
> If I comment out those 3 lines from dialplan, then FS RSS grows from initial 
> 23M to 50-60M (depending on the number of concurrent calls) and stays at this 
> value, no memory leaks.
> 
> Do anybody use mod_nibblebill?
> 
> I'm experiencing this problem with all svn versions I tried including latest 
> git version.



From sos at sokhapkin.dyndns.org  Sat Apr 17 13:34:29 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 17 Apr 2010 16:34:29 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <683E9FF7-1D74-4C7F-A3EF-92F972936BC2@jerris.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<683E9FF7-1D74-4C7F-A3EF-92F972936BC2@jerris.com>
Message-ID: <201004171634.29444.sos@sokhapkin.dyndns.org>

When memory stays at 50-60M without nibblebill and grows to 300M with... It's 
definetily a leak. valgring log is available, but there is nothing suspicious 
in it... Try to enable niblebill yourself.

On Saturday 17 April 2010, Michael Jerris wrote:
> 100M may just be the memory it is really using for what it is doing and not
>  a leak.  Have you seen anything to tell you this is actually a leak?  
>  Have you run under valgrind or some similar tool to confirm?
> 
> Mike
> 
> On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:
> > I want to utilize nibblebill features for prepaid support, but as soon as
> > I enable nibble billing with
> >
> > set nibble_rate=XXX
> > set nibble_account=NNNNNN
> > set enable_heartbeat_events=60
> >
> > in dialplan, FS process begins to eat more and more memory, RSS grows
> > from 23M to 100M and more after processing few thousands calls and
> > continues to grow. If I comment out those 3 lines from dialplan, then FS
> > RSS grows from initial 23M to 50-60M (depending on the number of
> > concurrent calls) and stays at this value, no memory leaks.
> >
> > Do anybody use mod_nibblebill?
> >
> > I'm experiencing this problem with all svn versions I tried including
> > latest git version.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From rupa at rupa.com  Sat Apr 17 15:20:09 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 17 Apr 2010 17:20:09 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004171634.29444.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<683E9FF7-1D74-4C7F-A3EF-92F972936BC2@jerris.com>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
Message-ID: <v2jb39273531004171520k7e3ec916x73c2af32f4d4f730@mail.gmail.com>

Are you using anything else that uses ODBC?  You may be seeing the initial
hit of ODBC. I use nibblebill (lightly) without any significant memory
utilization that I can see.  But then, I don't look at my memory usage so
closely.  300M isn't that much...

On Sat, Apr 17, 2010 at 3:34 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> When memory stays at 50-60M without nibblebill and grows to 300M with...
> It's
> definetily a leak. valgring log is available, but there is nothing
> suspicious
> in it... Try to enable niblebill yourself.
>
> On Saturday 17 April 2010, Michael Jerris wrote:
> > 100M may just be the memory it is really using for what it is doing and
> not
> >  a leak.  Have you seen anything to tell you this is actually a leak?
> >  Have you run under valgrind or some similar tool to confirm?
> >
> > Mike
> >
> > On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:
> > > I want to utilize nibblebill features for prepaid support, but as soon
> as
> > > I enable nibble billing with
> > >
> > > set nibble_rate=XXX
> > > set nibble_account=NNNNNN
> > > set enable_heartbeat_events=60
> > >
> > > in dialplan, FS process begins to eat more and more memory, RSS grows
> > > from 23M to 100M and more after processing few thousands calls and
> > > continues to grow. If I comment out those 3 lines from dialplan, then
> FS
> > > RSS grows from initial 23M to 50-60M (depending on the number of
> > > concurrent calls) and stays at this value, no memory leaks.
> > >
> > > Do anybody use mod_nibblebill?
> > >
> > > I'm experiencing this problem with all svn versions I tried including
> > > latest git version.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From sos at sokhapkin.dyndns.org  Sat Apr 17 15:46:56 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 17 Apr 2010 18:46:56 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <v2jb39273531004171520k7e3ec916x73c2af32f4d4f730@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
	<v2jb39273531004171520k7e3ec916x73c2af32f4d4f730@mail.gmail.com>
Message-ID: <201004171846.56193.sos@sokhapkin.dyndns.org>

No, I do not use core ODBC. I run lua with odbc support, but to check the 
problem I got rid of lua and reinvited the wheel with mod_xml_curl to build 
the dialplan. Absolutely the same memory leak.

FS takes about 60M RSS without mod_nibblebill but RSS grows up to infinity 
when mod_nibblebill is enabled.

On Saturday 17 April 2010, Rupa Schomaker wrote:
> Are you using anything else that uses ODBC?  You may be seeing the initial
> hit of ODBC. I use nibblebill (lightly) without any significant memory
> utilization that I can see.  But then, I don't look at my memory usage so
> closely.  300M isn't that much...
> 
> On Sat, Apr 17, 2010 at 3:34 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > When memory stays at 50-60M without nibblebill and grows to 300M with...
> > It's
> > definetily a leak. valgring log is available, but there is nothing
> > suspicious
> > in it... Try to enable niblebill yourself.
> >
> > On Saturday 17 April 2010, Michael Jerris wrote:
> > > 100M may just be the memory it is really using for what it is doing and
> >
> > not
> >
> > >  a leak.  Have you seen anything to tell you this is actually a leak?
> > >  Have you run under valgrind or some similar tool to confirm?
> > >
> > > Mike
> > >
> > > On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:
> > > > I want to utilize nibblebill features for prepaid support, but as
> > > > soon
> >
> > as
> >
> > > > I enable nibble billing with
> > > >
> > > > set nibble_rate=XXX
> > > > set nibble_account=NNNNNN
> > > > set enable_heartbeat_events=60
> > > >
> > > > in dialplan, FS process begins to eat more and more memory, RSS grows
> > > > from 23M to 100M and more after processing few thousands calls and
> > > > continues to grow. If I comment out those 3 lines from dialplan, then
> >
> > FS
> >
> > > > RSS grows from initial 23M to 50-60M (depending on the number of
> > > > concurrent calls) and stays at this value, no memory leaks.
> > > >
> > > > Do anybody use mod_nibblebill?
> > > >
> > > > I'm experiencing this problem with all svn versions I tried including
> > > > latest git version.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From bruce at nani.ca  Sat Apr 17 16:29:35 2010
From: bruce at nani.ca (Bruce Fletcher)
Date: Sat, 17 Apr 2010 16:29:35 -0700
Subject: [Freeswitch-users] mod_pocketsphinx link error on Mac OS X 10.6
	(solved)
In-Reply-To: <18B39E9E-4212-45D0-9F24-34DCEF3CAF45@nani.ca>
References: <18B39E9E-4212-45D0-9F24-34DCEF3CAF45@nani.ca>
Message-ID: <51F7D6A4-15A0-4741-85FE-0DE0E3F3EE02@nani.ca>

Thanks to bkw_ and some other friendly IRC helpers, it has been established that indeed the MacPorts install that I mentioned below was the source of my _iconv linker problem.  I'll just record a few things here on the mailing list as Google fodder in case someone else hits this same problem, and maybe make a small note somewhere in the wiki.

FreeSWITCH on Mac OS X assumes a standard Mac OS X build environment, and it appears that MacPorts can make things a bit non-standard.  I only had MacPorts installed to get the git-core port, and I've verified that uninstalling MacPorts and installing git from a disk image works fine.  Disk images of git are available here:

 http://code.google.com/p/git-osx-installer/downloads/list?can=3

Two other experiments to report.  I removed the MacPorts directories of /opt/local/bin and /opt/local/sbin from my $PATH and FreeSWITCH built and loaded fine, although the above git disk image install was required for 'make current'.  I then tried leaving the /opt dirs at the end of my path, and again ran into the _iconv problem below.  So if you need MacPorts on your computer for some other reason, you may have to selectively disable it in your $PATH whenever you want to build FreeSWITCH unless you can come up with some more elegant solution than I could.  I was a little unimpressed with how installing git from MacPorts required 17 other ports first, so I'm happy to go the "git from .dmg" route myself.

Thanks again everyone,
- Bruce


On 2010-04-16, at 12:11 AM, Bruce Fletcher wrote:

> I'm having trouble with mod_pocketsphinx on Mac OS X 10.6 using git head from Wed Apr 14 21:36:41 2010 -0400.
> 
> My freeswitch.log says:
> 
> 2010-04-15 22:50:30.418812 [NOTICE] switch_loadable_module.c:272 Adding API Function 'luarun'
> 2010-04-15 22:50:30.418863 [NOTICE] switch_loadable_module.c:272 Adding API Function 'lua'
> 2010-04-15 22:50:30.419302 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so
> **dlopen(/usr/local/freeswitch/mod/mod_pocketsphinx.so, 6): Symbol not found: _iconv
> Referenced from: /usr/local/freeswitch/mod/mod_pocketsphinx.so
> Expected in: flat namespace
> in /usr/local/freeswitch/mod/mod_pocketsphinx.so**
> 
>> From my freeswitch.la I see:
> 
> # Libraries that this one depends upon.
> dependency_libs=' -L/Users/admin/src/freeswitch/libs/apr-util/xml/expat/lib /Users/admin/src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la /Users/admin/src/freeswitch/libs/apr/libapr-1.la -liconv -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -lncurses'
> 
> That looks OK to my unskilled eyes, although I do have libiconv in two places, the default /usr/lib:
> 
> lrwxr-xr-x  1 root  wheel       16 Sep 10  2009 /usr/lib/libiconv.2.4.0.dylib -> libiconv.2.dylib
> -r-xr-xr-x  1 root  wheel  3205760 Nov 19 18:00 /usr/lib/libiconv.2.dylib
> lrwxr-xr-x  1 root  wheel       20 Sep 10  2009 /usr/lib/libiconv.dylib -> libiconv.2.4.0.dylib
> 
> and a copy from MacPorts, which came for the ride when I installed git:
> 
> -rw-r--r--  2 root  admin  1084720 Apr  2 12:40 /opt/local/lib/libiconv.2.dylib
> -rw-r--r--  2 root  admin  1115488 Apr  2 12:40 /opt/local/lib/libiconv.a
> lrw-r--r--  1 root  admin       16 Apr  2 12:40 /opt/local/lib/libiconv.dylib -> libiconv.2.dylib
> -rw-r--r--  2 root  admin      918 Apr  2 12:40 /opt/local/lib/libiconv.la
> 
> Does anyone have any thoughts on how I can debug this?
> 
> Thanks,
> - Bruce
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chris at fowler.cc  Sat Apr 17 19:06:24 2010
From: chris at fowler.cc (Chris Fowler)
Date: Sat, 17 Apr 2010 22:06:24 -0400
Subject: [Freeswitch-users] Audio quality
In-Reply-To: <C7EF39A3.8883%sean@obscuradigital.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C7DA80E6A@VMBX113.ihostexchange.net>
	<C7EF39A3.8883%sean@obscuradigital.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C7DA80EB6@VMBX113.ihostexchange.net>

>> What codec are you using?

For the 321's PCMU.

C.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
Sent: Saturday, April 17, 2010 10:11 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Audio quality

Yea I'm running the most current Polycom firmware release.  Not sure if it's an isolated incident.  Not sure the best way to troubleshoot this issue short of the standard siptrace tool and wireshark.

What codec are you using?

Sean

On 4/17/10 9:23 AM, "Chris Fowler" <chris at fowler.cc> wrote:
Hi Sean,

We have about 20 of the 321's running; the rest are 450's - total ~100 phones hooked into FS.

No similar audio problems - running with the standard 3.2.3 SPIP firmware from Polycom.
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html

Chris.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
Sent: Saturday, April 17, 2010 9:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Audio quality

Hello,

Wanted to say thanks for all the great work the freeswitch dev team is doing.  Also thanks for the support on this list.

My question is:

I have a polycom 321 phone and I'm having a audio issue when using the speaker.  The inbound quality is really good, but the outgoing audio sounds digitized or broken.  I'm curious is there a setting I can adjust to solve this issue.   Outgoing quality is good when using the handset.  I have the most recent svn checkout.

Has anyone ever experienced this issue?

Thanks in advance
Sean
________________________________
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Sat Apr 17 20:23:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Apr 2010 23:23:48 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004171634.29444.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<683E9FF7-1D74-4C7F-A3EF-92F972936BC2@jerris.com>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
Message-ID: <911AA46B-E9D3-441C-A31D-06C404DB197F@jerris.com>

That is a completely incorrect statement.  Using memory (and caching possibly), leaking memory, and unbound growth in memory are all totally different things.  to understand exactly what is going on here, the distinction is important.  If you run under valgrind, you should look when its running to see where all this extra allocation is.

Mike

On Apr 17, 2010, at 4:34 PM, Sergey Okhapkin wrote:

> When memory stays at 50-60M without nibblebill and grows to 300M with... It's 
> definetily a leak. valgring log is available, but there is nothing suspicious 
> in it... Try to enable niblebill yourself.
> 
> On Saturday 17 April 2010, Michael Jerris wrote:
>> 100M may just be the memory it is really using for what it is doing and not
>> a leak.  Have you seen anything to tell you this is actually a leak?  
>> Have you run under valgrind or some similar tool to confirm?
>> 
>> Mike
>> 
>> On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:
>>> I want to utilize nibblebill features for prepaid support, but as soon as
>>> I enable nibble billing with
>>> 
>>> set nibble_rate=XXX
>>> set nibble_account=NNNNNN
>>> set enable_heartbeat_events=60
>>> 
>>> in dialplan, FS process begins to eat more and more memory, RSS grows
>>> from 23M to 100M and more after processing few thousands calls and
>>> continues to grow. If I comment out those 3 lines from dialplan, then FS
>>> RSS grows from initial 23M to 50-60M (depending on the number of
>>> concurrent calls) and stays at this value, no memory leaks.
>>> 
>>> Do anybody use mod_nibblebill?
>>> 
>>> I'm experiencing this problem with all svn versions I tried including
>>> latest git version.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dome at tel.co.th  Sat Apr 17 21:47:10 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 18 Apr 2010 11:47:10 +0700
Subject: [Freeswitch-users] looking for someone who working on mod_redis
Message-ID: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>

Dear Sir,
              Now i'm using tokyo tyrant with my application by
mod_memcache. i have plan to move to redis but redis not complatible
with memcache.
So if someone working on mod_redis please let's me know.

BG

Dome C.


From john_re at fastmail.us  Sat Apr 17 22:18:29 2010
From: john_re at fastmail.us (giovanni_re)
Date: Sat, 17 Apr 2010 22:18:29 -0700
Subject: [Freeswitch-users] VOIP at BerkeleyTIP-Global meeting on Sunday
 April 18 12N-3P, & April 27
Message-ID: <1271567909.19399.1370527791@webmail.messagingengine.com>

Come discuss VOIP. :)  Join via VOIP or come to Berkeley
http://sites.google.com/site/berkeleytip/voice-voip-conferencing
FSCafe at Moffitt at UCBerkeley, opens 1pm, but can connect from outside
at 12N.

Hot topics: Ubuntu 10.04, Free Culuture, VOIP, Set up the web server &
mail list & asterisk/freeswitch on the BTIP box with Ubuntu 10.04?

Tues April 27 5-6P VOIP online meeting also.

http://sites.google.com/site/berkeleytip/

Join the mail list, tell us what you're interested in.
http://sites.google.com/site/berkeleytip/mailing-lists


From mike at jerris.com  Sat Apr 17 22:52:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 18 Apr 2010 01:52:27 -0400
Subject: [Freeswitch-users] looking for someone who working on mod_redis
In-Reply-To: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>
References: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>
Message-ID: <DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>

I spent some time looking at credis, but I think something better is probably needed, it may work, but its not architected with a lot of thought to performing as well as it could or working well in general.  Also, it only supports fairly old redis.  A better approach is probably building a stand alone lib in the redis tree itself.  the redis author says he has one or the next major version that will be checked in at some point.

Mike

On Apr 18, 2010, at 12:47 AM, Dome Charoenyost wrote:

> Dear Sir,
>              Now i'm using tokyo tyrant with my application by
> mod_memcache. i have plan to move to redis but redis not complatible
> with memcache.
> So if someone working on mod_redis please let's me know.
> 
> BG
> 
> Dome C.



From bekelemartins at gmail.com  Sun Apr 18 00:37:26 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Sun, 18 Apr 2010 03:37:26 -0400
Subject: [Freeswitch-users] Lua session disappears after failed call,
	so hangup cause becomes 	inaccessable
Message-ID: <q2w5cc9e8f31004180037v414fcf97n5db7fccdc657d224@mail.gmail.com>

Hi, I want my lua script to dial a phone number and return back the
disposition code (hangup cause). The problem is if the call fails, the
session variable is destroyed, and the call disposition code is destroyed as
well.

For example:

function make_call()
  sessiondata = "sofia/gateway/myout/15555555555"
  new_session = freeswitch.Session(sessiondata)
  if (new_session:ready()) then
    new_session:streamFile("hello.wav")
    disposition = "ANSWERED"
  else
    disposition = new_session:getVariable("hangup_cause")
    freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
    disposition = new_session:hangupCause()
    freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
  end
  return(disposition)
end
make_call()


This returns the following errors:

2010-04-18 03:29:58 [NOTICE] switch_channel.c:602 switch_channel_set_name()
New Channel sofia/external/15555555555
[4bfbb177-2eee-422d-8f2a-5f9d71082254]
2010-04-18 03:30:04 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup
sofia/external/15555555555 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
API CALL [lua(example.lua)] output:
-ERR encounterd

2010-04-18 03:30:04 [ERR] freeswitch_lua.cpp:102 ready() session is not
initalized
2010-04-18 03:30:04 [ERR] switch_cpp.cpp:600 getVariable() session is not
initalized
2010-04-18 03:30:04 [INFO] switch_cpp.cpp:1122 console_log() Hangup cause =
2010-04-18 03:30:04 [ERR] mod_lua.cpp:182 lua_parse_and_execute()
/usr/local/freeswitch/scripts/example.lua:10: attempt to call method
'hangupCause' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/example.lua:10: in function
'make_call'
        /usr/local/freeswitch/scripts/example.lua:16: in main chunk
2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1085
switch_core_session_thread() Session 1 (sofia/external/15555555555) Ended
2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1087
switch_core_session_thread() Close Channel sofia/external/15555555555
[CS_DESTROY]

Is there some way I can accomplish this?
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From sos at sokhapkin.dyndns.org  Sun Apr 18 05:28:04 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 18 Apr 2010 08:28:04 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <911AA46B-E9D3-441C-A31D-06C404DB197F@jerris.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
	<911AA46B-E9D3-441C-A31D-06C404DB197F@jerris.com>
Message-ID: <201004180828.04809.sos@sokhapkin.dyndns.org>

According to valgrind there are no big leaks. 

==12476== LEAK SUMMARY:
==12476==    definitely lost: 146,309 bytes in 189 blocks.
==12476==    indirectly lost: 176 bytes in 12 blocks.
==12476==      possibly lost: 12,336 bytes in 18 blocks.
==12476==    still reachable: 1,077,431 bytes in 550 blocks.
==12476==         suppressed: 0 bytes in 0 blocks.



On Saturday 17 April 2010, Michael Jerris wrote:
> That is a completely incorrect statement.  Using memory (and caching
>  possibly), leaking memory, and unbound growth in memory are all totally
>  different things.  to understand exactly what is going on here, the
>  distinction is important.  If you run under valgrind, you should look when
>  its running to see where all this extra allocation is.
> 
> Mike
> 
> On Apr 17, 2010, at 4:34 PM, Sergey Okhapkin wrote:
> > When memory stays at 50-60M without nibblebill and grows to 300M with...
> > It's definetily a leak. valgring log is available, but there is nothing
> > suspicious in it... Try to enable niblebill yourself.
> >
> > On Saturday 17 April 2010, Michael Jerris wrote:
> >> 100M may just be the memory it is really using for what it is doing and
> >> not a leak.  Have you seen anything to tell you this is actually a leak?
> >> Have you run under valgrind or some similar tool to confirm?
> >>
> >> Mike
> >>
> >> On Apr 17, 2010, at 9:26 AM, Sergey Okhapkin wrote:
> >>> I want to utilize nibblebill features for prepaid support, but as soon
> >>> as I enable nibble billing with
> >>>
> >>> set nibble_rate=XXX
> >>> set nibble_account=NNNNNN
> >>> set enable_heartbeat_events=60
> >>>
> >>> in dialplan, FS process begins to eat more and more memory, RSS grows
> >>> from 23M to 100M and more after processing few thousands calls and
> >>> continues to grow. If I comment out those 3 lines from dialplan, then
> >>> FS RSS grows from initial 23M to 50-60M (depending on the number of
> >>> concurrent calls) and stays at this value, no memory leaks.
> >>>
> >>> Do anybody use mod_nibblebill?
> >>>
> >>> I'm experiencing this problem with all svn versions I tried including
> >>> latest git version.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From vfclists at googlemail.com  Sun Apr 18 08:55:49 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 18 Apr 2010 16:55:49 +0100
Subject: [Freeswitch-users] Getting script to pass variable to dialplan
Message-ID: <p2n7abab2411004180855sf686dcael107a240532b4b284@mail.gmail.com>

I have a simple AGI based dialplan that I want a dialplan equivalent of.

The agi script checks the database if the extension making the call is
enabled, if it is not it hangups the call, or allows the call to continue
inside the dialplan. Do the scripting languages allow changing or setting of
variables that will be used further on in the dialplan?

Can they create new custom variables that can be used to direct the
dialplan?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From anthony.minessale at gmail.com  Sun Apr 18 09:35:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 18 Apr 2010 11:35:00 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004180828.04809.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
	<911AA46B-E9D3-441C-A31D-06C404DB197F@jerris.com>
	<201004180828.04809.sos@sokhapkin.dyndns.org>
Message-ID: <p2z191c3a031004180935g23a12e71n58db03227e5133bb@mail.gmail.com>

That's not very useful you need a full report with extended checking.

On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:

According to valgrind there are no big leaks.

==12476== LEAK SUMMARY:
==12476==    definitely lost: 146,309 bytes in 189 blocks.
==12476==    indirectly lost: 176 bytes in 12 blocks.
==12476==      possibly lost: 12,336 bytes in 18 blocks.
==12476==    still reachable: 1,077,431 bytes in 550 blocks.
==12476==         suppressed: 0 bytes in 0 blocks.




On Saturday 17 April 2010, Michael Jerris wrote:
> That is a completely incorrect statement. Usi...
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From anthony.minessale at gmail.com  Sun Apr 18 09:35:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 18 Apr 2010 11:35:58 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <p2z191c3a031004180935g23a12e71n58db03227e5133bb@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004171634.29444.sos@sokhapkin.dyndns.org>
	<911AA46B-E9D3-441C-A31D-06C404DB197F@jerris.com>
	<201004180828.04809.sos@sokhapkin.dyndns.org>
	<p2z191c3a031004180935g23a12e71n58db03227e5133bb@mail.gmail.com>
Message-ID: <w2t191c3a031004180935x41838351ma2d0f08cf6b99ebc@mail.gmail.com>

Also thatb suggests you leaked a whopping 100k

On Apr 18, 2010 11:35 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

That's not very useful you need a full report with extended checking.


>
> On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
wrote:
>
> According to v...

>
>
>
>
> On Saturday 17 April 2010, Michael Jerris wrote:
> That is a completely incorrect statement. Usi...
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From sos at sokhapkin.dyndns.org  Sun Apr 18 09:48:32 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 18 Apr 2010 12:48:32 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <w2t191c3a031004180935x41838351ma2d0f08cf6b99ebc@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<p2z191c3a031004180935g23a12e71n58db03227e5133bb@mail.gmail.com>
	<w2t191c3a031004180935x41838351ma2d0f08cf6b99ebc@mail.gmail.com>
Message-ID: <201004181248.33175.sos@sokhapkin.dyndns.org>

Which valgrind options should I specify? I did run

valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-
resolution=high --show-reachable=yes ./freeswitch -nonat -vg

100K leaked is nothing compared to FS process RSS of 300M...

On Sunday 18 April 2010, Anthony Minessale wrote:
> Also thatb suggests you leaked a whopping 100k
> 
> On Apr 18, 2010 11:35 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
> wrote:
> 
> That's not very useful you need a full report with extended checking.
> 
> > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> 
> wrote:
> > According to v...
> >
> >
> >
> >
> >
> > On Saturday 17 April 2010, Michael Jerris wrote:
> > That is a completely incorrect statement. Usi...
> 



From dome at tel.co.th  Sun Apr 18 10:52:23 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 19 Apr 2010 00:52:23 +0700
Subject: [Freeswitch-users] looking for someone who working on mod_redis
In-Reply-To: <DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>
References: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>
	<DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>
Message-ID: <h2u8ccbff061004181052yedd733ccl1fb9ad0a17ac551d@mail.gmail.com>

2010/4/18 Michael Jerris <mike at jerris.com>:
> I spent some time looking at credis, but I think something better is probably needed, it may work, but its not architected with a lot of thought to performing as well as it could or working well in general. ?Also, it only supports fairly old redis. ?A better approach is probably building a stand alone lib in the redis tree itself. ?the redis author says he has one or the next major version that will be checked in at some point.
>
So waiting for lib redis better :)

Dome C.

> Mike
>
> On Apr 18, 2010, at 12:47 AM, Dome Charoenyost wrote:
>
>> Dear Sir,
>> ? ? ? ? ? ? ?Now i'm using tokyo tyrant with my application by
>> mod_memcache. i have plan to move to redis but redis not complatible
>> with memcache.
>> So if someone working on mod_redis please let's me know.
>>
>> BG
>>
>> Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sun Apr 18 11:31:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 18 Apr 2010 14:31:50 -0400
Subject: [Freeswitch-users] looking for someone who working on mod_redis
In-Reply-To: <h2u8ccbff061004181052yedd733ccl1fb9ad0a17ac551d@mail.gmail.com>
References: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>
	<DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>
	<h2u8ccbff061004181052yedd733ccl1fb9ad0a17ac551d@mail.gmail.com>
Message-ID: <FF85A587-EA25-4EC5-B8C5-321AD301B3BA@jerris.com>

Or writing one.  I have some stuff local that started on fixing bits that would be needed for a cross platform lib out of the redis tree, but I don't think its even in a compiling state yet.

Mike

On Apr 18, 2010, at 1:52 PM, Dome Charoenyost wrote:

> 2010/4/18 Michael Jerris <mike at jerris.com>:
>> I spent some time looking at credis, but I think something better is probably needed, it may work, but its not architected with a lot of thought to performing as well as it could or working well in general.  Also, it only supports fairly old redis.  A better approach is probably building a stand alone lib in the redis tree itself.  the redis author says he has one or the next major version that will be checked in at some point.
>> 
> So waiting for lib redis better :)
> 
> Dome C.
> 
>> Mike
>> 
>> On Apr 18, 2010, at 12:47 AM, Dome Charoenyost wrote:
>> 
>>> Dear Sir,
>>>              Now i'm using tokyo tyrant with my application by
>>> mod_memcache. i have plan to move to redis but redis not complatible
>>> with memcache.
>>> So if someone working on mod_redis please let's me know.
>>> 
>>> BG
>>> 
>>> Dome C.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From neilp at cs.stanford.edu  Sun Apr 18 13:48:39 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sun, 18 Apr 2010 13:48:39 -0700
Subject: [Freeswitch-users] outbound call with sip gateway
Message-ID: <g2we5503bdb1004181348xc47513fdld80899bc61bf02e0@mail.gmail.com>

A newbie question:

I have setup a SIP gateway using
iptel<http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel>.
When I try and call my phone using this code:

new_session = freeswitch.Session("sofia/gateway/iptel/<my_mobile_num>")

... I get the following error:
2010-04-18 13:40:52.082802 [NOTICE] mod_sofia.c:1907 Pre-Answer
sofia/internal/1001@<server>!
2010-04-18 13:40:52.082802 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1234 at conference.freeswitch.org[ff407e08-e1cf-4d2b-8f83-d30d50431680]
2010-04-18 13:40:52.337877 [INFO] switch_rtp.c:2049 Auto Changing port from
<ip1> to <ip2>
2010-04-18 13:40:52.591882 [NOTICE] sofia.c:4789 Hangup sofia/internal/
1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2010-04-18 13:40:52.597793 [ERR] mod_conference.c:4563 Cannot create
outgoing channel, cause: NO_USER_RESPONSE

This error comes from the called party not responding with an alert or
connect indication. How do I make my endpoint respond? Besides a phone
number, what else can I set up as an endpoint and how?

Thanks in advance,
Neil
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From pjintheusa at gmail.com  Sun Apr 18 14:04:40 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sun, 18 Apr 2010 17:04:40 -0400
Subject: [Freeswitch-users] Getting script to pass variable to dialplan
In-Reply-To: <p2n7abab2411004180855sf686dcael107a240532b4b284@mail.gmail.com>
References: <p2n7abab2411004180855sf686dcael107a240532b4b284@mail.gmail.com>
Message-ID: <k2p367751821004181404vc24638c6s3183e357eb5e7cc5@mail.gmail.com>

You can set / get custom variables as follows:

var TheVariable = session.getVariable("TheVariableName")

session.setVariable("TheVariableName", "This is the val")


On Sun, Apr 18, 2010 at 11:55 AM, Frank Church <vfclists at googlemail.com>wrote:

> I have a simple AGI based dialplan that I want a dialplan equivalent of.
>
> The agi script checks the database if the extension making the call is
> enabled, if it is not it hangups the call, or allows the call to continue
> inside the dialplan. Do the scripting languages allow changing or setting of
> variables that will be used further on in the dialplan?
>
> Can they create new custom variables that can be used to direct the
> dialplan?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 bekelemartins at gmail.com  Sun Apr 18 18:43:34 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Sun, 18 Apr 2010 21:43:34 -0400
Subject: [Freeswitch-users] Lua session disappears after failed call,
	so hangup cause becomes 	inaccessable
In-Reply-To: <q2w5cc9e8f31004180037v414fcf97n5db7fccdc657d224@mail.gmail.com>
References: <q2w5cc9e8f31004180037v414fcf97n5db7fccdc657d224@mail.gmail.com>
Message-ID: <i2n5cc9e8f31004181843yd924bb35h8771c9bbd23353ac@mail.gmail.com>

Has anybody else tried accessing hangup causes when placing a call in Lua?


On Sun, Apr 18, 2010 at 3:37 AM, Bekele Martins <bekelemartins at gmail.com>wrote:

> Hi, I want my lua script to dial a phone number and return back the
> disposition code (hangup cause). The problem is if the call fails, the
> session variable is destroyed, and the call disposition code is destroyed as
> well.
>
> For example:
>
> function make_call()
>   sessiondata = "sofia/gateway/myout/15555555555"
>   new_session = freeswitch.Session(sessiondata)
>   if (new_session:ready()) then
>     new_session:streamFile("hello.wav")
>     disposition = "ANSWERED"
>   else
>     disposition = new_session:getVariable("hangup_cause")
>     freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
>     disposition = new_session:hangupCause()
>     freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
>   end
>   return(disposition)
> end
> make_call()
>
>
> This returns the following errors:
>
> 2010-04-18 03:29:58 [NOTICE] switch_channel.c:602 switch_channel_set_name()
> New Channel sofia/external/15555555555
> [4bfbb177-2eee-422d-8f2a-5f9d71082254]
> 2010-04-18 03:30:04 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup
> sofia/external/15555555555 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> API CALL [lua(example.lua)] output:
> -ERR encounterd
>
> 2010-04-18 03:30:04 [ERR] freeswitch_lua.cpp:102 ready() session is not
> initalized
> 2010-04-18 03:30:04 [ERR] switch_cpp.cpp:600 getVariable() session is not
> initalized
> 2010-04-18 03:30:04 [INFO] switch_cpp.cpp:1122 console_log() Hangup cause =
> 2010-04-18 03:30:04 [ERR] mod_lua.cpp:182 lua_parse_and_execute()
> /usr/local/freeswitch/scripts/example.lua:10: attempt to call method
> 'hangupCause' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/example.lua:10: in function
> 'make_call'
>         /usr/local/freeswitch/scripts/example.lua:16: in main chunk
> 2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1085
> switch_core_session_thread() Session 1 (sofia/external/15555555555) Ended
> 2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1087
> switch_core_session_thread() Close Channel sofia/external/15555555555
> [CS_DESTROY]
>
> Is there some way I can accomplish this?
>
>
>
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From dujinfang at gmail.com  Sun Apr 18 19:05:32 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 19 Apr 2010 10:05:32 +0800
Subject: [Freeswitch-users] Lua session disappears after failed call,
	so 	hangup cause becomes inaccessable
In-Reply-To: <i2n5cc9e8f31004181843yd924bb35h8771c9bbd23353ac@mail.gmail.com>
References: <q2w5cc9e8f31004180037v414fcf97n5db7fccdc657d224@mail.gmail.com>
	<i2n5cc9e8f31004181843yd924bb35h8771c9bbd23353ac@mail.gmail.com>
Message-ID: <q2t23f91031004181905pc3b2b4edrcb83c43b987cff41@mail.gmail.com>

the problem is that the session is not initialized

you could try     disposition = new_session:getVariable("originate_disposition")

or see:
http://fisheye.freeswitch.org/browse/~raw,r=1cb080f85f63ac4f666397d64024b8561d62cb75/freeswitch-contrib/seven/lua/dialer.lua


2010/4/19 Bekele Martins <bekelemartins at gmail.com>:
> Has anybody else tried accessing hangup causes when placing a call in Lua?
>
> On Sun, Apr 18, 2010 at 3:37 AM, Bekele Martins <bekelemartins at gmail.com>
> wrote:
>>
>> Hi, I want my lua script to dial a phone number and return back the
>> disposition code (hangup cause). The problem is if the call fails, the
>> session variable is destroyed, and the call disposition code is destroyed as
>> well.
>> For example:
>> function make_call()
>> ??sessiondata = "sofia/gateway/myout/15555555555"
>> ??new_session = freeswitch.Session(sessiondata)
>> ??if (new_session:ready()) then
>> ?? ?new_session:streamFile("hello.wav")
>> ?? ?disposition = "ANSWERED"
>> ??else
>> ?? ?disposition = new_session:getVariable("hangup_cause")
>> ?? ?freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
>> ?? ?disposition = new_session:hangupCause()
>> ?? ?freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
>> ??end
>> ??return(disposition)
>> end
>> make_call()
>>
>> This returns the following errors:
>> 2010-04-18 03:29:58 [NOTICE] switch_channel.c:602
>> switch_channel_set_name() New Channel sofia/external/15555555555
>> [4bfbb177-2eee-422d-8f2a-5f9d71082254]
>> 2010-04-18 03:30:04 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state()
>> Hangup sofia/external/15555555555 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>> API CALL [lua(example.lua)] output:
>> -ERR encounterd
>> 2010-04-18 03:30:04 [ERR] freeswitch_lua.cpp:102 ready() session is not
>> initalized
>> 2010-04-18 03:30:04 [ERR] switch_cpp.cpp:600 getVariable() session is not
>> initalized
>> 2010-04-18 03:30:04 [INFO] switch_cpp.cpp:1122 console_log() Hangup cause
>> =
>> 2010-04-18 03:30:04 [ERR] mod_lua.cpp:182 lua_parse_and_execute()
>> /usr/local/freeswitch/scripts/example.lua:10: attempt to call method
>> 'hangupCause' (a nil value)
>> stack traceback:
>> ?? ? ? ?/usr/local/freeswitch/scripts/example.lua:10: in function
>> 'make_call'
>> ?? ? ? ?/usr/local/freeswitch/scripts/example.lua:16: in main chunk
>> 2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1085
>> switch_core_session_thread() Session 1 (sofia/external/15555555555) Ended
>> 2010-04-18 03:30:04 [NOTICE] switch_core_session.c:1087
>> switch_core_session_thread() Close Channel sofia/external/15555555555
>> [CS_DESTROY]
>> Is there some way I can accomplish 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 cucku.cucku at yahoo.com.vn  Sun Apr 18 21:08:45 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Mon, 19 Apr 2010 12:08:45 +0800 (SGT)
Subject: [Freeswitch-users] need help on IVR
Message-ID: <879420.3757.qm@web76215.mail.sg1.yahoo.com>

Hi all

my network topology:

endpoint 1(100)-----sip server ---IVR(Freeswitch)
|
|
endpoint2(101)

endpoint1 + endpoint2 are registered to sip server
Freeswitch is regsitered to sip server with 103

my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
and RTP from endpoint 1 --> media proxy---> FS
then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101

can FS do it

Thank you

__________________________________________________
B?n C? S? D?ng Yahoo! Kh?ng?
M?t m?i v? th? r?c?  Yahoo! Th? c? ch??ng tr?nh b?o v? ch?ng th? r?c h?u hi?u nh?t tr?n m?ng 
http://vn.mail.yahoo.com 
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From steveayre at gmail.com  Mon Apr 19 01:59:28 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 19 Apr 2010 09:59:28 +0100
Subject: [Freeswitch-users] How to find out a sip extension is busy?
In-Reply-To: <r2td361ca7b1004162255hf56c63aza6e2678065c5950e@mail.gmail.com>
References: <v2m1fdf92721004162230o492905fene3d427bcbf5032e0@mail.gmail.com>
	<r2td361ca7b1004162255hf56c63aza6e2678065c5950e@mail.gmail.com>
Message-ID: <p2jeafa772f1004190159s63bb57f9w102bcd1a7b63a4e5@mail.gmail.com>

You could also use mod_limit

On 17 April 2010 06:55, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> You can query the sip_dialogs table from core DB. We use it to find whether
> an extension is busy and the act upon user's prefference (waiting call, play
> busy or?send to voicemail).
>
> If you need more help please contact me off the list.
>
> ???????????????????? __Yehavi:
>
> 2010/4/17 babak yakhchali <babak.freeswitch at gmail.com>
>>
>> Hi
>> Is there any way to check if a sip extension is busy or not? or find an
>> idle sip extension?
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lists at infosecurity.ch  Mon Apr 19 05:14:07 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 19 Apr 2010 14:14:07 +0200
Subject: [Freeswitch-users] Can FreeSWITCH act as a SIP Proxy with
	transcoding?
Message-ID: <4BCC490F.5010701@infosecurity.ch>

Hi all,

i would like to know if FreeSWITCH could be used as a simple SIP Proxy 
in order to:
- receive SIP/TLS connections
- receive SRTP secured connection
- forward it in "unsecured" way to a backend PBX (plain RTP, plain SIP/UDP)
- if required configured do simply transcoding before forwarding calls 
to backend PBX

So to run it as a "SIP Security Proxy" in front of an existing SIP PBX 
doing also transcoding if required (in case backend SIP PBX does not 
support SIP UA codecs) .

Is this setup feasible with a FreeSWITCH?

Does anyone know if cudatel (https://www.cudatel.com/) support such kind 
of configuration scenario?

Fabio


From sharad at coraltele.com  Mon Apr 19 04:58:14 2010
From: sharad at coraltele.com (sharad)
Date: Mon, 19 Apr 2010 17:28:14 +0530
Subject: [Freeswitch-users] How to know the live calls against a perticuler
	dialplan
Message-ID: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>

Hi All

I just want to know whow to know the number of live calls for a specific dialplan.

For example, if 5000 is a IVR access, so can we know the no. of concurrent calls on 5000 at a moment ?

Regards
Sharad
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From david.ponzone at ipeva.fr  Mon Apr 19 05:25:28 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 19 Apr 2010 14:25:28 +0200
Subject: [Freeswitch-users] How to know the live calls against a
	perticuler dialplan
In-Reply-To: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
Message-ID: <AB37E7C2-D616-4AAE-8D30-76F411508EA4@ipeva.fr>

Sharad,

if the only thing you need to know is the number of calls, I would  
recommend using mod_limit.
mod_limit, if you dont specify a limit, will just update the counter,  
without limiting it.
You can then define all sorts of counters in your dialplan.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 19/04/2010 ? 13:58, sharad a ?crit :

> Hi All
>
> I just want to know whow to know the number of live calls for a  
> specific dialplan.
>
> For example, if 5000 is a IVR access, so can we know the no. of  
> concurrent calls on 5000 at a moment ?
>
> Regards
> Sharad
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Mon Apr 19 05:26:55 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 19 Apr 2010 14:26:55 +0200
Subject: [Freeswitch-users] How to know the live calls against a
	perticuler dialplan
In-Reply-To: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
Message-ID: <EF359775-3C2F-4FC0-9F1E-51DBD2A0197C@gmail.com>

Sharad,

if the only thing you need to know is the number of calls, I would  
recommend using mod_limit.
mod_limit, if you dont specify a limit, will just update the counter,  
without limiting it.
You can then define all sorts of counters in your dialplan.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 19/04/2010 ? 13:58, sharad a ?crit :

> Hi All
>
> I just want to know whow to know the number of live calls for a  
> specific dialplan.
>
> For example, if 5000 is a IVR access, so can we know the no. of  
> concurrent calls on 5000 at a moment ?
>
> Regards
> Sharad
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From justin at ejtown.org  Mon Apr 19 05:46:52 2010
From: justin at ejtown.org (Justin B Newman)
Date: Mon, 19 Apr 2010 08:46:52 -0400
Subject: [Freeswitch-users] How to know the live calls against a
	perticuler dialplan
In-Reply-To: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
Message-ID: <n2q27b0d5bc1004190546qaca26186oe867aeddd5033025@mail.gmail.com>

On Mon, Apr 19, 2010 at 7:58 AM, sharad <sharad at coraltele.com> wrote:

> I just want to know whow to know the number of live calls for a specific
> dialplan.
>
> For example, if 5000 is a IVR access, so can we know the no. of concurrent
> calls on 5000 at a moment ?

In "core.db" lies a table named "channels".  It is your friend.

-jbn


From sharad at coraltele.com  Mon Apr 19 06:11:32 2010
From: sharad at coraltele.com (sharad)
Date: Mon, 19 Apr 2010 18:41:32 +0530
Subject: [Freeswitch-users] How to know the live calls against
	aperticuler dialplan
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
	<AB37E7C2-D616-4AAE-8D30-76F411508EA4@ipeva.fr>
Message-ID: <303270CB3E3149A796E4CA94F9D85942@sharad>

Thanks Mr. David for your quick reply.

I think I made my query wrongly...Let me ask again..

I want to make 5 max outgoing calls from Freeswitch using a dialplan say 5500.

So before using originate API for making the new call, I need to query how many calls are still alive for the dialplan 5500.

If the freeswitch answers the query with the value <5, my application will make another new call else application will wait.

So I want to know is there any way/API to query from freeswitch to find out how many calls are alive for dialplan 5500.

If I use status command, it shows a lot of calls which are alive with other dialplan also. It becomes difficult to find the no. of calls from this heavy stuff.

Thanks & regards
Sharad



----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, April 19, 2010 5:55 PM
  Subject: Re: [Freeswitch-users] How to know the live calls against aperticuler dialplan


  Sharad,


  if the only thing you need to know is the number of calls, I would recommend using mod_limit.
  mod_limit, if you dont specify a limit, will just update the counter, without limiting it.
  You can then define all sorts of counters in your dialplan.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 19/04/2010 ? 13:58, sharad a ?crit :


    Hi All

    I just want to know whow to know the number of live calls for a specific dialplan.

    For example, if 5000 is a IVR access, so can we know the no. of concurrent calls on 5000 at a moment ?

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


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From david.ponzone at gmail.com  Mon Apr 19 06:22:32 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 19 Apr 2010 15:22:32 +0200
Subject: [Freeswitch-users] How to know the live calls against
	aperticuler dialplan
In-Reply-To: <303270CB3E3149A796E4CA94F9D85942@sharad>
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
	<AB37E7C2-D616-4AAE-8D30-76F411508EA4@ipeva.fr>
	<303270CB3E3149A796E4CA94F9D85942@sharad>
Message-ID: <87941034-1988-4DF9-A41A-D5DE3D718360@gmail.com>

mod_limit still seems able to do that.
Though, you will perhaps be able to detect you are over the limit only  
after the call was originated.
To be tested.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 19/04/2010 ? 15:11, sharad a ?crit :

> Thanks Mr. David for your quick reply.
>
> I think I made my query wrongly...Let me ask again..
>
> I want to make 5 max outgoing calls from Freeswitch using a dialplan  
> say 5500.
>
> So before using originate API for making the new call, I need to  
> query how many calls are still alive for the dialplan 5500.
>
> If the freeswitch answers the query with the value <5, my  
> application will make another new call else application will wait.
>
> So I want to know is there any way/API to query from freeswitch to  
> find out how many calls are alive for dialplan 5500.
>
> If I use status command, it shows a lot of calls which are alive  
> with other dialplan also. It becomes difficult to find the no. of  
> calls from this heavy stuff.
>
> Thanks & regards
> Sharad
>
>
>
> ----- Original Message -----
> From: David Ponzone
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, April 19, 2010 5:55 PM
> Subject: Re: [Freeswitch-users] How to know the live calls against  
> aperticuler dialplan
>
> Sharad,
>
> if the only thing you need to know is the number of calls, I would  
> recommend using mod_limit.
> mod_limit, if you dont specify a limit, will just update the  
> counter, without limiting it.
> You can then define all sorts of counters in your dialplan.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 19/04/2010 ? 13:58, sharad a ?crit :
>
>> Hi All
>>
>> I just want to know whow to know the number of live calls for a  
>> specific dialplan.
>>
>> For example, if 5000 is a IVR access, so can we know the no. of  
>> concurrent calls on 5000 at a moment ?
>>
>> Regards
>> Sharad
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From shroukkhan at gmail.com  Sat Apr 17 23:42:36 2010
From: shroukkhan at gmail.com (Shrouk Khan)
Date: Sun, 18 Apr 2010 13:42:36 +0700
Subject: [Freeswitch-users] Billing system
Message-ID: <m2raea1d7231004172342z6b03b2dbh1bd4703f582a1542@mail.gmail.com>

hi all,
i am sure this question has been asked a million times , but i did not find
any satisfactory answers :) so here goes again :

what is the most preferred billing system to be used with freeswitch
(preferably something that is capable of leveraging the benefits of
mod_nibblebill )

ideas ?
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From mattmill30 at hotmail.com  Sun Apr 18 16:53:04 2010
From: mattmill30 at hotmail.com (Matthew Millar)
Date: Mon, 19 Apr 2010 00:53:04 +0100
Subject: [Freeswitch-users] Is it possible to stream audio to multiple
 people down the same line?
Message-ID: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>


Hi,

I've been looking to setup a telephone system at my church, 
as theres alot of elderly people who want to listen to the sermons, but 
can't make it due to ailments.

I thought the best idea would be 
to setup a phone system where they can dial into something like a one-way
 conference system.

I was wondering if someone could point me in the right direction?

Is it possible to stream audio to multiple people down the same line?

I hope that makes sense.

Sorry if this is a really stupid question, but i'm used to IP networks, where you can just setup a multicasting server.

Thanks for any help given,

Matthew
 		 	   		  
_________________________________________________________________
http://clk.atdmt.com/UKM/go/197222280/direct/01/
Do you have a story that started on Hotmail? Tell us now
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From jerome.meuret at gmail.com  Mon Apr 19 06:20:44 2010
From: jerome.meuret at gmail.com (=?ISO-8859-1?B?Suly9G1lIE0u?=)
Date: Mon, 19 Apr 2010 15:20:44 +0200
Subject: [Freeswitch-users] Users : Alphanumeric mapping
Message-ID: <w2o9d8570731004190620z2484c46ci3105d044f8481da0@mail.gmail.com>

Hi everyone,

I'm new to Freeswitch and just started to configure it.

I have read the wiki, and my first problem is the following :

- I want to use alphanumeric id for my users. In other words, they can
register to the PBX with a id like firstname.lastname at domain.com and can be
reached in this way, no need to know the phone number (300-399 in my case).

So, i declared my users as mentionned on the wiki, using number-alias. In my
dialplan, every call to 300-399 numbers are automatically associated to
alphanumeric id's, everything is working fine.

But in a real context, every user will only know the alphanumeric-id (it's
easier to remember) of another users. I thought that the mapping was
bidirectionnal, but it seems that I have to match every
firstname.lastname at domain.com in my dialplan.

I tried, but apparently destination_number contains only firstname.lastname
( no SIP URI, maybe because I use the local domain).

I' m sure I could do it using a "dirty trick" but i really want to do it in
a good way...

Do you have any ideas ?

Thank you so much for your help, sorry for my approximative english ;) !

Kind regards,
Jerome
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From sharad at coraltele.com  Mon Apr 19 06:43:16 2010
From: sharad at coraltele.com (sharad)
Date: Mon, 19 Apr 2010 19:13:16 +0530
Subject: [Freeswitch-users] How to know the live calls against
	aperticuler dialplan
References: <7A80A84DCBC64B3F9CBA77D48D6C16F8@sharad>
	<n2q27b0d5bc1004190546qaca26186oe867aeddd5033025@mail.gmail.com>
Message-ID: <31AE4BC3ABE241E2AE8545B69E99454C@sharad>

Thanks a lot Mr. Justin.

I got it...thanx once again.

Regards
Sharad


----- Original Message ----- 
From: "Justin B Newman" <justin at ejtown.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, April 19, 2010 6:16 PM
Subject: Re: [Freeswitch-users] How to know the live calls against 
aperticuler dialplan


> On Mon, Apr 19, 2010 at 7:58 AM, sharad <sharad at coraltele.com> wrote:
>
>> I just want to know whow to know the number of live calls for a specific
>> dialplan.
>>
>> For example, if 5000 is a IVR access, so can we know the no. of 
>> concurrent
>> calls on 5000 at a moment ?
>
> In "core.db" lies a table named "channels".  It is your friend.
>
> -jbn
>
>
> 



From pjintheusa at gmail.com  Mon Apr 19 07:01:14 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 19 Apr 2010 10:01:14 -0400
Subject: [Freeswitch-users] Is it possible to stream audio to multiple
	people down the same line?
In-Reply-To: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>
References: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>
Message-ID: <w2r367751821004190701gd297fb7dh381d6309077ab9c0@mail.gmail.com>

Not sure what you mean by "down the same line"

But if people can dial in, then you could just set up a conference as you
suggest.

See here:

http://wiki.freeswitch.org/wiki/Mod_conference



On Sun, Apr 18, 2010 at 7:53 PM, Matthew Millar <mattmill30 at hotmail.com>wrote:

>  Hi,
>
> I've been looking to setup a telephone system at my church, as theres alot
> of elderly people who want to listen to the sermons, but can't make it due
> to ailments.
>
> I thought the best idea would be to setup a phone system where they can
> dial into something like a one-way conference system.
>
> I was wondering if someone could point me in the right direction?
>
> Is it possible to stream audio to multiple people down the same line?
>
> I hope that makes sense.
>
> Sorry if this is a really stupid question, but i'm used to IP networks,
> where you can just setup a multicasting server.
>
> Thanks for any help given,
>
> Matthew
>
> ------------------------------
> Get a new e-mail account with Hotmail - Free. Sign-up now.<http://clk.atdmt.com/UKM/go/197222280/direct/01/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From daniel.neubert at solomo.de  Mon Apr 19 06:43:26 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Mon, 19 Apr 2010 15:43:26 +0200
Subject: [Freeswitch-users] Is it possible to stream audio to multiple
 people down the same line?
In-Reply-To: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>
References: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>
Message-ID: <4BCC5DFE.50407@solomo.de>

Hi Matthew,

what about setting up an icecast server and use mod_shout for listening 
to that stream?

Advantage: You can use the stream for other distribution channels as 
well (ie. on a website).

Best regards / Mit freundlichen Gr??en,
Daniel Neubert



On 19.04.2010 01:53, Matthew Millar wrote:
> Hi,
>
> I've been looking to setup a telephone system at my church, as theres 
> alot of elderly people who want to listen to the sermons, but can't 
> make it due to ailments.
>
> I thought the best idea would be to setup a phone system where they 
> can dial into something like a one-way conference system.
>
> I was wondering if someone could point me in the right direction?
>
> Is it possible to stream audio to multiple people down the same line?
>
> I hope that makes sense.
>
> Sorry if this is a really stupid question, but i'm used to IP 
> networks, where you can just setup a multicasting server.
>
> Thanks for any help given,
>
> Matthew
>
> ------------------------------------------------------------------------
> Get a new e-mail account with Hotmail - Free. Sign-up now. 
> <http://clk.atdmt.com/UKM/go/197222280/direct/01/>
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From jeremy at seadragons.us  Mon Apr 19 07:12:32 2010
From: jeremy at seadragons.us (Jeremy Shaffner)
Date: Mon, 19 Apr 2010 10:12:32 -0400
Subject: [Freeswitch-users] Is it possible to stream audio to multiple
	people down the same line?
In-Reply-To: <w2r367751821004190701gd297fb7dh381d6309077ab9c0@mail.gmail.com>
References: <SNT116-W53E2D3E95EF55502A0C152C40C0@phx.gbl>
	<w2r367751821004190701gd297fb7dh381d6309077ab9c0@mail.gmail.com>
Message-ID: <3BFD68D1-2339-47BC-9851-D6100AAC956C@seadragons.us>

And you would set all joining callers up so they cannot speak into the conference.


On Apr 19, 2010, at 10:01 AM, Phillip Jones wrote:

> Not sure what you mean by "down the same line"
> 
> But if people can dial in, then you could just set up a conference as you suggest. 
> 
> See here: 
> 
> http://wiki.freeswitch.org/wiki/Mod_conference
> 
> 
> 
> On Sun, Apr 18, 2010 at 7:53 PM, Matthew Millar <mattmill30 at hotmail.com> wrote:
> Hi,
> 
> I've been looking to setup a telephone system at my church, as theres alot of elderly people who want to listen to the sermons, but can't make it due to ailments.
> 
> I thought the best idea would be to setup a phone system where they can dial into something like a one-way conference system.
> 
> I was wondering if someone could point me in the right direction?
> 
> Is it possible to stream audio to multiple people down the same line?
> 
> I hope that makes sense.
> 
> Sorry if this is a really stupid question, but i'm used to IP networks, where you can just setup a multicasting server.
> 
> Thanks for any help given,
> 
> Matthew
> 
> Get a new e-mail account with Hotmail - Free. Sign-up 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  Mon Apr 19 07:49:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 09:49:45 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004181248.33175.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<p2z191c3a031004180935g23a12e71n58db03227e5133bb@mail.gmail.com>
	<w2t191c3a031004180935x41838351ma2d0f08cf6b99ebc@mail.gmail.com>
	<201004181248.33175.sos@sokhapkin.dyndns.org>
Message-ID: <g2n191c3a031004190749p208f98fdi6771b22af96e9a29@mail.gmail.com>

300M is tiny. try having 2000 channels up you can soar into a gig of usage.

That valgrind report only showed 100k leaked which means it was all
accounted for and torn down in the end.

If you can find a specific leak in the nibble bill module with valgrind we
will gladly fix it and you
can gladly thank us for giving you a way to make money by the second.

valgrind --tool=memcheck --log-file=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes
/usr/local/freeswitch/bin/freeswitch -vg



On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Which valgrind options should I specify? I did run
>
> valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-
> resolution=high --show-reachable=yes ./freeswitch -nonat -vg
>
> 100K leaked is nothing compared to FS process RSS of 300M...
>
> On Sunday 18 April 2010, Anthony Minessale wrote:
> > Also thatb suggests you leaked a whopping 100k
> >
> > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> anthony.minessale at gmail.com>
> > wrote:
> >
> > That's not very useful you need a full report with extended checking.
> >
> > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> >
> > wrote:
> > > According to v...
> > >
> > >
> > >
> > >
> > >
> > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > That is a completely incorrect statement. Usi...
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ivdreg at gmail.com  Mon Apr 19 08:13:25 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 19 Apr 2010 18:13:25 +0300
Subject: [Freeswitch-users] looking for someone who working on mod_redis
In-Reply-To: <DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>
References: <n2j8ccbff061004172147qf6ebfabcgacef84e2f910c7b9@mail.gmail.com>
	<DB714C37-593E-4D03-A4FB-8BF3FF1F051A@jerris.com>
Message-ID: <w2kb437a0e1004190813n76940b77y13ebc17368aac5b@mail.gmail.com>

I use MongoDB with perl app from dialplan and also for raw data storage for
FS CDRs. MongoDB is something between key/value databases and SQL.  It is
schema free and its very useful if you want DB that you can insert different
types of data. In one of my cases I use Perl module XMLSimple to parse CDR
and write directly obtained Perl hash from XML into MongoDB - 10 lines app
;)

2010/4/18 Michael Jerris <mike at jerris.com>

> I spent some time looking at credis, but I think something better is
> probably needed, it may work, but its not architected with a lot of thought
> to performing as well as it could or working well in general.  Also, it only
> supports fairly old redis.  A better approach is probably building a stand
> alone lib in the redis tree itself.  the redis author says he has one or the
> next major version that will be checked in at some point.
>
> Mike
>
> On Apr 18, 2010, at 12:47 AM, Dome Charoenyost wrote:
>
> > Dear Sir,
> >              Now i'm using tokyo tyrant with my application by
> > mod_memcache. i have plan to move to redis but redis not complatible
> > with memcache.
> > So if someone working on mod_redis please let's me know.
> >
> > BG
> >
> > Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Mon Apr 19 08:29:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 11:29:12 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <g2n191c3a031004190749p208f98fdi6771b22af96e9a29@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004181248.33175.sos@sokhapkin.dyndns.org>
	<g2n191c3a031004190749p208f98fdi6771b22af96e9a29@mail.gmail.com>
Message-ID: <201004191129.12109.sos@sokhapkin.dyndns.org>

Valgrind doesn't show leaks in mod_nibblebill, most of the leaked memory was 
in mod_sofia:

==12476== 53,064 bytes in 67 blocks are definitely lost in loss record 138 of 
141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
==12476==
==12476==
==12476== 92,664 bytes in 117 blocks are definitely lost in loss record 139 of 
141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
==12476==

BTW, do I understand FS code correct that channel private data are always 
handled by sqlite, even if I enable core ODBC?


On Monday 19 April 2010, Anthony Minessale wrote:
> 300M is tiny. try having 2000 channels up you can soar into a gig of usage.
> 
> That valgrind report only showed 100k leaked which means it was all
> accounted for and torn down in the end.
> 
> If you can find a specific leak in the nibble bill module with valgrind we
> will gladly fix it and you
> can gladly thank us for giving you a way to make money by the second.
> 
> valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> --leak-resolution=high --show-reachable=yes
> /usr/local/freeswitch/bin/freeswitch -vg
> 
> 
> 
> On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Which valgrind options should I specify? I did run
> >
> > valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-
> > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> >
> > 100K leaked is nothing compared to FS process RSS of 300M...
> >
> > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > Also thatb suggests you leaked a whopping 100k
> > >
> > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> >
> > anthony.minessale at gmail.com>
> >
> > > wrote:
> > >
> > > That's not very useful you need a full report with extended checking.
> > >
> > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> > >
> > > wrote:
> > > > According to v...
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > That is a completely incorrect statement. Usi...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.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 Apr 19 08:41:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 10:41:12 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191129.12109.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004181248.33175.sos@sokhapkin.dyndns.org>
	<g2n191c3a031004190749p208f98fdi6771b22af96e9a29@mail.gmail.com>
	<201004191129.12109.sos@sokhapkin.dyndns.org>
Message-ID: <z2q191c3a031004190841y3821603u42a96c996358357a@mail.gmail.com>

This is invalid.

Those places are where sip calls are created and they are most definitely
not leaking.
If anything, it suggests dialogs that are up still and not yet destroyed at
the termination of the program.
If this location in the code was a leak it would be gigs not megs missing.

Did you do the exact command I said (especially -vg param to FS) and then a
full clean shutdown all the way back to the shell?

Are you maybe not getting the BYE to your calls creating open dialogs?
You may want to turn on the sip trace.


On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Valgrind doesn't show leaks in mod_nibblebill, most of the leaked memory
> was
> in mod_sofia:
>
> ==12476== 53,064 bytes in 67 blocks are definitely lost in loss record 138
> of
> 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
> ==12476==
> ==12476==
> ==12476== 92,664 bytes in 117 blocks are definitely lost in loss record 139
> of
> 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
> ==12476==
>
> BTW, do I understand FS code correct that channel private data are always
> handled by sqlite, even if I enable core ODBC?
>
>
> On Monday 19 April 2010, Anthony Minessale wrote:
> > 300M is tiny. try having 2000 channels up you can soar into a gig of
> usage.
> >
> > That valgrind report only showed 100k leaked which means it was all
> > accounted for and torn down in the end.
> >
> > If you can find a specific leak in the nibble bill module with valgrind
> we
> > will gladly fix it and you
> > can gladly thank us for giving you a way to make money by the second.
> >
> > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > --leak-resolution=high --show-reachable=yes
> > /usr/local/freeswitch/bin/freeswitch -vg
> >
> >
> >
> > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Which valgrind options should I specify? I did run
> > >
> > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-
> > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > >
> > > 100K leaked is nothing compared to FS process RSS of 300M...
> > >
> > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > Also thatb suggests you leaked a whopping 100k
> > > >
> > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > >
> > > anthony.minessale at gmail.com>
> > >
> > > > wrote:
> > > >
> > > > That's not very useful you need a full report with extended checking.
> > > >
> > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> sos at sokhapkin.dyndns.org>
> > > >
> > > > wrote:
> > > > > According to v...
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > That is a completely incorrect statement. Usi...
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Mon Apr 19 09:03:05 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 12:03:05 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <z2q191c3a031004190841y3821603u42a96c996358357a@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191129.12109.sos@sokhapkin.dyndns.org>
	<z2q191c3a031004190841y3821603u42a96c996358357a@mail.gmail.com>
Message-ID: <201004191203.05924.sos@sokhapkin.dyndns.org>

The command was exactly as you suggested, FS was started with "-nonat -vg" 
command line options, after few hours run (about 5K calls processed) FS was 
shut down gracefully.

On Monday 19 April 2010, Anthony Minessale wrote:
> This is invalid.
> 
> Those places are where sip calls are created and they are most definitely
> not leaking.
> If anything, it suggests dialogs that are up still and not yet destroyed at
> the termination of the program.
> If this location in the code was a leak it would be gigs not megs missing.
> 
> Did you do the exact command I said (especially -vg param to FS) and then a
> full clean shutdown all the way back to the shell?
> 
> Are you maybe not getting the BYE to your calls creating open dialogs?
> You may want to turn on the sip trace.
> 
> 
> On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Valgrind doesn't show leaks in mod_nibblebill, most of the leaked memory
> > was
> > in mod_sofia:
> >
> > ==12476== 53,064 bytes in 67 blocks are definitely lost in loss record
> > 138 of
> > 141
> > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
> > ==12476==
> > ==12476==
> > ==12476== 92,664 bytes in 117 blocks are definitely lost in loss record
> > 139 of
> > 141
> > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
> > ==12476==
> >
> > BTW, do I understand FS code correct that channel private data are always
> > handled by sqlite, even if I enable core ODBC?
> >
> > On Monday 19 April 2010, Anthony Minessale wrote:
> > > 300M is tiny. try having 2000 channels up you can soar into a gig of
> >
> > usage.
> >
> > > That valgrind report only showed 100k leaked which means it was all
> > > accounted for and torn down in the end.
> > >
> > > If you can find a specific leak in the nibble bill module with valgrind
> >
> > we
> >
> > > will gladly fix it and you
> > > can gladly thank us for giving you a way to make money by the second.
> > >
> > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > --leak-resolution=high --show-reachable=yes
> > > /usr/local/freeswitch/bin/freeswitch -vg
> > >
> > >
> > >
> > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Which valgrind options should I specify? I did run
> > > >
> > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-
> > > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > > >
> > > > 100K leaked is nothing compared to FS process RSS of 300M...
> > > >
> > > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > > Also thatb suggests you leaked a whopping 100k
> > > > >
> > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > > >
> > > > anthony.minessale at gmail.com>
> > > >
> > > > > wrote:
> > > > >
> > > > > That's not very useful you need a full report with extended
> > > > > checking.
> > > > >
> > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> >
> > sos at sokhapkin.dyndns.org>
> >
> > > > > wrote:
> > > > > > According to v...
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > > That is a completely incorrect statement. Usi...
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From erkan at speedingtrade.com  Mon Apr 19 09:15:05 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Mon, 19 Apr 2010 19:15:05 +0300
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>

Hi FS Guys,

 

I installed FS on Windows and try to use the xml_cdr.conf.xml to post the calling information's to a website/script.

 

I change my xml_cdr.conf.xml as give on the website

http://wiki.freeswitch.org/wiki/Mod_xml_cdr

 

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
<settings>
   <param name="url" value="http://localhost/callreg.aspx"/>
   <param name="retries" value="2"/>
   <param name="delay" value="120"/>
   <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
   <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
   <param name="encode" value="True"/>
</settings>

 

I make a dummy aspx that write all given parameters to a file.

But the aspx don't called.

 

What I must do to get the information's into the http://localhost/callreg.aspx

 

My dummy read the information's from the http://localhost/callreg.aspx?xxx=yyy and so on

The dummy aspx read all parameters that given after the ?

But no information's coming.

 

Can anyone help me?

 

Greetings

Erkan

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From sos at sokhapkin.dyndns.org  Mon Apr 19 09:25:41 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 12:25:41 -0400
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
Message-ID: <201004191225.41988.sos@sokhapkin.dyndns.org>

Did you load the module?

On Monday 19 April 2010, Erkan ?nl? wrote:
> Hi FS Guys,
> 
> 
> 
> I installed FS on Windows and try to use the xml_cdr.conf.xml to post the
>  calling information's to a website/script.
> 
> 
> 
> I change my xml_cdr.conf.xml as give on the website
> 
> http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> 
> 
> 
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> <settings>
>    <param name="url" value="http://localhost/callreg.aspx"/>
>    <param name="retries" value="2"/>
>    <param name="delay" value="120"/>
>    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
>    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
>    <param name="encode" value="True"/>
> </settings>
> 
> 
> 
> I make a dummy aspx that write all given parameters to a file.
> 
> But the aspx don't called.
> 
> 
> 
> What I must do to get the information's into the
>  http://localhost/callreg.aspx
> 
> 
> 
> My dummy read the information's from the
>  http://localhost/callreg.aspx?xxx=yyy and so on
> 
> The dummy aspx read all parameters that given after the ?
> 
> But no information's coming.
> 
> 
> 
> Can anyone help me?
> 
> 
> 
> Greetings
> 
> Erkan
> 



From anthony.minessale at gmail.com  Mon Apr 19 09:31:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 11:31:52 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191203.05924.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191129.12109.sos@sokhapkin.dyndns.org>
	<z2q191c3a031004190841y3821603u42a96c996358357a@mail.gmail.com>
	<201004191203.05924.sos@sokhapkin.dyndns.org>
Message-ID: <w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>

welll you are losing sip dialogs somewhere.
It's not a leak its loss of sip dialogs most likely a side effect of
topology problems.


On Mon, Apr 19, 2010 at 11:03 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> The command was exactly as you suggested, FS was started with "-nonat -vg"
> command line options, after few hours run (about 5K calls processed) FS was
> shut down gracefully.
>
> On Monday 19 April 2010, Anthony Minessale wrote:
> > This is invalid.
> >
> > Those places are where sip calls are created and they are most definitely
> > not leaking.
> > If anything, it suggests dialogs that are up still and not yet destroyed
> at
> > the termination of the program.
> > If this location in the code was a leak it would be gigs not megs
> missing.
> >
> > Did you do the exact command I said (especially -vg param to FS) and then
> a
> > full clean shutdown all the way back to the shell?
> >
> > Are you maybe not getting the BYE to your calls creating open dialogs?
> > You may want to turn on the sip trace.
> >
> >
> > On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Valgrind doesn't show leaks in mod_nibblebill, most of the leaked
> memory
> > > was
> > > in mod_sofia:
> > >
> > > ==12476== 53,064 bytes in 67 blocks are definitely lost in loss record
> > > 138 of
> > > 141
> > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > linux/vgpreload_memcheck.so)
> > > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
> > > ==12476==
> > > ==12476==
> > > ==12476== 92,664 bytes in 117 blocks are definitely lost in loss record
> > > 139 of
> > > 141
> > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > linux/vgpreload_memcheck.so)
> > > ==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
> > > ==12476==
> > >
> > > BTW, do I understand FS code correct that channel private data are
> always
> > > handled by sqlite, even if I enable core ODBC?
> > >
> > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > 300M is tiny. try having 2000 channels up you can soar into a gig of
> > >
> > > usage.
> > >
> > > > That valgrind report only showed 100k leaked which means it was all
> > > > accounted for and torn down in the end.
> > > >
> > > > If you can find a specific leak in the nibble bill module with
> valgrind
> > >
> > > we
> > >
> > > > will gladly fix it and you
> > > > can gladly thank us for giving you a way to make money by the second.
> > > >
> > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > > --leak-resolution=high --show-reachable=yes
> > > > /usr/local/freeswitch/bin/freeswitch -vg
> > > >
> > > >
> > > >
> > > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> > > >
> > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > Which valgrind options should I specify? I did run
> > > > >
> > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> --leak-
> > > > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > > > >
> > > > > 100K leaked is nothing compared to FS process RSS of 300M...
> > > > >
> > > > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > > > Also thatb suggests you leaked a whopping 100k
> > > > > >
> > > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > > > >
> > > > > anthony.minessale at gmail.com>
> > > > >
> > > > > > wrote:
> > > > > >
> > > > > > That's not very useful you need a full report with extended
> > > > > > checking.
> > > > > >
> > > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> > >
> > > sos at sokhapkin.dyndns.org>
> > >
> > > > > > wrote:
> > > > > > > According to v...
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > > > That is a completely incorrect statement. Usi...
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From erkan at speedingtrade.com  Mon Apr 19 09:36:05 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Mon, 19 Apr 2010 19:36:05 +0300
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
	<201004191225.41988.sos@sokhapkin.dyndns.org>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>

I think yes, because the freeswitch.xml have the following lines inside.

  <section name="configuration" description="Various Configuration">
    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml" />
  </section>

Maybe I must activate it on another place?



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin
Sent: Monday, April 19, 2010 7:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call

Did you load the module?

On Monday 19 April 2010, Erkan ?nl? wrote:
> Hi FS Guys,
> 
> 
> 
> I installed FS on Windows and try to use the xml_cdr.conf.xml to post the
>  calling information's to a website/script.
> 
> 
> 
> I change my xml_cdr.conf.xml as give on the website
> 
> http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> 
> 
> 
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> <settings>
>    <param name="url" value="http://localhost/callreg.aspx"/>
>    <param name="retries" value="2"/>
>    <param name="delay" value="120"/>
>    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
>    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
>    <param name="encode" value="True"/>
> </settings>
> 
> 
> 
> I make a dummy aspx that write all given parameters to a file.
> 
> But the aspx don't called.
> 
> 
> 
> What I must do to get the information's into the
>  http://localhost/callreg.aspx
> 
> 
> 
> My dummy read the information's from the
>  http://localhost/callreg.aspx?xxx=yyy and so on
> 
> The dummy aspx read all parameters that given after the ?
> 
> But no information's coming.
> 
> 
> 
> Can anyone help me?
> 
> 
> 
> Greetings
> 
> Erkan
> 


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


From sos at sokhapkin.dyndns.org  Mon Apr 19 09:42:23 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 12:42:23 -0400
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
	<201004191225.41988.sos@sokhapkin.dyndns.org>
	<81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>
Message-ID: <201004191242.23056.sos@sokhapkin.dyndns.org>

You should enable mod_xml_cdr in modules.conf.xml

On Monday 19 April 2010, Erkan ?nl? wrote:
> I think yes, because the freeswitch.xml have the following lines inside.
> 
>   <section name="configuration" description="Various Configuration">
>     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml" />
>   </section>
> 
> Maybe I must activate it on another place?
> 
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
>  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
>  Okhapkin Sent: Monday, April 19, 2010 7:26 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
> 
> Did you load the module?
> 
> On Monday 19 April 2010, Erkan ?nl? wrote:
> > Hi FS Guys,
> >
> >
> >
> > I installed FS on Windows and try to use the xml_cdr.conf.xml to post the
> >  calling information's to a website/script.
> >
> >
> >
> > I change my xml_cdr.conf.xml as give on the website
> >
> > http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> >
> >
> >
> > <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> > <settings>
> >    <param name="url" value="http://localhost/callreg.aspx"/>
> >    <param name="retries" value="2"/>
> >    <param name="delay" value="120"/>
> >    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
> >    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
> >    <param name="encode" value="True"/>
> > </settings>
> >
> >
> >
> > I make a dummy aspx that write all given parameters to a file.
> >
> > But the aspx don't called.
> >
> >
> >
> > What I must do to get the information's into the
> >  http://localhost/callreg.aspx
> >
> >
> >
> > My dummy read the information's from the
> >  http://localhost/callreg.aspx?xxx=yyy and so on
> >
> > The dummy aspx read all parameters that given after the ?
> >
> > But no information's coming.
> >
> >
> >
> > Can anyone help me?
> >
> >
> >
> > Greetings
> >
> > Erkan
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From sos at sokhapkin.dyndns.org  Mon Apr 19 09:45:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 12:45:06 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191203.05924.sos@sokhapkin.dyndns.org>
	<w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>
Message-ID: <201004191245.06964.sos@sokhapkin.dyndns.org>

It's possible that some BYE or ACK could be lost, but shouldn't sip dialogs be 
destroyed automatically after a timeout?

On Monday 19 April 2010, Anthony Minessale wrote:
> welll you are losing sip dialogs somewhere.
> It's not a leak its loss of sip dialogs most likely a side effect of
> topology problems.
> 
> 
> On Mon, Apr 19, 2010 at 11:03 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > The command was exactly as you suggested, FS was started with "-nonat
> > -vg" command line options, after few hours run (about 5K calls processed)
> > FS was shut down gracefully.
> >
> > On Monday 19 April 2010, Anthony Minessale wrote:
> > > This is invalid.
> > >
> > > Those places are where sip calls are created and they are most
> > > definitely not leaking.
> > > If anything, it suggests dialogs that are up still and not yet
> > > destroyed
> >
> > at
> >
> > > the termination of the program.
> > > If this location in the code was a leak it would be gigs not megs
> >
> > missing.
> >
> > > Did you do the exact command I said (especially -vg param to FS) and
> > > then
> >
> > a
> >
> > > full clean shutdown all the way back to the shell?
> > >
> > > Are you maybe not getting the BYE to your calls creating open dialogs?
> > > You may want to turn on the sip trace.
> > >
> > >
> > > On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Valgrind doesn't show leaks in mod_nibblebill, most of the leaked
> >
> > memory
> >
> > > > was
> > > > in mod_sofia:
> > > >
> > > > ==12476== 53,064 bytes in 67 blocks are definitely lost in loss
> > > > record 138 of
> > > > 141
> > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > linux/vgpreload_memcheck.so)
> > > > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
> > > > ==12476==
> > > > ==12476==
> > > > ==12476== 92,664 bytes in 117 blocks are definitely lost in loss
> > > > record 139 of
> > > > 141
> > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > linux/vgpreload_memcheck.so)
> > > > ==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
> > > > ==12476==
> > > >
> > > > BTW, do I understand FS code correct that channel private data are
> >
> > always
> >
> > > > handled by sqlite, even if I enable core ODBC?
> > > >
> > > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > > 300M is tiny. try having 2000 channels up you can soar into a gig
> > > > > of
> > > >
> > > > usage.
> > > >
> > > > > That valgrind report only showed 100k leaked which means it was all
> > > > > accounted for and torn down in the end.
> > > > >
> > > > > If you can find a specific leak in the nibble bill module with
> >
> > valgrind
> >
> > > > we
> > > >
> > > > > will gladly fix it and you
> > > > > can gladly thank us for giving you a way to make money by the
> > > > > second.
> > > > >
> > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > > > --leak-resolution=high --show-reachable=yes
> > > > > /usr/local/freeswitch/bin/freeswitch -vg
> > > > >
> > > > >
> > > > >
> > > > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> > > > >
> > > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > > Which valgrind options should I specify? I did run
> > > > > >
> > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> >
> > --leak-
> >
> > > > > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > > > > >
> > > > > > 100K leaked is nothing compared to FS process RSS of 300M...
> > > > > >
> > > > > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > > > > Also thatb suggests you leaked a whopping 100k
> > > > > > >
> > > > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > > > > >
> > > > > > anthony.minessale at gmail.com>
> > > > > >
> > > > > > > wrote:
> > > > > > >
> > > > > > > That's not very useful you need a full report with extended
> > > > > > > checking.
> > > > > > >
> > > > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> > > >
> > > > sos at sokhapkin.dyndns.org>
> > > >
> > > > > > > wrote:
> > > > > > > > According to v...
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > > > > That is a completely incorrect statement. Usi...
> > > > > >
> > > > > > _______________________________________________
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > UNSUBSCRIBE:
> > > >
> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >
> > > > > > http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.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 Apr 19 09:57:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 11:57:40 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191245.06964.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191203.05924.sos@sokhapkin.dyndns.org>
	<w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>
	<201004191245.06964.sos@sokhapkin.dyndns.org>
Message-ID: <t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>

yes its a function of the sofia-sip project, but did you wait before you
stopped it?
We have people sending non stop 200+calls a second complaining about no
leaks.
So if you have some real problem it's going to be your network conditions
and interop with your
providers causing an edge case with the SIP stack itself who has it's own
list and irc channel.



On Mon, Apr 19, 2010 at 11:45 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> It's possible that some BYE or ACK could be lost, but shouldn't sip dialogs
> be
> destroyed automatically after a timeout?
>
> On Monday 19 April 2010, Anthony Minessale wrote:
> > welll you are losing sip dialogs somewhere.
> > It's not a leak its loss of sip dialogs most likely a side effect of
> > topology problems.
> >
> >
> > On Mon, Apr 19, 2010 at 11:03 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > The command was exactly as you suggested, FS was started with "-nonat
> > > -vg" command line options, after few hours run (about 5K calls
> processed)
> > > FS was shut down gracefully.
> > >
> > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > This is invalid.
> > > >
> > > > Those places are where sip calls are created and they are most
> > > > definitely not leaking.
> > > > If anything, it suggests dialogs that are up still and not yet
> > > > destroyed
> > >
> > > at
> > >
> > > > the termination of the program.
> > > > If this location in the code was a leak it would be gigs not megs
> > >
> > > missing.
> > >
> > > > Did you do the exact command I said (especially -vg param to FS) and
> > > > then
> > >
> > > a
> > >
> > > > full clean shutdown all the way back to the shell?
> > > >
> > > > Are you maybe not getting the BYE to your calls creating open
> dialogs?
> > > > You may want to turn on the sip trace.
> > > >
> > > >
> > > > On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
> > > >
> > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > Valgrind doesn't show leaks in mod_nibblebill, most of the leaked
> > >
> > > memory
> > >
> > > > > was
> > > > > in mod_sofia:
> > > > >
> > > > > ==12476== 53,064 bytes in 67 blocks are definitely lost in loss
> > > > > record 138 of
> > > > > 141
> > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > > linux/vgpreload_memcheck.so)
> > > > > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite (sofia.c:6538)
> > > > > ==12476==
> > > > > ==12476==
> > > > > ==12476== 92,664 bytes in 117 blocks are definitely lost in loss
> > > > > record 139 of
> > > > > 141
> > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > > linux/vgpreload_memcheck.so)
> > > > > ==12476==    by 0x6A2F76F: sofia_glue_do_invite (sofia_glue.c:1874)
> > > > > ==12476==
> > > > >
> > > > > BTW, do I understand FS code correct that channel private data are
> > >
> > > always
> > >
> > > > > handled by sqlite, even if I enable core ODBC?
> > > > >
> > > > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > > > 300M is tiny. try having 2000 channels up you can soar into a gig
> > > > > > of
> > > > >
> > > > > usage.
> > > > >
> > > > > > That valgrind report only showed 100k leaked which means it was
> all
> > > > > > accounted for and torn down in the end.
> > > > > >
> > > > > > If you can find a specific leak in the nibble bill module with
> > >
> > > valgrind
> > >
> > > > > we
> > > > >
> > > > > > will gladly fix it and you
> > > > > > can gladly thank us for giving you a way to make money by the
> > > > > > second.
> > > > > >
> > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > > > > --leak-resolution=high --show-reachable=yes
> > > > > > /usr/local/freeswitch/bin/freeswitch -vg
> > > > > >
> > > > > >
> > > > > >
> > > > > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> > > > > >
> > > > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > > > Which valgrind options should I specify? I did run
> > > > > > >
> > > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > >
> > > --leak-
> > >
> > > > > > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > > > > > >
> > > > > > > 100K leaked is nothing compared to FS process RSS of 300M...
> > > > > > >
> > > > > > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > > > > > Also thatb suggests you leaked a whopping 100k
> > > > > > > >
> > > > > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > > > > > >
> > > > > > > anthony.minessale at gmail.com>
> > > > > > >
> > > > > > > > wrote:
> > > > > > > >
> > > > > > > > That's not very useful you need a full report with extended
> > > > > > > > checking.
> > > > > > > >
> > > > > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> > > > >
> > > > > sos at sokhapkin.dyndns.org>
> > > > >
> > > > > > > > wrote:
> > > > > > > > > According to v...
> > > > > > > > >
> > > > > > > > >
> > > > > > > > >
> > > > > > > > >
> > > > > > > > >
> > > > > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > > > > > That is a completely incorrect statement. Usi...
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > FreeSWITCH-users mailing list
> > > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > > UNSUBSCRIBE:
> > > > >
> > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > >
> > > > > > > http://www.freeswitch.org
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
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From mardy at voysys.com  Mon Apr 19 07:27:18 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Mon, 19 Apr 2010 10:27:18 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when FS
	is the transfer target
Message-ID: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>

I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?

-Mardy



From sos at sokhapkin.dyndns.org  Mon Apr 19 10:06:28 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 13:06:28 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191245.06964.sos@sokhapkin.dyndns.org>
	<t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
Message-ID: <201004191306.28787.sos@sokhapkin.dyndns.org>

Yes, I stopped the traffic to the server and issued hupall CLI command before 
shutting down FS.

On Monday 19 April 2010, Anthony Minessale wrote:
> yes its a function of the sofia-sip project, but did you wait before you
> stopped it?
> We have people sending non stop 200+calls a second complaining about no
> leaks.
> So if you have some real problem it's going to be your network conditions
> and interop with your
> providers causing an edge case with the SIP stack itself who has it's own
> list and irc channel.
> 
> 
> 
> On Mon, Apr 19, 2010 at 11:45 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > It's possible that some BYE or ACK could be lost, but shouldn't sip
> > dialogs be
> > destroyed automatically after a timeout?
> >
> > On Monday 19 April 2010, Anthony Minessale wrote:
> > > welll you are losing sip dialogs somewhere.
> > > It's not a leak its loss of sip dialogs most likely a side effect of
> > > topology problems.
> > >
> > >
> > > On Mon, Apr 19, 2010 at 11:03 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > The command was exactly as you suggested, FS was started with "-nonat
> > > > -vg" command line options, after few hours run (about 5K calls
> >
> > processed)
> >
> > > > FS was shut down gracefully.
> > > >
> > > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > > This is invalid.
> > > > >
> > > > > Those places are where sip calls are created and they are most
> > > > > definitely not leaking.
> > > > > If anything, it suggests dialogs that are up still and not yet
> > > > > destroyed
> > > >
> > > > at
> > > >
> > > > > the termination of the program.
> > > > > If this location in the code was a leak it would be gigs not megs
> > > >
> > > > missing.
> > > >
> > > > > Did you do the exact command I said (especially -vg param to FS)
> > > > > and then
> > > >
> > > > a
> > > >
> > > > > full clean shutdown all the way back to the shell?
> > > > >
> > > > > Are you maybe not getting the BYE to your calls creating open
> >
> > dialogs?
> >
> > > > > You may want to turn on the sip trace.
> > > > >
> > > > >
> > > > > On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
> > > > >
> > > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > > Valgrind doesn't show leaks in mod_nibblebill, most of the leaked
> > > >
> > > > memory
> > > >
> > > > > > was
> > > > > > in mod_sofia:
> > > > > >
> > > > > > ==12476== 53,064 bytes in 67 blocks are definitely lost in loss
> > > > > > record 138 of
> > > > > > 141
> > > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > > > linux/vgpreload_memcheck.so)
> > > > > > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite
> > > > > > (sofia.c:6538) ==12476==
> > > > > > ==12476==
> > > > > > ==12476== 92,664 bytes in 117 blocks are definitely lost in loss
> > > > > > record 139 of
> > > > > > 141
> > > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > > > > > linux/vgpreload_memcheck.so)
> > > > > > ==12476==    by 0x6A2F76F: sofia_glue_do_invite
> > > > > > (sofia_glue.c:1874) ==12476==
> > > > > >
> > > > > > BTW, do I understand FS code correct that channel private data
> > > > > > are
> > > >
> > > > always
> > > >
> > > > > > handled by sqlite, even if I enable core ODBC?
> > > > > >
> > > > > > On Monday 19 April 2010, Anthony Minessale wrote:
> > > > > > > 300M is tiny. try having 2000 channels up you can soar into a
> > > > > > > gig of
> > > > > >
> > > > > > usage.
> > > > > >
> > > > > > > That valgrind report only showed 100k leaked which means it was
> >
> > all
> >
> > > > > > > accounted for and torn down in the end.
> > > > > > >
> > > > > > > If you can find a specific leak in the nibble bill module with
> > > >
> > > > valgrind
> > > >
> > > > > > we
> > > > > >
> > > > > > > will gladly fix it and you
> > > > > > > can gladly thank us for giving you a way to make money by the
> > > > > > > second.
> > > > > > >
> > > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > > > > > --leak-resolution=high --show-reachable=yes
> > > > > > > /usr/local/freeswitch/bin/freeswitch -vg
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
> > > > > > >
> > > > > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > > > > Which valgrind options should I specify? I did run
> > > > > > > >
> > > > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > > >
> > > > --leak-
> > > >
> > > > > > > > resolution=high --show-reachable=yes ./freeswitch -nonat -vg
> > > > > > > >
> > > > > > > > 100K leaked is nothing compared to FS process RSS of 300M...
> > > > > > > >
> > > > > > > > On Sunday 18 April 2010, Anthony Minessale wrote:
> > > > > > > > > Also thatb suggests you leaked a whopping 100k
> > > > > > > > >
> > > > > > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
> > > > > > > >
> > > > > > > > anthony.minessale at gmail.com>
> > > > > > > >
> > > > > > > > > wrote:
> > > > > > > > >
> > > > > > > > > That's not very useful you need a full report with extended
> > > > > > > > > checking.
> > > > > > > > >
> > > > > > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
> > > > > >
> > > > > > sos at sokhapkin.dyndns.org>
> > > > > >
> > > > > > > > > wrote:
> > > > > > > > > > According to v...
> > > > > > > > > >
> > > > > > > > > >
> > > > > > > > > >
> > > > > > > > > >
> > > > > > > > > >
> > > > > > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
> > > > > > > > > > That is a completely incorrect statement. Usi...
> > > > > > > >
> > > > > > > > _______________________________________________
> > > > > > > > FreeSWITCH-users mailing list
> > > > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > > > UNSUBSCRIBE:
> > > > > >
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > >
> > > > > > > > http://www.freeswitch.org
> > > > > >
> > > > > > _______________________________________________
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > UNSUBSCRIBE:
> > > >
> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >
> > > > > > http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.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 Apr 19 10:12:35 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 12:12:35 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191306.28787.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191245.06964.sos@sokhapkin.dyndns.org>
	<t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
	<201004191306.28787.sos@sokhapkin.dyndns.org>
Message-ID: <FFA4F0C1-9BF6-41EE-B786-64E056BE92ED@freeswitch.org>

That still doesn't mean the dialogs were properly cleared by the sip stack before shutdown.  That is what anthony is getting at.  Valgrind does make mistakes at times about what is leaking in cases like this.

/b

On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:

> Yes, I stopped the traffic to the server and issued hupall CLI command before 
> shutting down FS.



From sos at sokhapkin.dyndns.org  Mon Apr 19 10:27:38 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 13:27:38 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <FFA4F0C1-9BF6-41EE-B786-64E056BE92ED@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191306.28787.sos@sokhapkin.dyndns.org>
	<FFA4F0C1-9BF6-41EE-B786-64E056BE92ED@freeswitch.org>
Message-ID: <201004191327.38189.sos@sokhapkin.dyndns.org>

Anyway it's offtopic already:-) The thread was about unbounded memory grows 
with enabled mod_nibblebill. On relatively busy server memory stays at about 
100M RSS for weeks if mod_nibblebill is not used, but grows to 800M in one day 
if nibbling is enabled.

On Monday 19 April 2010, Brian West wrote:
> That still doesn't mean the dialogs were properly cleared by the sip stack
>  before shutdown.  That is what anthony is getting at.  Valgrind does make
>  mistakes at times about what is leaking in cases like this.
> 
> /b
> 
> On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
> > Yes, I stopped the traffic to the server and issued hupall CLI command
> > before shutting down 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 anthony.minessale at gmail.com  Mon Apr 19 10:36:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 12:36:34 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191327.38189.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191306.28787.sos@sokhapkin.dyndns.org>
	<FFA4F0C1-9BF6-41EE-B786-64E056BE92ED@freeswitch.org>
	<201004191327.38189.sos@sokhapkin.dyndns.org>
Message-ID: <j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>

and did anything in the valgrind report even mention nibblebill?
you only sent the tiny excerpt.


On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Anyway it's offtopic already:-) The thread was about unbounded memory grows
> with enabled mod_nibblebill. On relatively busy server memory stays at
> about
> 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M in one
> day
> if nibbling is enabled.
>
> On Monday 19 April 2010, Brian West wrote:
> > That still doesn't mean the dialogs were properly cleared by the sip
> stack
> >  before shutdown.  That is what anthony is getting at.  Valgrind does
> make
> >  mistakes at times about what is leaking in cases like this.
> >
> > /b
> >
> > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
> > > Yes, I stopped the traffic to the server and issued hupall CLI command
> > > before shutting down 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
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Mon Apr 19 10:48:34 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 13:48:34 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191327.38189.sos@sokhapkin.dyndns.org>
	<j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>
Message-ID: <201004191348.34420.sos@sokhapkin.dyndns.org>

Few unimportant records only:

==12476== 7 bytes in 1 blocks are still reachable in loss record 21 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BECC00: mod_nibblebill_load (mod_nibblebill.c:132)

==12476== 7 bytes in 1 blocks are still reachable in loss record 22 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BECD14: mod_nibblebill_load (mod_nibblebill.c:134)

==12476== 7 bytes in 1 blocks are still reachable in loss record 23 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BECACC: mod_nibblebill_load (mod_nibblebill.c:128)

==12476== 8 bytes in 1 blocks are still reachable in loss record 28 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BEC9FB: mod_nibblebill_load (mod_nibblebill.c:127)

==12476== 9 bytes in 1 blocks are still reachable in loss record 30 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BECB19: mod_nibblebill_load (mod_nibblebill.c:129)

==12476== 10 bytes in 1 blocks are still reachable in loss record 34 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BECC8A: mod_nibblebill_load (mod_nibblebill.c:133)
==12476==
==12476==
==12476== 10 bytes in 1 blocks are still reachable in loss record 35 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BEC5D6: mod_nibblebill_load (mod_nibblebill.c:125)

==12476== 14 bytes in 1 blocks are still reachable in loss record 45 of 141
==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
linux/vgpreload_memcheck.so)
==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
==12476==    by 0x7BEC54D: mod_nibblebill_load (mod_nibblebill.c:124)

On Monday 19 April 2010, Anthony Minessale wrote:
> and did anything in the valgrind report even mention nibblebill?
> you only sent the tiny excerpt.
> 
> 
> On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Anyway it's offtopic already:-) The thread was about unbounded memory
> > grows with enabled mod_nibblebill. On relatively busy server memory stays
> > at about
> > 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M in
> > one day
> > if nibbling is enabled.
> >
> > On Monday 19 April 2010, Brian West wrote:
> > > That still doesn't mean the dialogs were properly cleared by the sip
> >
> > stack
> >
> > >  before shutdown.  That is what anthony is getting at.  Valgrind does
> >
> > make
> >
> > >  mistakes at times about what is leaking in cases like this.
> > >
> > > /b
> > >
> > > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
> > > > Yes, I stopped the traffic to the server and issued hupall CLI
> > > > command before shutting down 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-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From pjintheusa at gmail.com  Mon Apr 19 10:56:06 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 19 Apr 2010 13:56:06 -0400
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
Message-ID: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>

Hi there,

Has anyone got the android app SIP Agent working with FreeSWITCH.

I have the app saying it is registered correctly, but when I make a call it
is in the public context and not default.

I just wondered whether anyone has had any success.

thanks


pj


(PS: I have sipDroid working very nicely and iSIP on the iPhone also)
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From rupa at rupa.com  Mon Apr 19 11:12:53 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 19 Apr 2010 13:12:53 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191348.34420.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191327.38189.sos@sokhapkin.dyndns.org>
	<j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
Message-ID: <x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>

Those strdups are probably easily fixed.  But...  THey are just in the
module load, not as part of the module runtime.  So it is a one-time leak
not a per-call or per-nibble leak.

On Mon, Apr 19, 2010 at 12:48 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Few unimportant records only:
>
> ==12476== 7 bytes in 1 blocks are still reachable in loss record 21 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BECC00: mod_nibblebill_load (mod_nibblebill.c:132)
>
> ==12476== 7 bytes in 1 blocks are still reachable in loss record 22 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BECD14: mod_nibblebill_load (mod_nibblebill.c:134)
>
> ==12476== 7 bytes in 1 blocks are still reachable in loss record 23 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BECACC: mod_nibblebill_load (mod_nibblebill.c:128)
>
> ==12476== 8 bytes in 1 blocks are still reachable in loss record 28 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BEC9FB: mod_nibblebill_load (mod_nibblebill.c:127)
>
> ==12476== 9 bytes in 1 blocks are still reachable in loss record 30 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BECB19: mod_nibblebill_load (mod_nibblebill.c:129)
>
> ==12476== 10 bytes in 1 blocks are still reachable in loss record 34 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BECC8A: mod_nibblebill_load (mod_nibblebill.c:133)
> ==12476==
> ==12476==
> ==12476== 10 bytes in 1 blocks are still reachable in loss record 35 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BEC5D6: mod_nibblebill_load (mod_nibblebill.c:125)
>
> ==12476== 14 bytes in 1 blocks are still reachable in loss record 45 of 141
> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> linux/vgpreload_memcheck.so)
> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> ==12476==    by 0x7BEC54D: mod_nibblebill_load (mod_nibblebill.c:124)
>
> On Monday 19 April 2010, Anthony Minessale wrote:
> > and did anything in the valgrind report even mention nibblebill?
> > you only sent the tiny excerpt.
> >
> >
> > On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Anyway it's offtopic already:-) The thread was about unbounded memory
> > > grows with enabled mod_nibblebill. On relatively busy server memory
> stays
> > > at about
> > > 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M in
> > > one day
> > > if nibbling is enabled.
> > >
> > > On Monday 19 April 2010, Brian West wrote:
> > > > That still doesn't mean the dialogs were properly cleared by the sip
> > >
> > > stack
> > >
> > > >  before shutdown.  That is what anthony is getting at.  Valgrind does
> > >
> > > make
> > >
> > > >  mistakes at times about what is leaking in cases like this.
> > > >
> > > > /b
> > > >
> > > > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
> > > > > Yes, I stopped the traffic to the server and issued hupall CLI
> > > > > command before shutting down 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-user
> > > >s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Mon Apr 19 11:20:41 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 19 Apr 2010 13:20:41 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191327.38189.sos@sokhapkin.dyndns.org>
	<j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
	<x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
Message-ID: <h2nb39273531004191120r9bcc5d81l2d4e75c46c7089a9@mail.gmail.com>

I committed a change to free up the config vars at module shutdown.  Doesn't
fix your problem but should shut up valgrind for those allocations:

   3e600fb..ca9dfc3  master -> master
 .../applications/mod_nibblebill/mod_nibblebill.c   |   12 ++++++++++++
 1 files changed, 12 insertions(+), 0 deletions(-)


so if you update your git to at least version "ca9dfc3"  you'll have it.

On Mon, Apr 19, 2010 at 1:12 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Those strdups are probably easily fixed.  But...  THey are just in the
> module load, not as part of the module runtime.  So it is a one-time leak
> not a per-call or per-nibble leak.
>
> On Mon, Apr 19, 2010 at 12:48 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org> wrote:
>
>> Few unimportant records only:
>>
>> ==12476== 7 bytes in 1 blocks are still reachable in loss record 21 of 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BECC00: mod_nibblebill_load (mod_nibblebill.c:132)
>>
>> ==12476== 7 bytes in 1 blocks are still reachable in loss record 22 of 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BECD14: mod_nibblebill_load (mod_nibblebill.c:134)
>>
>> ==12476== 7 bytes in 1 blocks are still reachable in loss record 23 of 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BECACC: mod_nibblebill_load (mod_nibblebill.c:128)
>>
>> ==12476== 8 bytes in 1 blocks are still reachable in loss record 28 of 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BEC9FB: mod_nibblebill_load (mod_nibblebill.c:127)
>>
>> ==12476== 9 bytes in 1 blocks are still reachable in loss record 30 of 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BECB19: mod_nibblebill_load (mod_nibblebill.c:129)
>>
>> ==12476== 10 bytes in 1 blocks are still reachable in loss record 34 of
>> 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BECC8A: mod_nibblebill_load (mod_nibblebill.c:133)
>> ==12476==
>> ==12476==
>> ==12476== 10 bytes in 1 blocks are still reachable in loss record 35 of
>> 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BEC5D6: mod_nibblebill_load (mod_nibblebill.c:125)
>>
>> ==12476== 14 bytes in 1 blocks are still reachable in loss record 45 of
>> 141
>> ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>> linux/vgpreload_memcheck.so)
>> ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>> ==12476==    by 0x7BEC54D: mod_nibblebill_load (mod_nibblebill.c:124)
>>
>> On Monday 19 April 2010, Anthony Minessale wrote:
>> > and did anything in the valgrind report even mention nibblebill?
>> > you only sent the tiny excerpt.
>> >
>> >
>> > On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
>> >
>> > <sos at sokhapkin.dyndns.org>wrote:
>> > > Anyway it's offtopic already:-) The thread was about unbounded memory
>> > > grows with enabled mod_nibblebill. On relatively busy server memory
>> stays
>> > > at about
>> > > 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M in
>> > > one day
>> > > if nibbling is enabled.
>> > >
>> > > On Monday 19 April 2010, Brian West wrote:
>> > > > That still doesn't mean the dialogs were properly cleared by the sip
>> > >
>> > > stack
>> > >
>> > > >  before shutdown.  That is what anthony is getting at.  Valgrind
>> does
>> > >
>> > > make
>> > >
>> > > >  mistakes at times about what is leaking in cases like this.
>> > > >
>> > > > /b
>> > > >
>> > > > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
>> > > > > Yes, I stopped the traffic to the server and issued hupall CLI
>> > > > > command before shutting down 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-user
>> > > >s http://www.freeswitch.org
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>



-- 
-Rupa
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From brian at freeswitch.org  Mon Apr 19 11:21:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 13:21:48 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191327.38189.sos@sokhapkin.dyndns.org>
	<j2j191c3a031004191036kc2a584dfgee5ed643c7b0a9fc@mail.gmail.com>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
	<x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
Message-ID: <49324F46-FBCF-4654-9E61-92F2EC7252FE@freeswitch.org>

The question is why aren't they on the module pool?  Sounds like to me its not a leak but pool swell.  Someone is duping off the wrong pool.  Should use session pool if its session related data.  Sounds like maybe something is using the wrong pool.

/b

On Apr 19, 2010, at 1:12 PM, Rupa Schomaker wrote:

> Those strdups are probably easily fixed.  But...  THey are just in the module load, not as part of the module runtime.  So it is a one-time leak not a per-call or per-nibble leak.



From sos at sokhapkin.dyndns.org  Mon Apr 19 11:25:00 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 14:25:00 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
	<x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
Message-ID: <201004191425.00614.sos@sokhapkin.dyndns.org>

Yes, I agree that these strdups can be ignored. I believe I found where the 
memory problem comes from, the beginning of bill_event function in 
mod_nubblebill has

SWITCH_STANDARD_STREAM(sql_stream);

But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in the 
function where the memory is freed. Am I right?

#define SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data = 
malloc(SWITCH_CMD_CHUNK_LEN); \
        switch_assert(s.data);                                                                                          
\
        memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);                                                        
\
        s.end = s.data;                                                                                                         
\
        s.data_size = SWITCH_CMD_CHUNK_LEN;                                                                     
\
        s.write_function = switch_console_stream_write;                                         
\
        s.raw_write_function = switch_console_stream_raw_write;                         
\
        s.alloc_len = SWITCH_CMD_CHUNK_LEN;                                                                     
\
        s.alloc_chunk = SWITCH_CMD_CHUNK_LEN



On Monday 19 April 2010, Rupa Schomaker wrote:
> Those strdups are probably easily fixed.  But...  THey are just in the
> module load, not as part of the module runtime.  So it is a one-time leak
> not a per-call or per-nibble leak.
> 
> On Mon, Apr 19, 2010 at 12:48 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Few unimportant records only:
> >
> > ==12476== 7 bytes in 1 blocks are still reachable in loss record 21 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BECC00: mod_nibblebill_load (mod_nibblebill.c:132)
> >
> > ==12476== 7 bytes in 1 blocks are still reachable in loss record 22 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BECD14: mod_nibblebill_load (mod_nibblebill.c:134)
> >
> > ==12476== 7 bytes in 1 blocks are still reachable in loss record 23 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BECACC: mod_nibblebill_load (mod_nibblebill.c:128)
> >
> > ==12476== 8 bytes in 1 blocks are still reachable in loss record 28 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BEC9FB: mod_nibblebill_load (mod_nibblebill.c:127)
> >
> > ==12476== 9 bytes in 1 blocks are still reachable in loss record 30 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BECB19: mod_nibblebill_load (mod_nibblebill.c:129)
> >
> > ==12476== 10 bytes in 1 blocks are still reachable in loss record 34 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BECC8A: mod_nibblebill_load (mod_nibblebill.c:133)
> > ==12476==
> > ==12476==
> > ==12476== 10 bytes in 1 blocks are still reachable in loss record 35 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BEC5D6: mod_nibblebill_load (mod_nibblebill.c:125)
> >
> > ==12476== 14 bytes in 1 blocks are still reachable in loss record 45 of
> > 141 ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
> > linux/vgpreload_memcheck.so)
> > ==12476==    by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
> > ==12476==    by 0x7BEC54D: mod_nibblebill_load (mod_nibblebill.c:124)
> >
> > On Monday 19 April 2010, Anthony Minessale wrote:
> > > and did anything in the valgrind report even mention nibblebill?
> > > you only sent the tiny excerpt.
> > >
> > >
> > > On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Anyway it's offtopic already:-) The thread was about unbounded memory
> > > > grows with enabled mod_nibblebill. On relatively busy server memory
> >
> > stays
> >
> > > > at about
> > > > 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M
> > > > in one day
> > > > if nibbling is enabled.
> > > >
> > > > On Monday 19 April 2010, Brian West wrote:
> > > > > That still doesn't mean the dialogs were properly cleared by the
> > > > > sip
> > > >
> > > > stack
> > > >
> > > > >  before shutdown.  That is what anthony is getting at.  Valgrind
> > > > > does
> > > >
> > > > make
> > > >
> > > > >  mistakes at times about what is leaking in cases like this.
> > > > >
> > > > > /b
> > > > >
> > > > > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
> > > > > > Yes, I stopped the traffic to the server and issued hupall CLI
> > > > > > command before shutting down 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-user
> >
> > > > >s http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.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 Apr 19 11:46:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 13:46:22 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191425.00614.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
	<x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
	<201004191425.00614.sos@sokhapkin.dyndns.org>
Message-ID: <j2i191c3a031004191146m96d7a6c8zbc1d5aee7c8b7a9d@mail.gmail.com>

stream.data must be freed, in that case i think its being converted to char
by assigning sql to point at it but it is still freed in the end, that type
of mistake would be all over the place in valgrind.

look for a pool that is being reused over and over and never destroyed.



On Mon, Apr 19, 2010 at 1:25 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

>  Yes, I agree that these strdups can be ignored. I believe I found where
> the memory problem comes from, the beginning of bill_event function in
> mod_nubblebill has
>
> SWITCH_STANDARD_STREAM(sql_stream);
>
> But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in the
> function where the memory is freed. Am I right?
>
> #define SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> malloc(SWITCH_CMD_CHUNK_LEN); \
>
> switch_assert(s.data); \
>
> memset(s.data, 0, SWITCH_CMD_CHUNK_LEN); \
>
> s.end = s.data; \
>
> s.data_size = SWITCH_CMD_CHUNK_LEN; \
>
> s.write_function = switch_console_stream_write; \
>
> s.raw_write_function = switch_console_stream_raw_write; \
>
> s.alloc_len = SWITCH_CMD_CHUNK_LEN; \
>
> s.alloc_chunk = SWITCH_CMD_CHUNK_LEN
>
>  On Monday 19 April 2010, Rupa Schomaker wrote:
>
> > Those strdups are probably easily fixed. But... THey are just in the
>
> > module load, not as part of the module runtime. So it is a one-time leak
>
> > not a per-call or per-nibble leak.
>
> >
>
> > On Mon, Apr 19, 2010 at 12:48 PM, Sergey Okhapkin
>
> >
>
> > <sos at sokhapkin.dyndns.org>wrote:
>
> > > Few unimportant records only:
>
> > >
>
> > > ==12476== 7 bytes in 1 blocks are still reachable in loss record 21 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BECC00: mod_nibblebill_load (mod_nibblebill.c:132)
>
> > >
>
> > > ==12476== 7 bytes in 1 blocks are still reachable in loss record 22 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BECD14: mod_nibblebill_load (mod_nibblebill.c:134)
>
> > >
>
> > > ==12476== 7 bytes in 1 blocks are still reachable in loss record 23 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BECACC: mod_nibblebill_load (mod_nibblebill.c:128)
>
> > >
>
> > > ==12476== 8 bytes in 1 blocks are still reachable in loss record 28 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BEC9FB: mod_nibblebill_load (mod_nibblebill.c:127)
>
> > >
>
> > > ==12476== 9 bytes in 1 blocks are still reachable in loss record 30 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BECB19: mod_nibblebill_load (mod_nibblebill.c:129)
>
> > >
>
> > > ==12476== 10 bytes in 1 blocks are still reachable in loss record 34 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BECC8A: mod_nibblebill_load (mod_nibblebill.c:133)
>
> > > ==12476==
>
> > > ==12476==
>
> > > ==12476== 10 bytes in 1 blocks are still reachable in loss record 35 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BEC5D6: mod_nibblebill_load (mod_nibblebill.c:125)
>
> > >
>
> > > ==12476== 14 bytes in 1 blocks are still reachable in loss record 45 of
>
> > > 141 ==12476== at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>
> > > linux/vgpreload_memcheck.so)
>
> > > ==12476== by 0x457B6CF: strdup (in /lib/libc-2.10.1.so)
>
> > > ==12476== by 0x7BEC54D: mod_nibblebill_load (mod_nibblebill.c:124)
>
> > >
>
> > > On Monday 19 April 2010, Anthony Minessale wrote:
>
> > > > and did anything in the valgrind report even mention nibblebill?
>
> > > > you only sent the tiny excerpt.
>
> > > >
>
> > > >
>
> > > > On Mon, Apr 19, 2010 at 12:27 PM, Sergey Okhapkin
>
> > > >
>
> > > > <sos at sokhapkin.dyndns.org>wrote:
>
> > > > > Anyway it's offtopic already:-) The thread was about unbounded
> memory
>
> > > > > grows with enabled mod_nibblebill. On relatively busy server memory
>
> > >
>
> > > stays
>
> > >
>
> > > > > at about
>
> > > > > 100M RSS for weeks if mod_nibblebill is not used, but grows to 800M
>
> > > > > in one day
>
> > > > > if nibbling is enabled.
>
> > > > >
>
> > > > > On Monday 19 April 2010, Brian West wrote:
>
> > > > > > That still doesn't mean the dialogs were properly cleared by the
>
> > > > > > sip
>
> > > > >
>
> > > > > stack
>
> > > > >
>
> > > > > > before shutdown. That is what anthony is getting at. Valgrind
>
> > > > > > does
>
> > > > >
>
> > > > > make
>
> > > > >
>
> > > > > > mistakes at times about what is leaking in cases like this.
>
> > > > > >
>
> > > > > > /b
>
> > > > > >
>
> > > > > > On Apr 19, 2010, at 12:06 PM, Sergey Okhapkin wrote:
>
> > > > > > > Yes, I stopped the traffic to the server and issued hupall CLI
>
> > > > > > > command before shutting down 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-user
>
> > >
>
> > > > > >s http://www.freeswitch.org
>
> > > > >
>
> > > > > _______________________________________________
>
> > > > > FreeSWITCH-users mailing list
>
> > > > > FreeSWITCH-users at lists.freeswitch.org
>
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > > > > UNSUBSCRIBE:
>
> > >
>
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > >
>
> > > > > http://www.freeswitch.org
>
> > >
>
> > > _______________________________________________
>
> > > FreeSWITCH-users mailing list
>
> > > FreeSWITCH-users at lists.freeswitch.org
>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > > http://www.freeswitch.org
>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Mon Apr 19 12:09:57 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 14:09:57 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191425.00614.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191348.34420.sos@sokhapkin.dyndns.org>
	<x2rb39273531004191112m138877dr2228de50a9728d78@mail.gmail.com>
	<201004191425.00614.sos@sokhapkin.dyndns.org>
Message-ID: <24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>

No this isn't the problem it would show up on valgrind like crazy.

/b

On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:

> Yes, I agree that these strdups can be ignored. I believe I found where the memory problem comes from, the beginning of bill_event function in mod_nubblebill has
> SWITCH_STANDARD_STREAM(sql_stream);
> But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in the function where the memory is freed. Am I right?
> #define SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data = malloc(SWITCH_CMD_CHUNK_LEN); \
>         switch_assert(s.data);                                                                                          \
>         memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);                                                        \
>         s.end = s.data;                                                                                                         \
>         s.data_size = SWITCH_CMD_CHUNK_LEN;                                                                     \
>         s.write_function = switch_console_stream_write;                                         \
>         s.raw_write_function = switch_console_stream_raw_write;                         \
>         s.alloc_len = SWITCH_CMD_CHUNK_LEN;                                                                     \
>         s.alloc_chunk = SWITCH_CMD_CHUNK_LEN

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From sos at sokhapkin.dyndns.org  Mon Apr 19 12:17:09 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 15:17:09 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191425.00614.sos@sokhapkin.dyndns.org>
	<24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
Message-ID: <201004191517.09725.sos@sokhapkin.dyndns.org>

It would be the problem if custom_sql_* is set in nibblebill configuration, 
but it's not my case, I do not use custom sql. I'm looking at the code 
more....

On Monday 19 April 2010, Brian West wrote:
> No this isn't the problem it would show up on valgrind like crazy.
> 
> /b
> 
> On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:
> > Yes, I agree that these strdups can be ignored. I believe I found where
> > the memory problem comes from, the beginning of bill_event function in
> > mod_nubblebill has SWITCH_STANDARD_STREAM(sql_stream);
> > But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in the
> > function where the memory is freed. Am I right? #define
> > SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> > malloc(SWITCH_CMD_CHUNK_LEN); \ switch_assert(s.data);                   
> >                                                                       \
> > memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);                                 
> >                       \ s.end = s.data;                                  
> >                                                                       \
> > s.data_size = SWITCH_CMD_CHUNK_LEN;                                      
> >                               \ s.write_function =
> > switch_console_stream_write;                                         \
> > s.raw_write_function = switch_console_stream_raw_write;                  
> >       \ s.alloc_len = SWITCH_CMD_CHUNK_LEN;                              
> >                                       \ s.alloc_chunk =
> > SWITCH_CMD_CHUNK_LEN
> 



From brian at freeswitch.org  Mon Apr 19 12:21:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 14:21:15 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191517.09725.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191425.00614.sos@sokhapkin.dyndns.org>
	<24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
Message-ID: <C4557F48-E02E-438D-94F3-914621C50EE5@freeswitch.org>

We have many sets of eyes looking at this code also.  What distro are you on? what compiler are you using?

/b

On Apr 19, 2010, at 2:17 PM, Sergey Okhapkin wrote:

> It would be the problem if custom_sql_* is set in nibblebill configuration, 
> but it's not my case, I do not use custom sql. I'm looking at the code 
> more....
> 
> On Monday 19 April 2010, Brian West wrote:
>> No this isn't the problem it would show up on valgrind like crazy.
>> 
>> /b



From sos at sokhapkin.dyndns.org  Mon Apr 19 12:32:45 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 15:32:45 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <C4557F48-E02E-438D-94F3-914621C50EE5@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
	<C4557F48-E02E-438D-94F3-914621C50EE5@freeswitch.org>
Message-ID: <201004191532.45487.sos@sokhapkin.dyndns.org>

gentoo, gcc version 4.3.4 (Gentoo 4.3.4 p1.0, pie-10.1.5)

On Monday 19 April 2010, Brian West wrote:
> We have many sets of eyes looking at this code also.  What distro are you
>  on? what compiler are you using?
> 
> /b
> 
> On Apr 19, 2010, at 2:17 PM, Sergey Okhapkin wrote:
> > It would be the problem if custom_sql_* is set in nibblebill
> > configuration, but it's not my case, I do not use custom sql. I'm looking
> > at the code more....
> >
> > On Monday 19 April 2010, Brian West wrote:
> >> No this isn't the problem it would show up on valgrind like crazy.
> >>
> >> /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 rupa at rupa.com  Mon Apr 19 12:43:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 19 Apr 2010 14:43:32 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191517.09725.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191425.00614.sos@sokhapkin.dyndns.org>
	<24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
Message-ID: <t2yb39273531004191243p77c318bfn298d57c83c801f2a@mail.gmail.com>

ok, custom_sql path is fixed as of:

[master 9a74958] nibblebill - free properly if using custom_sql

On Mon, Apr 19, 2010 at 2:17 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> It would be the problem if custom_sql_* is set in nibblebill configuration,
> but it's not my case, I do not use custom sql. I'm looking at the code
> more....
>
> On Monday 19 April 2010, Brian West wrote:
> > No this isn't the problem it would show up on valgrind like crazy.
> >
> > /b
> >
> > On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:
> > > Yes, I agree that these strdups can be ignored. I believe I found where
> > > the memory problem comes from, the beginning of bill_event function in
> > > mod_nubblebill has SWITCH_STANDARD_STREAM(sql_stream);
> > > But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in
> the
> > > function where the memory is freed. Am I right? #define
> > > SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> > > malloc(SWITCH_CMD_CHUNK_LEN); \ switch_assert(s.data);
> > >                                                                       \
> > > memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);
> > >                       \ s.end = s.data;
> > >                                                                       \
> > > s.data_size = SWITCH_CMD_CHUNK_LEN;
> > >                               \ s.write_function =
> > > switch_console_stream_write;                                         \
> > > s.raw_write_function = switch_console_stream_raw_write;
> > >       \ s.alloc_len = SWITCH_CMD_CHUNK_LEN;
> > >                                       \ s.alloc_chunk =
> > > SWITCH_CMD_CHUNK_LEN
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From erkan at speedingtrade.com  Mon Apr 19 12:54:05 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Mon, 19 Apr 2010 22:54:05 +0300
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local><201004191225.41988.sos@sokhapkin.dyndns.org><81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>
	<201004191242.23056.sos@sokhapkin.dyndns.org>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>

Thank you. Now I see the call of URL in freeswitch command prompt, but the website will be not triggered.

Got error [500] posting to web server
Retry will be with url.....


will be coming up.
It's interesting because if I call my URL with my browser that work and I can give them all kind of parameters that I want.

For example:

http://192.168.2.21/callrec.aspx
http://192.168.2.21/callrec.aspx?var1=test
http://192.168.2.21/callrec.aspx?var1=test&var2=test2

and so on all kind of callings worked and I can read all parameters.
But I don't understand why FS gives an error and can't call the website.
The logfile of iis show me 

2010-04-19 19:48:50 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 0
2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 2
2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 4

Ok error 500 internal server error. But why error 500 if I call it with my browser that works perfect.

Thanks for your help.
Erkan





-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin
Sent: Monday, April 19, 2010 7:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call

You should enable mod_xml_cdr in modules.conf.xml

On Monday 19 April 2010, Erkan ?nl? wrote:
> I think yes, because the freeswitch.xml have the following lines inside.
> 
>   <section name="configuration" description="Various Configuration">
>     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml" />
>   </section>
> 
> Maybe I must activate it on another place?
> 
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
>  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
>  Okhapkin Sent: Monday, April 19, 2010 7:26 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
> 
> Did you load the module?
> 
> On Monday 19 April 2010, Erkan ?nl? wrote:
> > Hi FS Guys,
> >
> >
> >
> > I installed FS on Windows and try to use the xml_cdr.conf.xml to post the
> >  calling information's to a website/script.
> >
> >
> >
> > I change my xml_cdr.conf.xml as give on the website
> >
> > http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> >
> >
> >
> > <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> > <settings>
> >    <param name="url" value="http://localhost/callreg.aspx"/>
> >    <param name="retries" value="2"/>
> >    <param name="delay" value="120"/>
> >    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
> >    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
> >    <param name="encode" value="True"/>
> > </settings>
> >
> >
> >
> > I make a dummy aspx that write all given parameters to a file.
> >
> > But the aspx don't called.
> >
> >
> >
> > What I must do to get the information's into the
> >  http://localhost/callreg.aspx
> >
> >
> >
> > My dummy read the information's from the
> >  http://localhost/callreg.aspx?xxx=yyy and so on
> >
> > The dummy aspx read all parameters that given after the ?
> >
> > But no information's coming.
> >
> >
> >
> > Can anyone help me?
> >
> >
> >
> > Greetings
> >
> > Erkan
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Apr 19 12:58:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 14:58:36 -0500
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local><201004191225.41988.sos@sokhapkin.dyndns.org><81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>
	<201004191242.23056.sos@sokhapkin.dyndns.org>
	<81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
Message-ID: <E87F3A26-36FB-4B1E-888D-4DAAC2C304EB@freeswitch.org>

You're not doing a post when you go to it in the browser I suspect and we are in xml_cdrl...

/b

On Apr 19, 2010, at 2:54 PM, Erkan ?nl? wrote:

> Thank you. Now I see the call of URL in freeswitch command prompt, but the website will be not triggered.
> 
> Got error [500] posting to web server
> Retry will be with url.....
> 
> 
> will be coming up.
> It's interesting because if I call my URL with my browser that work and I can give them all kind of parameters that I want.
> 
> For example:
> 
> http://192.168.2.21/callrec.aspx
> http://192.168.2.21/callrec.aspx?var1=test
> http://192.168.2.21/callrec.aspx?var1=test&var2=test2
> 
> and so on all kind of callings worked and I can read all parameters.
> But I don't understand why FS gives an error and can't call the website.
> The logfile of iis show me 
> 
> 2010-04-19 19:48:50 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 0
> 2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 2
> 2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 4
> 
> Ok error 500 internal server error. But why error 500 if I call it with my browser that works perfect.
> 
> Thanks for your help.
> Erkan

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From sos at sokhapkin.dyndns.org  Mon Apr 19 13:03:36 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 16:03:36 -0400
Subject: [Freeswitch-users] xml_cdr.conf.xml    and ASPX call
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
	<201004191242.23056.sos@sokhapkin.dyndns.org>
	<81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
Message-ID: <201004191603.37052.sos@sokhapkin.dyndns.org>

You send GET requests from web browser, but FS sends POST requests.

On Monday 19 April 2010, Erkan ?nl? wrote:
> Thank you. Now I see the call of URL in freeswitch command prompt, but the
>  website will be not triggered.
> 
> Got error [500] posting to web server
> Retry will be with url.....
> 
> 
> will be coming up.
> It's interesting because if I call my URL with my browser that work and I
>  can give them all kind of parameters that I want.
> 
> For example:
> 
> http://192.168.2.21/callrec.aspx
> http://192.168.2.21/callrec.aspx?var1=test
> http://192.168.2.21/callrec.aspx?var1=test&var2=test2
> 
> and so on all kind of callings worked and I can read all parameters.
> But I don't understand why FS gives an error and can't call the website.
> The logfile of iis show me
> 
> 2010-04-19 19:48:50 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1
>  freeswitch-xml/1.0 500 0 0 0 2010-04-19 19:51:16 127.0.0.1 POST
>  /callregister.aspx - 81 - 127.0.0.1 freeswitch-xml/1.0 500 0 0 2
>  2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1
>  freeswitch-xml/1.0 500 0 0 4
> 
> Ok error 500 internal server error. But why error 500 if I call it with my
>  browser that works perfect.
> 
> Thanks for your help.
> Erkan
> 
> 
> 
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
>  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
>  Okhapkin Sent: Monday, April 19, 2010 7:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
> 
> You should enable mod_xml_cdr in modules.conf.xml
> 
> On Monday 19 April 2010, Erkan ?nl? wrote:
> > I think yes, because the freeswitch.xml have the following lines inside.
> >
> >   <section name="configuration" description="Various Configuration">
> >     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml" />
> >   </section>
> >
> > Maybe I must activate it on another place?
> >
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> >  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Sergey Okhapkin Sent: Monday, April 19, 2010 7:26 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
> >
> > Did you load the module?
> >
> > On Monday 19 April 2010, Erkan ?nl? wrote:
> > > Hi FS Guys,
> > >
> > >
> > >
> > > I installed FS on Windows and try to use the xml_cdr.conf.xml to post
> > > the calling information's to a website/script.
> > >
> > >
> > >
> > > I change my xml_cdr.conf.xml as give on the website
> > >
> > > http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> > >
> > >
> > >
> > > <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> > > <settings>
> > >    <param name="url" value="http://localhost/callreg.aspx"/>
> > >    <param name="retries" value="2"/>
> > >    <param name="delay" value="120"/>
> > >    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
> > >    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
> > >    <param name="encode" value="True"/>
> > > </settings>
> > >
> > >
> > >
> > > I make a dummy aspx that write all given parameters to a file.
> > >
> > > But the aspx don't called.
> > >
> > >
> > >
> > > What I must do to get the information's into the
> > >  http://localhost/callreg.aspx
> > >
> > >
> > >
> > > My dummy read the information's from the
> > >  http://localhost/callreg.aspx?xxx=yyy and so on
> > >
> > > The dummy aspx read all parameters that given after the ?
> > >
> > > But no information's coming.
> > >
> > >
> > >
> > > Can anyone help me?
> > >
> > >
> > >
> > > Greetings
> > >
> > > Erkan
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 19 13:23:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Apr 2010 16:23:27 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191532.45487.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
	<C4557F48-E02E-438D-94F3-914621C50EE5@freeswitch.org>
	<201004191532.45487.sos@sokhapkin.dyndns.org>
Message-ID: <5A41D9D6-6DDA-4954-ACE7-D7E3EFFC5B52@jerris.com>

Can you try to reproduce this on centos?  I have seen too many issues specific to Gentoo over the years that I would like to eliminate it from the start.

Mike

On Apr 19, 2010, at 3:32 PM, Sergey Okhapkin wrote:

> gentoo, gcc version 4.3.4 (Gentoo 4.3.4 p1.0, pie-10.1.5)
> 
> On Monday 19 April 2010, Brian West wrote:
>> We have many sets of eyes looking at this code also.  What distro are you
>> on? what compiler are you using?
>> 
>> /b
>> 
>> On Apr 19, 2010, at 2:17 PM, Sergey Okhapkin wrote:
>>> It would be the problem if custom_sql_* is set in nibblebill
>>> configuration, but it's not my case, I do not use custom sql. I'm looking
>>> at the code more....
>>> 
>>> On Monday 19 April 2010, Brian West wrote:
>>>> No this isn't the problem it would show up on valgrind like crazy.
>>>> 
>>>> /b
>> 



From josh at radianttiger.com  Mon Apr 19 13:26:05 2010
From: josh at radianttiger.com (Josh Rivers)
Date: Mon, 19 Apr 2010 13:26:05 -0700
Subject: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C5433@server1.st.local>
	<201004191225.41988.sos@sokhapkin.dyndns.org>
	<81C2CEF80046FB4F863A60D4347DD33A0C5434@server1.st.local>
	<201004191242.23056.sos@sokhapkin.dyndns.org>
	<81C2CEF80046FB4F863A60D4347DD33A0C5435@server1.st.local>
Message-ID: <k2sf8d0fbe51004191326jc251edaep45b57b4bad54b05c@mail.gmail.com>

I just ran into this one (using ASP.NET MVC, not ASPX). I had to use a
network sniffer to see the actual 500 error. It turned out to be
ASP.NETrequest validation on the post that was causing the failure.
The xml in the
post body trips the ASP.NET XSS injection protections. You can declaratively
turn off that feature per-page.

Josh

2010/4/19 Erkan ?nl? <erkan at speedingtrade.com>

> Thank you. Now I see the call of URL in freeswitch command prompt, but the
> website will be not triggered.
>
> Got error [500] posting to web server
> Retry will be with url.....
>
>
> will be coming up.
> It's interesting because if I call my URL with my browser that work and I
> can give them all kind of parameters that I want.
>
> For example:
>
> http://192.168.2.21/callrec.aspx
> http://192.168.2.21/callrec.aspx?var1=test
> http://192.168.2.21/callrec.aspx?var1=test&var2=test2
>
> and so on all kind of callings worked and I can read all parameters.
> But I don't understand why FS gives an error and can't call the website.
> The logfile of iis show me
>
> 2010-04-19 19:48:50 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1
> freeswitch-xml/1.0 500 0 0 0
> 2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1
> freeswitch-xml/1.0 500 0 0 2
> 2010-04-19 19:51:16 127.0.0.1 POST /callregister.aspx - 81 - 127.0.0.1
> freeswitch-xml/1.0 500 0 0 4
>
> Ok error 500 internal server error. But why error 500 if I call it with my
> browser that works perfect.
>
> Thanks for your help.
> Erkan
>
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
> Okhapkin
> Sent: Monday, April 19, 2010 7:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
>
> You should enable mod_xml_cdr in modules.conf.xml
>
> On Monday 19 April 2010, Erkan ?nl? wrote:
> > I think yes, because the freeswitch.xml have the following lines inside.
> >
> >   <section name="configuration" description="Various Configuration">
> >     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml" />
> >   </section>
> >
> > Maybe I must activate it on another place?
> >
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> >  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Sergey
> >  Okhapkin Sent: Monday, April 19, 2010 7:26 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] xml_cdr.conf.xml and ASPX call
> >
> > Did you load the module?
> >
> > On Monday 19 April 2010, Erkan ?nl? wrote:
> > > Hi FS Guys,
> > >
> > >
> > >
> > > I installed FS on Windows and try to use the xml_cdr.conf.xml to post
> the
> > >  calling information's to a website/script.
> > >
> > >
> > >
> > > I change my xml_cdr.conf.xml as give on the website
> > >
> > > http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> > >
> > >
> > >
> > > <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> > > <settings>
> > >    <param name="url" value="http://localhost/callreg.aspx"/>
> > >    <param name="retries" value="2"/>
> > >    <param name="delay" value="120"/>
> > >    <param name="log-dir" value="C:\Freeswitch\log\cdr"/>
> > >    <param name="err-log-dir" value=" C:\Freeswitch\log\errors"/>
> > >    <param name="encode" value="True"/>
> > > </settings>
> > >
> > >
> > >
> > > I make a dummy aspx that write all given parameters to a file.
> > >
> > > But the aspx don't called.
> > >
> > >
> > >
> > > What I must do to get the information's into the
> > >  http://localhost/callreg.aspx
> > >
> > >
> > >
> > > My dummy read the information's from the
> > >  http://localhost/callreg.aspx?xxx=yyy and so on
> > >
> > > The dummy aspx read all parameters that given after the ?
> > >
> > > But no information's coming.
> > >
> > >
> > >
> > > Can anyone help me?
> > >
> > >
> > >
> > > Greetings
> > >
> > > Erkan
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From josh at radianttiger.com  Mon Apr 19 13:28:50 2010
From: josh at radianttiger.com (Josh Rivers)
Date: Mon, 19 Apr 2010 13:28:50 -0700
Subject: [Freeswitch-users] Best low cost router/firewall
In-Reply-To: <y2z7d0bfd8c1004141601n29d6f2c9kf3436d5302d5fb18@mail.gmail.com>
References: <ED788322-BA9C-4A8A-8CFE-9C9CF1082AA1@tlainvestments.com>
	<m2o6b65470d1004141127ia45c735ci5dff0ebe7b6484b9@mail.gmail.com>
	<y2z7d0bfd8c1004141601n29d6f2c9kf3436d5302d5fb18@mail.gmail.com>
Message-ID: <y2of8d0fbe51004191328udd768418ubf17200c07ab3b3e@mail.gmail.com>

I've been using these:
http://www.netgate.com/product_info.php?cPath=60_85&products_id=553&osCsid=a81b30087c6039b659b00ca8c24d5010

<http://www.netgate.com/product_info.php?cPath=60_85&products_id=553&osCsid=a81b30087c6039b659b00ca8c24d5010>for
several years. They aren't quite as cheap as a consumer solution, but they
work well, and they last.

On Wed, Apr 14, 2010 at 4:01 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> the asus wlg-520gu also works nicely with DD-WRT. one thing nice w/ the
> unit - it has USB port for printer sharing and i was able to use LAN ports
> 2, 3, and 4 as a VLAN switch.
>
>
> On Thu, Apr 15, 2010 at 2:27 AM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> I've had good luck with Linksys routers with 3rd party firmware. (OpenWRT
>> or DD-Wrt)
>>
>> WRT54GS & WRT300N atm. Other routers are supported by the 3rd party
>> firmwares listed above as well.
>> The work nicely with the FS upnp firewall support.
>>
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Mon Apr 19 13:34:24 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Apr 2010 16:34:24 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <5A41D9D6-6DDA-4954-ACE7-D7E3EFFC5B52@jerris.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191532.45487.sos@sokhapkin.dyndns.org>
	<5A41D9D6-6DDA-4954-ACE7-D7E3EFFC5B52@jerris.com>
Message-ID: <201004191634.24959.sos@sokhapkin.dyndns.org>

No, I do not have centos servers, gentoo only...

But looks like I'm not the only unlucky:

http://www.mail-archive.com/freeswitch-
users at lists.freeswitch.org/msg18567.html

On Monday 19 April 2010, Michael Jerris wrote:
> Can you try to reproduce this on centos?  I have seen too many issues
>  specific to Gentoo over the years that I would like to eliminate it from
>  the start.
> 
> Mike
> 
> On Apr 19, 2010, at 3:32 PM, Sergey Okhapkin wrote:
> > gentoo, gcc version 4.3.4 (Gentoo 4.3.4 p1.0, pie-10.1.5)
> >
> > On Monday 19 April 2010, Brian West wrote:
> >> We have many sets of eyes looking at this code also.  What distro are
> >> you on? what compiler are you using?
> >>
> >> /b
> >>
> >> On Apr 19, 2010, at 2:17 PM, Sergey Okhapkin wrote:
> >>> It would be the problem if custom_sql_* is set in nibblebill
> >>> configuration, but it's not my case, I do not use custom sql. I'm
> >>> looking at the code more....
> >>>
> >>> On Monday 19 April 2010, Brian West wrote:
> >>>> No this isn't the problem it would show up on valgrind like crazy.
> >>>>
> >>>> /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  Mon Apr 19 13:54:06 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Apr 2010 15:54:06 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191634.24959.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191532.45487.sos@sokhapkin.dyndns.org>
	<5A41D9D6-6DDA-4954-ACE7-D7E3EFFC5B52@jerris.com>
	<201004191634.24959.sos@sokhapkin.dyndns.org>
Message-ID: <113747C3-8C7E-482E-8633-41D3E73821F6@freeswitch.org>

It only takes a few moments to go on over to www.centos.org and create one.

./b

On Apr 19, 2010, at 3:34 PM, Sergey Okhapkin wrote:

> No, I do not have centos servers, gentoo only...
> 
> But looks like I'm not the only unlucky:
> 
> http://www.mail-archive.com/freeswitch-
> users at lists.freeswitch.org/msg18567.html
> 
> On Monday 19 April 2010, Michael Jerris wrote:
>> Can you try to reproduce this on centos?  I have seen too many issues
>> specific to Gentoo over the years that I would like to eliminate it from
>> the start.
>> 
>> Mike

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From jmesquita at freeswitch.org  Mon Apr 19 14:12:49 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 19 Apr 2010 18:12:49 -0300
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
Message-ID: <z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>

uuid_simplify will issue the refer...

May I ask what application you are developing?

Regards,
Jo?o Mesquita
FSComm developer


On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:

> I'm having a problem with attended transfers where the destination of the
> transfer is a FreeSWITCH based application such as FSComm.  (It should be
> noted that in my setup the phone performing the transfer and the caller
> which is being transferred are parties of another SIP server.)  What I see,
> from a SIP signaling standpoint, is that after FreeSWITCH receives and
> acknowledges the INVITE w/Replaces it does not terminate the initial call
> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
> application side, FS still thinks that both the initial call leg and
> transferred call leg are active.  I experimented with trying to explicitly
> terminate the initial call leg by using uuid_kill, but this caused FS to
> kill all legs of the call.  Is there a specific action that the application
> must take in order for the transfer to complete?
>
> -Mardy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lon at kickasspixels.com  Mon Apr 19 14:44:22 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 19 Apr 2010 14:44:22 -0700
Subject: [Freeswitch-users] Billing system
In-Reply-To: <m2raea1d7231004172342z6b03b2dbh1bd4703f582a1542@mail.gmail.com>
References: <m2raea1d7231004172342z6b03b2dbh1bd4703f582a1542@mail.gmail.com>
Message-ID: <04D8C40A-835A-4D5F-9A98-438D89BE4427@kickasspixels.com>

Hi,

In addition to rolling your own on top of mod_nibblebill, I know of two commercial providers.

Cyneric at http://cyneric.com

And

DTH at http://dthsoftware.com

Both rely on Microsoft SQL Server, in case that is an issue.

There is an open source solution, ASTPP, at http://astpp.org

I'll add those to the wiki now.

--
Lon Baker

On Apr 17, 2010, at 11:42 PM, Shrouk Khan <shroukkhan at gmail.com> wrote:

> hi all,
> i am sure this question has been asked a million times , but i did not find any satisfactory answers :) so here goes again :
> 
> what is the most preferred billing system to be used with freeswitch (preferably something that is capable of leveraging the benefits of mod_nibblebill )
> 
> 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 mardy at voysys.com  Mon Apr 19 14:47:26 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Mon, 19 Apr 2010 17:47:26 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
Message-ID: <461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>


On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:

> uuid_simplify will issue the refer...

I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.

> 
> May I ask what application you are developing?

An ACD.

> 
> Regards,
> Jo?o Mesquita
> FSComm developer
> 
> 
> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
> 
> -Mardy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 19 15:19:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 17:19:32 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
Message-ID: <g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>

but what is the client sending the REFER?

FS gets refer+replaces all the time, if it's the one where the dest is on
another box (aka the nightmare xfer that you should see references to in the
debug log if so) then it will not complete until that far end call is
answered.

FS handles this scenerio for us hundreds of times a day using a wide range
of sip devices so perhaps
your UA has an interop problem.


On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:

>
> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>
> uuid_simplify will issue the refer...
>
>
> I looked at uuid_simplify and if I understand it correctly it is for use
> when one wants to act as the transfer controller.  In my case, FS is the
> transfer destination.  Another phone has already generated the refer and FS
> has been sent an invite with replaces.
>
>
> May I ask what application you are developing?
>
>
> An ACD.
>
>
> Regards,
> Jo?o Mesquita
> FSComm developer
>
>
> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> I'm having a problem with attended transfers where the destination of the
>> transfer is a FreeSWITCH based application such as FSComm.  (It should be
>> noted that in my setup the phone performing the transfer and the caller
>> which is being transferred are parties of another SIP server.)  What I see,
>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>> application side, FS still thinks that both the initial call leg and
>> transferred call leg are active.  I experimented with trying to explicitly
>> terminate the initial call leg by using uuid_kill, but this caused FS to
>> kill all legs of the call.  Is there a specific action that the application
>> must take in order for the transfer to complete?
>>
>> -Mardy
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Apr 19 16:14:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Apr 2010 16:14:18 -0700
Subject: [Freeswitch-users] FreeSWITCH Connector For Vestec Speech
	Recognition Engine
Message-ID: <w2y87f2f3b91004191614h9a66f6e0jdedb9e493d109742@mail.gmail.com>

Good news! FreeSWITCH and Vestec have teamed up to deliver a FreeSWITCH
connector for their speech recognition engine:

http://www.freeswitch.org/node/252

Kudos to everyone who made this possible. Please go buy some speech
recognition licenses and start building some great new applications!

-MC
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From mardy at voysys.com  Mon Apr 19 16:33:40 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Mon, 19 Apr 2010 19:33:40 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
Message-ID: <0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>

I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.

-Mardy


On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:

> but what is the client sending the REFER?
> 
> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
> 
> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
> your UA has an interop problem.
> 
> 
> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
> 
> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
> 
>> uuid_simplify will issue the refer...
> 
> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
> 
>> 
>> May I ask what application you are developing?
> 
> An ACD.
> 
>> 
>> Regards,
>> Jo?o Mesquita
>> FSComm developer
>> 
>> 
>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>> 
>> -Mardy
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 19 16:53:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Apr 2010 18:53:51 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
Message-ID: <i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>

did you try just setting up 2 phones on plain fresh FS install, and calling
them normally and transferring them around?
That description is still pretty vague? What is an Event Socket application,
which has nothing to do with sip and sip transfers, that's a FS protocol.


On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:

> I have two phones (Polycom) and an event_socket application, all of which
> are using a SIP proxy for call routing.  The first phone calls the second
> phone.  The second phone then attempts to transfer the call to the
> FS/event_socket application by first placing the call on hold and then
> calling the FS application, followed by a consultative transfer.  The REFER
> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>  The transferred call leg appears to be answered by FS and the application
> receives a uuid_bridge event with the UUID of the new call leg.  The problem
> that I see is that the original call leg, created when the user called the
> FS application to announce the transfer, does not get canceled by FS and
> subsequently does not send the BYE back to the Polycom.  Is there something
> that I need to do at the event_socket application to complete the transfer?
>  I've tried killing the UUID associated with the first call leg as well as
> issuing an "answer" command to the transferred call leg UUID, but no luck.
>
> -Mardy
>
>
> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>
> but what is the client sending the REFER?
>
> FS gets refer+replaces all the time, if it's the one where the dest is on
> another box (aka the nightmare xfer that you should see references to in the
> debug log if so) then it will not complete until that far end call is
> answered.
>
> FS handles this scenerio for us hundreds of times a day using a wide range
> of sip devices so perhaps
> your UA has an interop problem.
>
>
> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>
>>
>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>
>> uuid_simplify will issue the refer...
>>
>>
>> I looked at uuid_simplify and if I understand it correctly it is for use
>> when one wants to act as the transfer controller.  In my case, FS is the
>> transfer destination.  Another phone has already generated the refer and FS
>> has been sent an invite with replaces.
>>
>>
>> May I ask what application you are developing?
>>
>>
>> An ACD.
>>
>>
>> Regards,
>> Jo?o Mesquita
>> FSComm developer
>>
>>
>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>
>>> I'm having a problem with attended transfers where the destination of the
>>> transfer is a FreeSWITCH based application such as FSComm.  (It should be
>>> noted that in my setup the phone performing the transfer and the caller
>>> which is being transferred are parties of another SIP server.)  What I see,
>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>> application side, FS still thinks that both the initial call leg and
>>> transferred call leg are active.  I experimented with trying to explicitly
>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>> kill all legs of the call.  Is there a specific action that the application
>>> must take in order for the transfer to complete?
>>>
>>> -Mardy
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mardy at voysys.com  Mon Apr 19 17:21:34 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Mon, 19 Apr 2010 20:21:34 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
Message-ID: <1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>

The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.

To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.

-Mardy

On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:

> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
> 
> 
> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
> 
> -Mardy
> 
> 
> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
> 
>> but what is the client sending the REFER?
>> 
>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>> 
>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>> your UA has an interop problem.
>> 
>> 
>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>> 
>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>> 
>>> uuid_simplify will issue the refer...
>> 
>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>> 
>>> 
>>> May I ask what application you are developing?
>> 
>> An ACD.
>> 
>>> 
>>> Regards,
>>> Jo?o Mesquita
>>> FSComm developer
>>> 
>>> 
>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>> 
>>> -Mardy
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From devel at thom.fr.eu.org  Tue Apr 20 02:23:57 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 20 Apr 2010 11:23:57 +0200
Subject: [Freeswitch-users] Problem sending DTMF using an FXS channel
In-Reply-To: <005701cad755$78090710$681b1530$@fr.eu.org>
References: <005701cad755$78090710$681b1530$@fr.eu.org>
Message-ID: <5b4f305f7bc59508b887d571c979e3f7@thom.fr.eu.org>



Any news on this topic ? 

On Thu, 8 Apr 2010 21:55:35 +0200, 

wrote:  


Log is in http://pastebin.freeswitch.org/12665 [1] 

Dialplan is
following (included in default freeswitch) : 

I forgot to mention that I'm
using sangoma A400 with HW DTMF detection. 

Fran?ois 

DE :
freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] DE LA PART DE
Michael Collins
ENVOY : jeudi 8 avril 2010 19:59
 :
freeswitch-users at lists.freeswitch.org
OBJET : Re: [Freeswitch-users]
Problem sending DTMF using an FXS channel  

Pastebin your dialplan config
that handles this call as well as a debug trace from the console.
 -MC 


On Thu, Apr 8, 2010 at 10:03 AM,    

Hello, 

I'm having trouble with
calls to remote IVR using DTMF, when the A leg is an FXS port. 

What
happens is when the key is pressed on the phone, the DTMF is sent inband to
the callee party as voice, but also detected by freeswitch and so resent by
freeswitch to the callee party. This results in unusability of called IVR.


Is there any setting that could be used to prevent freeswitch from
detecting DTMF and/or prevent freeswitch from resending the DTMF. 

Thanks


Fran?ois   

 _______________________________________________

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
[2]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[4]  

 

Links:
------
[1] http://pastebin.freeswitch.org/12665
[2]
mailto:FreeSWITCH-users at lists.freeswitch.org
[3]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[4]
http://www.freeswitch.org
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From mcampbellsmith at gmail.com  Tue Apr 20 06:12:45 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 20 Apr 2010 23:12:45 +1000
Subject: [Freeswitch-users] FreeSwitch GUI's: Import current FS configuration
Message-ID: <n2n33c87fa31004200612i1190cb29z433e3d3e6e306d81@mail.gmail.com>

Hi!

Of the FS GUI's that are available, does anyone know if any support
importing the current configuration into the GUI database?  ie So that
I dont have to start from scratch and redo dialplans, extensions,
contexts etc.

I would like to try a GUI, but it will be quite a bit of work to start
from scratch...

Any info appreciated!

Thanks!


From brian at freeswitch.org  Tue Apr 20 06:33:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Apr 2010 08:33:31 -0500
Subject: [Freeswitch-users] FreeSwitch GUI's: Import current FS
	configuration
In-Reply-To: <n2n33c87fa31004200612i1190cb29z433e3d3e6e306d81@mail.gmail.com>
References: <n2n33c87fa31004200612i1190cb29z433e3d3e6e306d81@mail.gmail.com>
Message-ID: <460B4CD5-3D2B-4A67-A588-19076E7D41C4@freeswitch.org>

Yes the only gui I'm aware of that does that is emacs.  :P

/b

On Apr 20, 2010, at 8:12 AM, Mark Campbell-Smith wrote:

> Hi!
> 
> Of the FS GUI's that are available, does anyone know if any support
> importing the current configuration into the GUI database?  ie So that
> I dont have to start from scratch and redo dialplans, extensions,
> contexts etc.
> 
> I would like to try a GUI, but it will be quite a bit of work to start
> from scratch...
> 
> Any info appreciated!
> 
> Thanks!



From jcasale at activenetwerx.com  Tue Apr 20 08:09:45 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 20 Apr 2010 15:09:45 +0000
Subject: [Freeswitch-users] FreeSwitch GUI's: Import current
	FS	configuration
In-Reply-To: <460B4CD5-3D2B-4A67-A588-19076E7D41C4@freeswitch.org>
References: <n2n33c87fa31004200612i1190cb29z433e3d3e6e306d81@mail.gmail.com>
	<460B4CD5-3D2B-4A67-A588-19076E7D41C4@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E995A265@prato.activenetwerx.local>

>Yes the only gui I'm aware of that does that is emacs.  :P

Lol... people still use emacs:) vi all the way!
Otoh, you might guess while I havent been able to give up `joe`, heh.


From kenfulmer at icstechnologysolutions.com  Tue Apr 20 09:38:29 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 11:38:29 -0500
Subject: [Freeswitch-users] ANI / DNIS Number Translation
Message-ID: <00e001cae0a7$e8996e10$b9cc4a30$@com>

Is it possible to translate the ANI or the DNIS using Freeswitch? 

 

Thanks,

 

Ken Fulmer

 

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From brian at freeswitch.org  Tue Apr 20 09:49:36 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Apr 2010 11:49:36 -0500
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <00e001cae0a7$e8996e10$b9cc4a30$@com>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>
Message-ID: <6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>

Can you provide some examples?  All of what you want is doable but use cases would be most helpful.

/b

On Apr 20, 2010, at 11:38 AM, Ken Fulmer wrote:

> Is it possible to translate the ANI or the DNIS using Freeswitch?
>  
> Thanks,
>  
> Ken Fulmer

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From david.ponzone at gmail.com  Tue Apr 20 09:55:04 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 20 Apr 2010 18:55:04 +0200
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <00e001cae0a7$e8996e10$b9cc4a30$@com>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>
Message-ID: <E075E30C-CAE9-41DE-914B-C24242A1C131@gmail.com>

Ken,

can you define "translate" ?
You can change them, of course, as you can change most parameters of a  
call.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 20/04/2010 ? 18:38, Ken Fulmer a ?crit :

> Is it possible to translate the ANI or the DNIS using Freeswitch?
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kenfulmer at icstechnologysolutions.com  Tue Apr 20 10:02:36 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 12:02:36 -0500
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>
	<6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>
Message-ID: <010301cae0ab$471f8d40$d55ea7c0$@com>

For outbound calls to the PSTN, we might need to strip a leading "9". For
inbound calls, we might need to convert 10 digit DID's to internal 4 digit
directory numbers.

 

Those are two of the most common examples we encounter.

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, April 20, 2010 11:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ANI / DNIS Number Translation

 

Can you provide some examples?  All of what you want is doable but use cases
would be most helpful.

 

/b

 

On Apr 20, 2010, at 11:38 AM, Ken Fulmer wrote:





Is it possible to translate the ANI or the DNIS using Freeswitch?

 

Thanks,

 

Ken Fulmer

 

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From david.ponzone at gmail.com  Tue Apr 20 10:23:22 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 20 Apr 2010 19:23:22 +0200
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <010301cae0ab$471f8d40$d55ea7c0$@com>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>
	<6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>
	<010301cae0ab$471f8d40$d55ea7c0$@com>
Message-ID: <2ED8B5B6-0D83-4EBA-B3CC-8CE50042BBA6@gmail.com>

Ken,

with FS, that's more than trivial.

To strip a prefix from the dialed number for outbound calls, the idea  
is to do this in your extension (syntax is not correct, I dont want to  
spoil you from the pleasure to write it yourself :) ):

condition destination_number regexp ^9(\d+)$
  bridge sofia/gateway/whatever/$1

-> $1 will contain what's between the () in the regexp


For inbound calls, what you want to do is the typical config of FS  
(like in the default one) as all phones on FS use a 4 digits userid.
So the idea is to do:
condition destination_number regexp ^415917220(\d)$
  bridge user/100$1

-> in this example, I consider you have 10 DIDs from 4159172200 to  
4159172209 and that those DIDs are allocated to 10 users, from 1000 to  
1009.

That's really an example. Your mileage may vary.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 20/04/2010 ? 19:02, Ken Fulmer a ?crit :

> For outbound calls to the PSTN, we might need to strip a leading  
> ?9?. For inbound calls, we might need to convert 10 digit DID?s  
> to internal 4 digit directory numbers.
>
> Those are two of the most common examples we encounter.
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Brian West
> Sent: Tuesday, April 20, 2010 11:50 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ANI / DNIS Number Translation
>
> Can you provide some examples?  All of what you want is doable but  
> use cases would be most helpful.
>
> /b
>
> On Apr 20, 2010, at 11:38 AM, Ken Fulmer wrote:
>
>
> Is it possible to translate the ANI or the DNIS using Freeswitch?
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jcasale at activenetwerx.com  Tue Apr 20 10:34:58 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 20 Apr 2010 17:34:58 +0000
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <010301cae0ab$471f8d40$d55ea7c0$@com>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>
	<6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>
	<010301cae0ab$471f8d40$d55ea7c0$@com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E995A4C5@prato.activenetwerx.local>

>For outbound calls to the PSTN, we might need to strip a leading "9". For inbound calls, we might need to convert 10 digit DID's to internal 4 digit directory numbers.
>
>Those are two of the most common examples we encounter.

Regex group matches? That's how I do it...

<condition field="destination_number" expression="^(?:1{0,1})(403|587)(\d{7})$">
...
<action application="bridge" data="sofia/gateway/foo.ca/1$1$2|$1$2 at bar.local:5060"/>

Just an example how I dial as needed from a more flexible input...



From kenfulmer at icstechnologysolutions.com  Tue Apr 20 11:35:05 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 13:35:05 -0500
Subject: [Freeswitch-users] ANI / DNIS Number Translation
In-Reply-To: <2ED8B5B6-0D83-4EBA-B3CC-8CE50042BBA6@gmail.com>
References: <00e001cae0a7$e8996e10$b9cc4a30$@com>	<6AF29F73-8251-4503-957E-80FBC9CB3E56@freeswitch.org>	<010301cae0ab$471f8d40$d55ea7c0$@com>
	<2ED8B5B6-0D83-4EBA-B3CC-8CE50042BBA6@gmail.com>
Message-ID: <014401cae0b8$34118c50$9c34a4f0$@com>

Thanks guys. That is exactly what I needed. I?m re-reading the dial plan page to better understand pattern matching / substitutions.

 

Thanks again!

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Tuesday, April 20, 2010 12:23 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ANI / DNIS Number Translation

 

Ken,

 

with FS, that's more than trivial.

 

To strip a prefix from the dialed number for outbound calls, the idea is to do this in your extension (syntax is not correct, I dont want to spoil you from the pleasure to write it yourself :) ):

 

condition destination_number regexp ^9(\d+)$

 bridge sofia/gateway/whatever/$1

 

-> $1 will contain what's between the () in the regexp

 

 

For inbound calls, what you want to do is the typical config of FS (like in the default one) as all phones on FS use a 4 digits userid.

So the idea is to do:

condition destination_number regexp ^415917220(\d)$

 bridge user/100$1

 

-> in this example, I consider you have 10 DIDs from 4159172200 to 4159172209 and that those DIDs are allocated to 10 users, from 1000 to 1009.

 

That's really an example. Your mileage may vary.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 20/04/2010 ? 19:02, Ken Fulmer a ?crit :





For outbound calls to the PSTN, we might need to strip a leading ?9?. For inbound calls, we might need to convert 10 digit DID?s to internal 4 digit directory numbers.

 

Those are two of the most common examples we encounter.

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, April 20, 2010 11:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ANI / DNIS Number Translation

 

Can you provide some examples?  All of what you want is doable but use cases would be most helpful.

 

/b

 

On Apr 20, 2010, at 11:38 AM, Ken Fulmer wrote:






Is it possible to translate the ANI or the DNIS using Freeswitch?

 

Thanks,

 

Ken Fulmer

 

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

 

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From kenfulmer at icstechnologysolutions.com  Tue Apr 20 13:03:41 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 15:03:41 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>	<012d01cad807$27f5a0a0$77e0e1e0$@com>
	<AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>
Message-ID: <015b01cae0c4$933c12c0$b9b43840$@com>

I'm now using an ACL list called "lan". 

 

    <list name="lan" default="allow">

     <node type="allow" cidr="10.10.3.0/25"/>

    </list>

 

In the external sip profile, I have the following statement:

 

    <param name="local-network-acl" value="lan"/>

 

I still get the Proxy Authentication Required error. Am I doing something
wrong?

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tjardick
van der Kraan
Sent: Sunday, April 11, 2010 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

Use the CIDR XML key not domain:

 

http://wiki.freeswitch.org/wiki/Acl

 

Regards,

 

Tj

 

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:





Per your suggestion, I changed the following in the
conf/autoload_configs/acl.conf.xml file:

 

    <list name="domains" default="deny">

      <node type="allow" domain="10.10.3.10"/>

      <node type="allow" domain="10.10.3.11"/>

    </list>

 

10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
However, the calls still fail with the 407 Proxy Authentication Required
message.

 

I get the following log output when I issue the command, reloadacl:

 

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
rfc1918.auto default (deny)

freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195
Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
wan.auto default (allow)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
nat.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
10.10.3.12/255.255.255.128 (deny) to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
loopback.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8
(allow) [] to list loopback.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
localnet.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
10.10.3.12/255.255.255.128 (allow) to list localnet.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
domains default (deny)

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.10

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.11

 

Am I doing something incorrectly?

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 08, 2010 6:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

 

On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Actually, I did purchase a license and installed it today. One call
establishes at 729. When I hang up the phone and try again, it's 711.

Make sure that the encoder/decoder isn't still in use prior to trying the
second call. After you hang up, do a "show channels" and see if the call is
still "up" or not. Also, do "g729_status" to see if the encoder or decoder
is in use. Keep doing "g729_status" until the 'coders are not in use. If
there is a long delay then open up a JIRA ticket on jira.freeswitch.org.

 

The Proxy Authentication Required is being sent by FreeSwitch to the
internal PBX. I have registration disabled on the FreeSwitch gateway and the
internal server.

By default the SIP profile will challenge if the IP address of the caller is
not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
"domains" node. Add your PBX's IP address. You'll see an example in the
comments. Once you're done editing, save the file and then go to the fs_cli
and do:
reloadacl reloadxml

Then make a call from PBX to FS and it should go through.
-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 david.ponzone at gmail.com  Tue Apr 20 13:24:31 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 20 Apr 2010 22:24:31 +0200
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <015b01cae0c4$933c12c0$b9b43840$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>	<012d01cad807$27f5a0a0$77e0e1e0$@com>
	<AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>
	<015b01cae0c4$933c12c0$b9b43840$@com>
Message-ID: <E687000A-27B0-4013-8240-92397B4B5481@gmail.com>

ken,

are you sure it's not apply-inbound-acl you want to set ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 20/04/2010 ? 22:03, Ken Fulmer a ?crit :

> I?m now using an ACL list called ?lan?.
>
>     <list name="lan" default="allow">
>      <node type="allow" cidr="10.10.3.0/25"/>
>     </list>
>
> In the external sip profile, I have the following statement:
>
>     <param name="local-network-acl" value="lan"/>
>
> I still get the Proxy Authentication Required error. Am I doing  
> something wrong?
>
> Thanks,
>
> Ken
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Tjardick van der Kraan
> Sent: Sunday, April 11, 2010 4:36 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
> Use the CIDR XML key not domain:
>
> http://wiki.freeswitch.org/wiki/Acl
>
> Regards,
>
> Tj
>
> On 09 Apr 2010, at 19:07, Ken Fulmer wrote:
>
>
> Per your suggestion, I changed the following in the conf/ 
> autoload_configs/acl.conf.xml file:
>
>     <list name="domains" default="deny">
>       <node type="allow" domain="10.10.3.10"/>
>       <node type="allow" domain="10.10.3.11"/>
>     </list>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal  
> servers. However, the calls still fail with the 407 Proxy  
> Authentication Required message.
>
> I get the following log output when I issue the command, reloadacl:
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip  
> list rfc1918.auto default (deny)
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]  
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip  
> list wan.auto default (allow)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (deny) [] to list wan.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip  
> list nat.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding  
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 10.0.0.0/8 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 172.16.0.0/12 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 192.168.0.0/16 (allow) [] to list nat.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip  
> list loopback.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding  
> 127.0.0.0/8 (allow) [] to list loopback.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip  
> list localnet.auto default (deny)
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding  
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip  
> list domains default (deny)
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.10
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot  
> locate domain 10.10.3.11
>
> Am I doing something incorrectly?
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, April 08, 2010 6:25 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Two Major Problems
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Actually, I did purchase a license and installed it today. One call  
> establishes at 729. When I hang up the phone and try again, it?s  
> 711.
> Make sure that the encoder/decoder isn't still in use prior to  
> trying the second call. After you hang up, do a "show channels" and  
> see if the call is still "up" or not. Also, do "g729_status" to see  
> if the encoder or decoder is in use. Keep doing "g729_status" until  
> the 'coders are not in use. If there is a long delay then open up a  
> JIRA ticket on jira.freeswitch.org.
>
> The Proxy Authentication Required is being sent by FreeSwitch to the  
> internal PBX. I have registration disabled on the FreeSwitch gateway  
> and the internal server.
> By default the SIP profile will challenge if the IP address of the  
> caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml  
> and locate the "domains" node. Add your PBX's IP address. You'll see  
> an example in the comments. Once you're done editing, save the file  
> and then go to the fs_cli and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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 kenfulmer at icstechnologysolutions.com  Tue Apr 20 13:52:21 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 15:52:21 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <E687000A-27B0-4013-8240-92397B4B5481@gmail.com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>	<012d01cad807$27f5a0a0$77e0e1e0$@com>	<AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>	<015b01cae0c4$933c12c0$b9b43840$@com>
	<E687000A-27B0-4013-8240-92397B4B5481@gmail.com>
Message-ID: <019001cae0cb$5ff19050$1fd4b0f0$@com>

I?ve tried this setting as well:

 

    <param name="apply-inbound-acl" value="lan"/> 

 

Neither way seems to work. What?s strange is, I have an Adtran voice gateway at 172.16.15.11 that is sending calls to the FS box without any problems. The sipX server on the same subnet, 10.10.3.0 /25 can?t send calls without generating the Proxy Authentication Required message. Is this just because it?s on the same LAN as the FS box? 

 

Thanks,


Ken

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Tuesday, April 20, 2010 3:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

ken,

 

are you sure it's not apply-inbound-acl you want to set ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 20/04/2010 ? 22:03, Ken Fulmer a ?crit :





I?m now using an ACL list called ?lan?.

 

    <list name="lan" default="allow">

     <node type="allow" cidr="10.10.3.0/25"/>

    </list>

 

In the external sip profile, I have the following statement:

 

    <param name="local-network-acl" value="lan"/>

 

I still get the Proxy Authentication Required error. Am I doing something wrong?

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tjardick van der Kraan
Sent: Sunday, April 11, 2010 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

Use the CIDR XML key not domain:

 

http://wiki.freeswitch.org/wiki/Acl

 

Regards,

 

Tj

 

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:






Per your suggestion, I changed the following in the conf/autoload_configs/acl.conf.xml file:

 

    <list name="domains" default="deny">

      <node type="allow" domain="10.10.3.10"/>

      <node type="allow" domain="10.10.3.11"/>

    </list>

 

10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers. However, the calls still fail with the 407 Proxy Authentication Required message.

 

I get the following log output when I issue the command, reloadacl:

 

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list rfc1918.auto default (deny)

freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list wan.auto default (allow)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8 (deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12 (deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16 (deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list nat.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding 10.10.3.12/255.255.255.128 (deny) to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12 (allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16 (allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list loopback.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8 (allow) [] to list loopback.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list localnet.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding 10.10.3.12/255.255.255.128 (allow) to list localnet.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list domains default (deny)

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain 10.10.3.10

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain 10.10.3.11

 

Am I doing something incorrectly?

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, April 08, 2010 6:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

 

On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:

Actually, I did purchase a license and installed it today. One call establishes at 729. When I hang up the phone and try again, it?s 711.

Make sure that the encoder/decoder isn't still in use prior to trying the second call. After you hang up, do a "show channels" and see if the call is still "up" or not. Also, do "g729_status" to see if the encoder or decoder is in use. Keep doing "g729_status" until the 'coders are not in use. If there is a long delay then open up a JIRA ticket on jira.freeswitch.org.

 

The Proxy Authentication Required is being sent by FreeSwitch to the internal PBX. I have registration disabled on the FreeSwitch gateway and the internal server.

By default the SIP profile will challenge if the IP address of the caller is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the "domains" node. Add your PBX's IP address. You'll see an example in the comments. Once you're done editing, save the file and then go to the fs_cli and do:
reloadacl reloadxml

Then make a call from PBX to FS and it should go through.
-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 anthony.minessale at gmail.com  Tue Apr 20 14:21:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Apr 2010 16:21:00 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <019001cae0cb$5ff19050$1fd4b0f0$@com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>
	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>
	<011701cad769$714e8670$53eb9350$@com>
	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>
	<012d01cad807$27f5a0a0$77e0e1e0$@com>
	<AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>
	<015b01cae0c4$933c12c0$b9b43840$@com>
	<E687000A-27B0-4013-8240-92397B4B5481@gmail.com>
	<019001cae0cb$5ff19050$1fd4b0f0$@com>
Message-ID: <o2p191c3a031004201421i1d6449afla0455ba45cac7600@mail.gmail.com>

sometimes on sipx you need

<param name="accept-blind-auth" value="true"/>

in your profile because it tries to send auth info even when it's not
necessary and FS will enforce
auth packets even with everything else wide open.


On Tue, Apr 20, 2010 at 3:52 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  I?ve tried this setting as well:
>
>
>
>     <param name="apply-inbound-acl" value="lan"/>
>
>
>
> Neither way seems to work. What?s strange is, I have an Adtran voice
> gateway at 172.16.15.11 that is sending calls to the FS box without any
> problems. The sipX server on the same subnet, 10.10.3.0 /25 can?t send calls
> without generating the Proxy Authentication Required message. Is this just
> because it?s on the same LAN as the FS box?
>
>
>
> Thanks,
>
>
> Ken
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Ponzone
> *Sent:* Tuesday, April 20, 2010 3:25 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
> ken,
>
>
>
> are you sure it's not apply-inbound-acl you want to set ?
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
>
>
> Le 20/04/2010 ? 22:03, Ken Fulmer a ?crit :
>
>
>
>   I?m now using an ACL list called ?lan?.
>
>
>
>     <list name="lan" default="allow">
>
>      <node type="allow" cidr="10.10.3.0/25"/>
>
>     </list>
>
>
>
> In the external sip profile, I have the following statement:
>
>
>
>     <param name="local-network-acl" value="lan"/>
>
>
>
> I still get the Proxy Authentication Required error. Am I doing something
> wrong?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>
> ] *On Behalf Of *Tjardick van der Kraan
> *Sent:* Sunday, April 11, 2010 4:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
> Use the CIDR XML key not domain:
>
>
>
> http://wiki.freeswitch.org/wiki/Acl
>
>
>
> Regards,
>
>
>
> Tj
>
>
>
> On 09 Apr 2010, at 19:07, Ken Fulmer wrote:
>
>
>
>
>    Per your suggestion, I changed the following in the
> conf/autoload_configs/acl.conf.xml file:
>
>
>
>     <list name="domains" default="deny">
>
>       <node type="allow" domain="10.10.3.10"/>
>
>       <node type="allow" domain="10.10.3.11"/>
>
>     </list>
>
>
>
> 10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
> However, the calls still fail with the 407 Proxy Authentication Required
> message.
>
>
>
> I get the following log output when I issue the command, reloadacl:
>
>
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
> rfc1918.auto default (deny)
>
> freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE]
> switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list rfc1918.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
> wan.auto default (allow)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (deny) [] to list wan.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
> nat.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
> 10.10.3.12/255.255.255.128 (deny) to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8(allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 172.16.0.0/12 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding
> 192.168.0.0/16 (allow) [] to list nat.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
> loopback.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8(allow) [] to list loopback.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
> localnet.auto default (deny)
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
> 10.10.3.12/255.255.255.128 (allow) to list localnet.auto
>
> 2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
> domains default (deny)
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.10
>
> 2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate
> domain 10.10.3.11
>
>
>
> Am I doing something incorrectly?
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>
> ] *On Behalf Of *Michael Collins
> *Sent:* Thursday, April 08, 2010 6:25 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Two Major Problems
>
>
>
>
>
> On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Actually, I did purchase a license and installed it today. One call
> establishes at 729. When I hang up the phone and try again, it?s 711.
>
> Make sure that the encoder/decoder isn't still in use prior to trying the
> second call. After you hang up, do a "show channels" and see if the call is
> still "up" or not. Also, do "g729_status" to see if the encoder or decoder
> is in use. Keep doing "g729_status" until the 'coders are not in use. If
> there is a long delay then open up a JIRA ticket on jira.freeswitch.org.
>
>
>
> The Proxy Authentication Required is being sent by FreeSwitch to the
> internal PBX. I have registration disabled on the FreeSwitch gateway and the
> internal server.
>
> By default the SIP profile will challenge if the IP address of the caller
> is not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
> "domains" node. Add your PBX's IP address. You'll see an example in the
> comments. Once you're done editing, save the file and then go to the fs_cli
> and do:
> reloadacl reloadxml
>
> Then make a call from PBX to FS and it should go through.
> -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
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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

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From troy at tlainvestments.com  Tue Apr 20 14:30:53 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 20 Apr 2010 14:30:53 -0700
Subject: [Freeswitch-users] lua api_hangup_hook missing env?
Message-ID: <E4274C77-BB6F-4B79-9208-97ADEDA67956@tlainvestments.com>

The wiki (http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) refers to the env global in mod_lua when executing a lua script set by api_hangup_hook, however, in practice, I don't see env set.  Has this been changed or have I messed up somehow?  I'm using the following from within the dial plan to get it going.

<action application="set" data="api_hangup_hook=luarun recording.lua"/>
<action application="bridge" data="user/107@${domain_name}" />

Here are the global variables lua does have set:

2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: string
2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: xpcall
2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: package
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: tostring
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: print
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: os
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: unpack
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: swig_type
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: require
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: getfenv
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: setmetatable
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: next
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: freeswitch
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: assert
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: argv
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: tonumber
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: io
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawequal
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: collectgarbage
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: getmetatable
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: module
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawset
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: ipairs
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: script_name
2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: swig_equals
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: math
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: debug
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: pcall
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: table
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: newproxy
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: type
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: coroutine
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _G
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: select
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: gcinfo
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: pairs
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawget
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: loadstring
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: tellme
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _VERSION
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: dofile
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: setfenv
2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: load
2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua: error
2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua: loadfile

Thanks!
Troy
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From kenfulmer at icstechnologysolutions.com  Tue Apr 20 14:35:59 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 20 Apr 2010 16:35:59 -0500
Subject: [Freeswitch-users] Two Major Problems
In-Reply-To: <o2p191c3a031004201421i1d6449afla0455ba45cac7600@mail.gmail.com>
References: <00da01cad755$474b9c50$d5e2d4f0$@com>	<y2l87f2f3b91004081425r5eb6f070zce49053182635bf1@mail.gmail.com>	<011701cad769$714e8670$53eb9350$@com>	<y2o87f2f3b91004081624za7735169u5a0f273248b1031c@mail.gmail.com>	<012d01cad807$27f5a0a0$77e0e1e0$@com>	<AACA4B79-F44F-48BB-BAF3-88F7E787DEF4@vanderkraan.net>	<015b01cae0c4$933c12c0$b9b43840$@com>	<E687000A-27B0-4013-8240-92397B4B5481@gmail.com>	<019001cae0cb$5ff19050$1fd4b0f0$@com>
	<o2p191c3a031004201421i1d6449afla0455ba45cac7600@mail.gmail.com>
Message-ID: <01db01cae0d1$77cfe360$676faa20$@com>

Wow, that did it. Thanks!

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, April 20, 2010 4:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

sometimes on sipx you need

<param name="accept-blind-auth" value="true"/>

in your profile because it tries to send auth info even when it's not
necessary and FS will enforce
auth packets even with everything else wide open.



On Tue, Apr 20, 2010 at 3:52 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

I?ve tried this setting as well:

 

    <param name="apply-inbound-acl" value="lan"/> 

 

Neither way seems to work. What?s strange is, I have an Adtran voice gateway
at 172.16.15.11 that is sending calls to the FS box without any problems.
The sipX server on the same subnet, 10.10.3.0 /25 can?t send calls without
generating the Proxy Authentication Required message. Is this just because
it?s on the same LAN as the FS box? 

 

Thanks,


Ken

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
Ponzone
Sent: Tuesday, April 20, 2010 3:25 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

ken,

 

are you sure it's not apply-inbound-acl you want to set ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autoris?e est interdite. Tout message ?lectronique est susceptible
d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 20/04/2010 ? 22:03, Ken Fulmer a ?crit :

 

I?m now using an ACL list called ?lan?.

 

    <list name="lan" default="allow">

     <node type="allow" cidr="10.10.3.0/25"/>

    </list>

 

In the external sip profile, I have the following statement:

 

    <param name="local-network-acl" value="lan"/>

 

I still get the Proxy Authentication Required error. Am I doing something
wrong?

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tjardick
van der Kraan
Sent: Sunday, April 11, 2010 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

Use the CIDR XML key not domain:

 

http://wiki.freeswitch.org/wiki/Acl

 

Regards,

 

Tj

 

On 09 Apr 2010, at 19:07, Ken Fulmer wrote:





Per your suggestion, I changed the following in the
conf/autoload_configs/acl.conf.xml file:

 

    <list name="domains" default="deny">

      <node type="allow" domain="10.10.3.10"/>

      <node type="allow" domain="10.10.3.11"/>

    </list>

 

10.10.3.10 and 10.10.3.11 are the ip addresses of our internal servers.
However, the calls still fail with the 407 Proxy Authentication Required
message.

 

I get the following log output when I issue the command, reloadacl:

 

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:954 Created ip list
rfc1918.auto default (deny)

freeswitch at internal> 2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195
Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list rfc1918.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:962 Created ip list
wan.auto default (allow)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(deny) [] to list wan.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:970 Created ip list
nat.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:972 Adding
10.10.3.12/255.255.255.128 (deny) to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16
(allow) [] to list nat.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:981 Created ip list
loopback.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8
(allow) [] to list loopback.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:987 Created ip list
localnet.auto default (deny)

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:990 Adding
10.10.3.12/255.255.255.128 (allow) to list localnet.auto

2010-04-09 12:06:31.259954 [NOTICE] switch_core.c:1015 Created ip list
domains default (deny)

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.10

2010-04-09 12:06:31.259954 [WARNING] switch_core.c:1046 Cannot locate domain
10.10.3.11

 

Am I doing something incorrectly?

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 08, 2010 6:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

 

On Thu, Apr 8, 2010 at 3:18 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Actually, I did purchase a license and installed it today. One call
establishes at 729. When I hang up the phone and try again, it?s 711.

Make sure that the encoder/decoder isn't still in use prior to trying the
second call. After you hang up, do a "show channels" and see if the call is
still "up" or not. Also, do "g729_status" to see if the encoder or decoder
is in use. Keep doing "g729_status" until the 'coders are not in use. If
there is a long delay then open up a JIRA ticket on jira.freeswitch.org.

 

The Proxy Authentication Required is being sent by FreeSwitch to the
internal PBX. I have registration disabled on the FreeSwitch gateway and the
internal server.

By default the SIP profile will challenge if the IP address of the caller is
not in the ACL. Open conf/autoload_configs/acl.conf.xml and locate the
"domains" node. Add your PBX's IP address. You'll see an example in the
comments. Once you're done editing, save the file and then go to the fs_cli
and do:
reloadacl reloadxml

Then make a call from PBX to FS and it should go through.
-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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From ranjtech at gmail.com  Tue Apr 20 16:24:51 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 20 Apr 2010 19:24:51 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a B2BUA
Message-ID: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>

Hello list,

we have an FS server configured as a call-distributor (using
"mod_distributor") to a bank/farm of Asterisk servers. However, since the FS
is acting like a B2BUA, it's not passing the originating IP (network_addr)
from the originating switch/gateway to the Asterisk farm (as expected).
However, we need to have this information in the Asterisk servers. How can
we achieve this? Do we need to create/insert a custom SIP header when
passing on the call or set a variable or any other way?

Any help on the issue will be much appreciated

Thanks in advance,
RR
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From vhatz at kinetix.gr  Tue Apr 20 17:29:22 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Wed, 21 Apr 2010 03:29:22 +0300
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in
	ODBC	core?
In-Reply-To: <t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>	<201004191203.05924.sos@sokhapkin.dyndns.org>	<w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>	<201004191245.06964.sos@sokhapkin.dyndns.org>
	<t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
Message-ID: <4BCE46E2.1030200@kinetix.gr>

Hello all,

A quick question since the topic of lost sip dialogs was touched: in 
cases of such dialogs, I assume that FS keeps the reserved memory for 
the SIP dialog. But what happens when many lost SIP dialogs start to 
accumulate over time? Does FS have any internal time-out for too old 
dialogs to declare them dead and release the reserved memory?

Best regards,
Vlasis Hatzistavrou
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz at kinetixtele.com
http://www.kinetixtele.com



Anthony Minessale wrote:
> yes its a function of the sofia-sip project, but did you wait before 
> you stopped it?
> We have people sending non stop 200+calls a second complaining about 
> no leaks.
> So if you have some real problem it's going to be your network 
> conditions and interop with your
> providers causing an edge case with the SIP stack itself who has it's 
> own list and irc channel.
>
>
>
> On Mon, Apr 19, 2010 at 11:45 AM, Sergey Okhapkin 
> <sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>> wrote:
>
>     It's possible that some BYE or ACK could be lost, but shouldn't
>     sip dialogs be
>     destroyed automatically after a timeout?
>
>     On Monday 19 April 2010, Anthony Minessale wrote:
>     > welll you are losing sip dialogs somewhere.
>     > It's not a leak its loss of sip dialogs most likely a side effect of
>     > topology problems.
>     >
>     >
>     > On Mon, Apr 19, 2010 at 11:03 AM, Sergey Okhapkin
>     >
>     > <sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>>wrote:
>     > > The command was exactly as you suggested, FS was started with
>     "-nonat
>     > > -vg" command line options, after few hours run (about 5K calls
>     processed)
>     > > FS was shut down gracefully.
>     > >
>     > > On Monday 19 April 2010, Anthony Minessale wrote:
>     > > > This is invalid.
>     > > >
>     > > > Those places are where sip calls are created and they are most
>     > > > definitely not leaking.
>     > > > If anything, it suggests dialogs that are up still and not yet
>     > > > destroyed
>     > >
>     > > at
>     > >
>     > > > the termination of the program.
>     > > > If this location in the code was a leak it would be gigs not
>     megs
>     > >
>     > > missing.
>     > >
>     > > > Did you do the exact command I said (especially -vg param to
>     FS) and
>     > > > then
>     > >
>     > > a
>     > >
>     > > > full clean shutdown all the way back to the shell?
>     > > >
>     > > > Are you maybe not getting the BYE to your calls creating
>     open dialogs?
>     > > > You may want to turn on the sip trace.
>     > > >
>     > > >
>     > > > On Mon, Apr 19, 2010 at 10:29 AM, Sergey Okhapkin
>     > > >
>     > > > <sos at sokhapkin.dyndns.org
>     <mailto:sos at sokhapkin.dyndns.org>>wrote:
>     > > > > Valgrind doesn't show leaks in mod_nibblebill, most of the
>     leaked
>     > >
>     > > memory
>     > >
>     > > > > was
>     > > > > in mod_sofia:
>     > > > >
>     > > > > ==12476== 53,064 bytes in 67 blocks are definitely lost in
>     loss
>     > > > > record 138 of
>     > > > > 141
>     > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>     > > > > linux/vgpreload_memcheck.so)
>     > > > > ==12476==    by 0x6A0DC8F: sofia_handle_sip_i_invite
>     (sofia.c:6538)
>     > > > > ==12476==
>     > > > > ==12476==
>     > > > > ==12476== 92,664 bytes in 117 blocks are definitely lost
>     in loss
>     > > > > record 139 of
>     > > > > 141
>     > > > > ==12476==    at 0x4026378: malloc (in /usr/lib/valgrind/x86-
>     > > > > linux/vgpreload_memcheck.so)
>     > > > > ==12476==    by 0x6A2F76F: sofia_glue_do_invite
>     (sofia_glue.c:1874)
>     > > > > ==12476==
>     > > > >
>     > > > > BTW, do I understand FS code correct that channel private
>     data are
>     > >
>     > > always
>     > >
>     > > > > handled by sqlite, even if I enable core ODBC?
>     > > > >
>     > > > > On Monday 19 April 2010, Anthony Minessale wrote:
>     > > > > > 300M is tiny. try having 2000 channels up you can soar
>     into a gig
>     > > > > > of
>     > > > >
>     > > > > usage.
>     > > > >
>     > > > > > That valgrind report only showed 100k leaked which means
>     it was all
>     > > > > > accounted for and torn down in the end.
>     > > > > >
>     > > > > > If you can find a specific leak in the nibble bill
>     module with
>     > >
>     > > valgrind
>     > >
>     > > > > we
>     > > > >
>     > > > > > will gladly fix it and you
>     > > > > > can gladly thank us for giving you a way to make money
>     by the
>     > > > > > second.
>     > > > > >
>     > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
>     > > > > > --leak-resolution=high --show-reachable=yes
>     > > > > > /usr/local/freeswitch/bin/freeswitch -vg
>     > > > > >
>     > > > > >
>     > > > > >
>     > > > > > On Sun, Apr 18, 2010 at 11:48 AM, Sergey Okhapkin
>     > > > > >
>     > > > > > <sos at sokhapkin.dyndns.org
>     <mailto:sos at sokhapkin.dyndns.org>>wrote:
>     > > > > > > Which valgrind options should I specify? I did run
>     > > > > > >
>     > > > > > > valgrind --tool=memcheck --log-file=vg.log
>     --leak-check=full
>     > >
>     > > --leak-
>     > >
>     > > > > > > resolution=high --show-reachable=yes ./freeswitch
>     -nonat -vg
>     > > > > > >
>     > > > > > > 100K leaked is nothing compared to FS process RSS of
>     300M...
>     > > > > > >
>     > > > > > > On Sunday 18 April 2010, Anthony Minessale wrote:
>     > > > > > > > Also thatb suggests you leaked a whopping 100k
>     > > > > > > >
>     > > > > > > > On Apr 18, 2010 11:35 AM, "Anthony Minessale" <
>     > > > > > >
>     > > > > > > anthony.minessale at gmail.com
>     <mailto:anthony.minessale at gmail.com>>
>     > > > > > >
>     > > > > > > > wrote:
>     > > > > > > >
>     > > > > > > > That's not very useful you need a full report with
>     extended
>     > > > > > > > checking.
>     > > > > > > >
>     > > > > > > > > On Apr 18, 2010 7:34 AM, "Sergey Okhapkin" <
>     > > > >
>     > > > > sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>>
>     > > > >
>     > > > > > > > wrote:
>     > > > > > > > > According to v...
>     > > > > > > > >
>     > > > > > > > >
>     > > > > > > > >
>     > > > > > > > >
>     > > > > > > > >
>     > > > > > > > > On Saturday 17 April 2010, Michael Jerris wrote:
>     > > > > > > > > That is a completely incorrect statement. Usi...
>     > > > > > >
>     > > > > > > _______________________________________________
>     > > > > > > 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
>     <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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing 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 anthony.minessale at gmail.com  Tue Apr 20 17:53:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Apr 2010 19:53:08 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <4BCE46E2.1030200@kinetix.gr>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201004191203.05924.sos@sokhapkin.dyndns.org>
	<w2k191c3a031004190931ndaa1201bif7186bc1c99df523@mail.gmail.com>
	<201004191245.06964.sos@sokhapkin.dyndns.org>
	<t2r191c3a031004190957t56f61fadvc51e02c7f0310f25@mail.gmail.com>
	<4BCE46E2.1030200@kinetix.gr>
Message-ID: <l2y191c3a031004201753r79218a33p4e562b272082d126@mail.gmail.com>

Dialogs are managed by the sofia sip stack.
Its a separate project and a seperate library.
But it more or less follows the rfc and keeps them for the allotted time and
then destroys them unless there is a bug. One such bug exists when enabling
100rel in some specific edge cases for instance.....

On Apr 20, 2010 7:35 PM, "Vlasis Hatzistavrou (KTI)" <vhatz at kinetix.gr>
wrote:

 Hello all,

A quick question since the topic of lost sip dialogs was touched: in cases
of such dialogs, I assume that FS keeps the reserved memory for the SIP
dialog. But what happens when many lost SIP dialogs start to accumulate over
time? Does FS have any internal time-out for too old dialogs to declare them
dead and release the reserved memory?

Best regards,
Vlasis Hatzistavrou
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz at kinetixtele.comhttp://www.kinetixtele.com



Anthony Minessale wrote:


>
> yes its a function of the sofia-sip project, but did you wait before you
stopped it?
> We have ...

------------------------------


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-user...


_______________________________________________
FreeSWITCH-users mailing 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 david.ponzone at gmail.com  Tue Apr 20 18:16:53 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 21 Apr 2010 03:16:53 +0200
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
Message-ID: <A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>

Setting a customer SIP header seems a nice way to do that. And easy.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/04/2010 ? 01:24, RR a ?crit :

> Hello list,
>
> we have an FS server configured as a call-distributor (using  
> "mod_distributor") to a bank/farm of Asterisk servers. However,  
> since the FS is acting like a B2BUA, it's not passing the  
> originating IP (network_addr) from the originating switch/gateway to  
> the Asterisk farm (as expected). However, we need to have this  
> information in the Asterisk servers. How can we achieve this? Do we  
> need to create/insert a custom SIP header when passing on the call  
> or set a variable or any other way?
>
> Any help on the issue will be much appreciated
>
> Thanks in advance,
> RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From elihayun at gmail.com  Tue Apr 20 23:54:52 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Wed, 21 Apr 2010 09:54:52 +0300
Subject: [Freeswitch-users] where file "xmlrpc.inc" gone?
Message-ID: <4BCEA13C.5060707@gmail.com>

Hi

I am trying to add some xmlprc functionality to FS. I want to use it
with PHP. The php example has the following line:


include("xmlrpc.inc");

I noticed that in version 1.0.5-xx the file included but not with 1.0.6

Is there other way to use it with php or the file is wrongly missing ?

Thanks

Eli





From troy at tlainvestments.com  Tue Apr 20 23:58:09 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 20 Apr 2010 23:58:09 -0700
Subject: [Freeswitch-users] Responding to bridge after rejected attended
	transfer
Message-ID: <1B03EB8E-8F82-4688-9FDA-B296BFD0FEC7@tlainvestments.com>

I have the latest version of FS (90913b8e26265fd381318334f40e0b1a038bb066 committed Apr 21) and am using the default config with a small change that allows me to respond to the various values originate_disposition values.

First, is there a better way than what I am doing?  Everything seems to work fine until I try it with an attended transfer.  I've experienced this with Polycom, Cisco, and a soft phone client, so I don't think it's the phones, but who knows?  With 3 extensions, say 1001, 1002, 1003, each on a different phone, I can call from 1001 to 1002.  1002 initiates a transfer to 1003.  1003 rejects so 1002 starts hearing voicemail.  1002 hits transfer again to "connect" 1001 to 1003's voicemail.  With the default FS dialplan, it starts 1003's voicemail over, which is good.  With this slight modification, it hangs up on 1001.

If the feedback I get is that this modification looks fine, I can open a jira and supply a SIP trace and fs logs for the calls.

Incidentally, to keep it simple, this example does the same thing regardless of the originate_disposition.  In practice, there would be different actions taken.

Thanks for any help!

-Troy

Here's what I have:

<extension name="Local_Extension">
  <condition field="destination_number" expression="^(10[01]?[0-9])$">
    <action application="set" data="dialed_extension=$1"/>
    <action application="export" data="dialed_extension=$1"/>
    <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
    <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
    <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="transfer_ringback=$${hold_music}"/>
    <action application="set" data="call_timeout=30"/>
    <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
    <action application="set" data="hangup_after_bridge=true"/>
    <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
    <action application="set" data="continue_on_fail=true"/>
    <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
    <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
    <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
    <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
    <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
<!--    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> -->
    <action application="transfer" data="BRIDGE-FOLLOWUP"/>
  </condition>
</extension>

<!-- CHANGES TO DEFAULT DIALPLAN HERE  -->
<extension name="BRIDGE-FOLLOWUP">
  <condition field="destination_number" expression="BRIDGE-FOLLOWUP"/>
  <condition field="${originate_disposition}" expression="^SUCCESS$" break="on-true">
    <action application="log" data="ERR (${originate_disposition})successful bridge is now done for ${dialed_extension}"/>
    <action application="hangup" data="NORMAL_CLEARING"/>
  </condition>
  <condition field="${originate_disposition}" expression="USER_BUSY|CALL_REJECTED" break="on-true">
    <action application="log" data="ERR (${originate_disposition})busy/rejected for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
  <condition field="${originate_disposition}" expression="NO_ANSWER" break="on-true">
    <action application="log" data="ERR (${originate_disposition})no answer for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
  <condition>
    <action application="log" data="ERR (${originate_disposition})other error for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
</extension>

I also tried:

<extension name="Local_Extension">
  <condition field="destination_number" expression="^(10[01]?[0-9])$">
    <action application="set" data="dialed_extension=$1"/>
    <action application="export" data="dialed_extension=$1"/>
    <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
    <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
    <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="transfer_ringback=$${hold_music}"/>
    <action application="set" data="call_timeout=30"/>
    <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
    <action application="set" data="hangup_after_bridge=true"/>
    <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
    <action application="set" data="continue_on_fail=true"/>
    <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
    <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
    <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
    <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
    <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
<!--    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> -->
  </condition>

<!-- CHANGES TO DEFAULT DIALPLAN HERE  -->
  <condition field="${originate_disposition}" expression="^SUCCESS$" break="on-true">
    <action application="log" data="ERR (${originate_disposition})successful bridge is now done for ${dialed_extension}"/>
    <action application="hangup" data="NORMAL_CLEARING"/>
  </condition>
  <condition field="${originate_disposition}" expression="^USER_BUSY|CALL_REJECTED$" break="on-true">
    <action application="log" data="ERR busy/rejected for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
  <condition field="${originate_disposition}" expression="^NO_ANSWER$" break="on-true">
    <action application="log" data="ERR no answer for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
  <condition>
    <action application="log" data="ERR other error for ${dialed_extension}"/>
    <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
  </condition>
</extension>

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From Prometheus001 at gmx.net  Wed Apr 21 03:14:31 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 21 Apr 2010 12:14:31 +0200
Subject: [Freeswitch-users] effective_caller_id_name not working while
 calling multiple endpoints with :_:
Message-ID: <4BCED007.80900@gmx.net>

Setting the effective_caller_id_name when dialing multiple endpoints
with :_: do not seem to work.
See example:
  <action application="set" data="effective_caller_id_name=MyName"/>
  <action application="bridge" 
 
data="user/30 at mydomain.com:_:user/31 at mydomain.com:_:user/32 at mydomain.com:_:user/33 at mydomain.com:_:user/34 at mydomain.com"/>

Freeswitch tries to set it:
   EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060
set(effective_caller_id_name=MyName)
   2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060 SET
[effective_caller_id_name]=[MyName]

But the SIP messages do not contain the effective_caller_id_name.

If we change the ":_:" sperator to "," then the effective_caller_id_name
is correctly submittted (hower I cannot call multiple-registrations on
one number then).

We are on
  FreeSWITCH Version 1.0.head (svn-17188)

Any ideas how to overcome this? Or shall I open a JIRA?

Best regards
Peter

See example SIP message:

U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
Max-Forwards: 70.
From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
CSeq: 129803060 INVITE.
Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 920.
X-FS-Support: update_display.
Remote-Party-ID: "069xxxxxxxx"
<sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
s=FreeSWITCH.
c=IN IP4 10.xx.xx.141.
t=0 0.
m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
101 13.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:96 AMR/8000.
a=fmtp:96 octet-align=0.
a=rtpmap:99 SPEEX/32000.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:7 LPC/8000.
a=rtpmap:124 G726-16/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:6 DVI4/16000.
a=rtpmap:123 G726-24/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:10 L16/22050.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode
#




From Prometheus001 at gmx.net  Wed Apr 21 03:23:27 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 21 Apr 2010 12:23:27 +0200
Subject: [Freeswitch-users] Code negotiation - all codec offered
Message-ID: <4BCED21F.7080004@gmx.net>

Hello

since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)"
we see that - despite only allowing PMCA and PCMU codecs in vars.xml - 
Freeswitch offers all available codec to the called phone.

Any idea how to solve this?


Best regards
Peter

See Invite  example

U 10.xx.xx.1:5060 -> 10.xx.xx.172:2048
INVITE sip:31 at 10.xx.xx.172:2048;line=hxbudrul SIP/2.0.
Via: SIP/2.0/UDP 10.xx.xx.1;rport;branch=z9hG4bKZD66c84339SHH.
Max-Forwards: 70.
From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.1>;tag=5evr6508K9S3K.
To: <sip:31 at 10.xx.xx.172:2048;line=hxbudrul>.
Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
CSeq: 129803060 INVITE.
Contact: <sip:mod_sofia at 10.xx.xx.1:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 920.
X-FS-Support: update_display.
Remote-Party-ID: "069xxxxxxxx"
<sip:069xxxxxxxx at 10.xx.xx.1>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.1.
s=FreeSWITCH.
c=IN IP4 10.xx.xx.1.
t=0 0.
m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
101 13.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:96 AMR/8000.
a=fmtp:96 octet-align=0.
a=rtpmap:99 SPEEX/32000.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:7 LPC/8000.
a=rtpmap:124 G726-16/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:6 DVI4/16000.
a=rtpmap:123 G726-24/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:10 L16/22050.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode
#


From david.ponzone at gmail.com  Wed Apr 21 03:27:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 21 Apr 2010 12:27:16 +0200
Subject: [Freeswitch-users] effective_caller_id_name not working while
	calling multiple endpoints with :_:
In-Reply-To: <4BCED007.80900@gmx.net>
References: <4BCED007.80900@gmx.net>
Message-ID: <66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>

I think you should first thing update to latest GIT :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :

> Setting the effective_caller_id_name when dialing multiple endpoints
> with :_: do not seem to work.
> See example:
>  <action application="set" data="effective_caller_id_name=MyName"/>
>  <action application="bridge"
>
> data="user/30 at mydomain.com:_:user/31 at mydomain.com:_:user/32 at mydomain.com 
> :_:user/33 at mydomain.com:_:user/34 at mydomain.com"/>
>
> Freeswitch tries to set it:
>   EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060
> set(effective_caller_id_name=MyName)
>   2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
> sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060 SET
> [effective_caller_id_name]=[MyName]
>
> But the SIP messages do not contain the effective_caller_id_name.
>
> If we change the ":_:" sperator to "," then the  
> effective_caller_id_name
> is correctly submittted (hower I cannot call multiple-registrations on
> one number then).
>
> We are on
>  FreeSWITCH Version 1.0.head (svn-17188)
>
> Any ideas how to overcome this? Or shall I open a JIRA?
>
> Best regards
> Peter
>
> See example SIP message:
>
> U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
> INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
> Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
> Max-Forwards: 70.
> From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
> To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
> Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> CSeq: 129803060 INVITE.
> Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 920.
> X-FS-Support: update_display.
> Remote-Party-ID: "069xxxxxxxx"
> <sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
> s=FreeSWITCH.
> c=IN IP4 10.xx.xx.141.
> t=0 0.
> m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
> 101 13.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:96 AMR/8000.
> a=fmtp:96 octet-align=0.
> a=rtpmap:99 SPEEX/32000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:7 LPC/8000.
> a=rtpmap:124 G726-16/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:6 DVI4/16000.
> a=rtpmap:123 G726-24/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:10 L16/22050.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode
> #
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Wed Apr 21 04:17:30 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 21 Apr 2010 12:17:30 +0100
Subject: [Freeswitch-users] Code negotiation - all codec offered
In-Reply-To: <4BCED21F.7080004@gmx.net>
References: <4BCED21F.7080004@gmx.net>
Message-ID: <q2seafa772f1004210417xbaf41ac0ye5f305635c55ce3d@mail.gmail.com>

What do you have defined in the SIP profile for the following parameters?

outbound-codec-prefs
codec-prefs

You should have either outbound-codec-prefs or codec-prefs set to the
variable you set in vars.xml, you can also put the codec choices into
the SIP profile directly if you wish.

Also, just to check it's correct, what do you have defined for the
codecs in the vars.xml file?

-Steve


On 21 April 2010 11:23, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello
>
> since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)"
> we see that - despite only allowing PMCA and PCMU codecs in vars.xml -
> Freeswitch offers all available codec to the called phone.
>
> Any idea how to solve this?
>
>
> Best regards
> Peter
>
> See Invite ?example
>
> U 10.xx.xx.1:5060 -> 10.xx.xx.172:2048
> INVITE sip:31 at 10.xx.xx.172:2048;line=hxbudrul SIP/2.0.
> Via: SIP/2.0/UDP 10.xx.xx.1;rport;branch=z9hG4bKZD66c84339SHH.
> Max-Forwards: 70.
> From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.1>;tag=5evr6508K9S3K.
> To: <sip:31 at 10.xx.xx.172:2048;line=hxbudrul>.
> Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> CSeq: 129803060 INVITE.
> Contact: <sip:mod_sofia at 10.xx.xx.1:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 920.
> X-FS-Support: update_display.
> Remote-Party-ID: "069xxxxxxxx"
> <sip:069xxxxxxxx at 10.xx.xx.1>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.1.
> s=FreeSWITCH.
> c=IN IP4 10.xx.xx.1.
> t=0 0.
> m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
> 101 13.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:96 AMR/8000.
> a=fmtp:96 octet-align=0.
> a=rtpmap:99 SPEEX/32000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:7 LPC/8000.
> a=rtpmap:124 G726-16/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:6 DVI4/16000.
> a=rtpmap:123 G726-24/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:10 L16/22050.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode
> #
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Wed Apr 21 04:23:13 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 21 Apr 2010 13:23:13 +0200
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>

Actually, I just noticed the same behaviour today in my lab system, when dialing from a Polycom phone. Though in this case I have "apply-inbound-acl" set to allow access everything from my whole network (192.168.94.0/24), could this cause the call fr?n Polycom ext not being treated as a "authenticated" call, and not apply the user_context to it?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Phillip Jones
Skickat: den 19 april 2010 19:56
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] SIP Agent (Android SIP client)

Hi there,

Has anyone got the android app SIP Agent working with FreeSWITCH.

I have the app saying it is registered correctly, but when I make a call it is in the public context and not default.

I just wondered whether anyone has had any success.

thanks


pj


(PS: I have sipDroid working very nicely and iSIP on the iPhone also)


!DSPAM:4bcc9ac332931158613844!
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From jaybinks at gmail.com  Wed Apr 21 04:54:41 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 21 Apr 2010 21:54:41 +1000
Subject: [Freeswitch-users] tiny bug in bootstrap
Message-ID: <o2pec2ce17e1004210454o1f6bb258z3e7da846ffdf12c8@mail.gmail.com>

just downloaded latest from GIT..
ran bootstrap and got this :

*./bootstrap.sh
bootstrap: checking installation...
bootstrap: autoconf not found.
           You need autoconf version 2.59 or newer installed
           to build FreeSWITCH from SVN.*



might need to change that so its saying SVN anymore..


-- 
Sincerely

Jay
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From steve.d.ward at gmail.com  Wed Apr 21 05:31:25 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Wed, 21 Apr 2010 08:31:25 -0400
Subject: [Freeswitch-users] lua api_hangup_hook missing env?
In-Reply-To: <E4274C77-BB6F-4B79-9208-97ADEDA67956@tlainvestments.com>
References: <E4274C77-BB6F-4B79-9208-97ADEDA67956@tlainvestments.com>
Message-ID: <g2m4ea6e8f21004210531w5f67d6fva20029ba1c8b8386@mail.gmail.com>

I'm invoking a lua script via api_hangup_hook and I can read the env object
just fine.

Did you try setting api_hangup_hook=*lua* recording.lua instead of luarun?

The example dialplan at *
http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object*<http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object>uses
*lua *instead of *luarun* as well.

There is a brief statement on luarun vs. lua on the wiki - perhaps when
luarun spawns its thread to run the script it doesn't get the env object?
Just a guess...
http://wiki.freeswitch.org/wiki/Mod_lua#luarun_at_the_CLI



On Tue, Apr 20, 2010 at 5:30 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> The wiki (http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object)
> refers to the env global in mod_lua when executing a lua script set by
> api_hangup_hook, however, in practice, I don't see env set.  Has this been
> changed or have I messed up somehow?  I'm using the following from within
> the dial plan to get it going.
>
> <action application="set" data="api_hangup_hook=luarun recording.lua"/>
> <action application="bridge" data="user/107@${domain_name}" />
>
> *Here are the global variables lua does have set:*
>
> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> string
> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> xpcall
> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> package
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> tostring
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> print
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: os
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> unpack
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> swig_type
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> require
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> getfenv
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> setmetatable
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: next
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> freeswitch
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> assert
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: argv
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> tonumber
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: io
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> rawequal
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> collectgarbage
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> getmetatable
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> module
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> rawset
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> ipairs
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> script_name
> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> swig_equals
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: math
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> debug
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> pcall
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> table
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> newproxy
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: type
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> coroutine
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _G
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> select
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> gcinfo
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> pairs
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> rawget
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> loadstring
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> tellme
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> _VERSION
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> dofile
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> setfenv
> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: load
> 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> error
> 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua:
> loadfile
>
> Thanks!
> Troy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 21 05:57:19 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 07:57:19 -0500
Subject: [Freeswitch-users] Responding to bridge after rejected attended
	transfer
In-Reply-To: <1B03EB8E-8F82-4688-9FDA-B296BFD0FEC7@tlainvestments.com>
References: <1B03EB8E-8F82-4688-9FDA-B296BFD0FEC7@tlainvestments.com>
Message-ID: <01FC915A-16BB-4EE1-BD97-83F81DD8E423@freeswitch.org>

You don't have CALL_REJECTED in your continue_on_fail list.

/b

On Apr 21, 2010, at 1:58 AM, Troy Anderson wrote:

> I have the latest version of FS (90913b8e26265fd381318334f40e0b1a038bb066 committed Apr 21) and am using the default config with a small change that allows me to respond to the various values originate_disposition values.
> 
> First, is there a better way than what I am doing?  Everything seems to work fine until I try it with an attended transfer.  I've experienced this with Polycom, Cisco, and a soft phone client, so I don't think it's the phones, but who knows?  With 3 extensions, say 1001, 1002, 1003, each on a different phone, I can call from 1001 to 1002.  1002 initiates a transfer to 1003.  1003 rejects so 1002 starts hearing voicemail.  1002 hits transfer again to "connect" 1001 to 1003's voicemail.  With the default FS dialplan, it starts 1003's voicemail over, which is good.  With this slight modification, it hangs up on 1001.
> 
> If the feedback I get is that this modification looks fine, I can open a jira and supply a SIP trace and fs logs for the calls.
> 
> Incidentally, to keep it simple, this example does the same thing regardless of the originate_disposition.  In practice, there would be different actions taken.
> 
> Thanks for any help!
> 
> -Troy



From brian at freeswitch.org  Wed Apr 21 05:59:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 07:59:55 -0500
Subject: [Freeswitch-users] Code negotiation - all codec offered
In-Reply-To: <4BCED21F.7080004@gmx.net>
References: <4BCED21F.7080004@gmx.net>
Message-ID: <9801FA3A-5289-486B-9EC1-DDF9BB81ACAB@freeswitch.org>

codec-prefs will define both.

inbound-codec-prefs
outbound-codec-prefs 

were added.

/b

On Apr 21, 2010, at 5:23 AM, Peter P GMX wrote:

> Hello
> 
> since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)"
> we see that - despite only allowing PMCA and PCMU codecs in vars.xml - 
> Freeswitch offers all available codec to the called phone.
> 
> Any idea how to solve this?
> 
> 
> Best regards
> Peter



From brian at freeswitch.org  Wed Apr 21 06:00:58 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 08:00:58 -0500
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
Message-ID: <A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>

Correct.

/b

On Apr 21, 2010, at 6:23 AM, Peter Olsson wrote:

> Actually, I just noticed the same behaviour today in my lab system, when dialing from a Polycom phone. Though in this case I have ?apply-inbound-acl? set to allow access everything from my whole network (192.168.94.0/24), could this cause the call fr?n Polycom ext not being treated as a ?authenticated? call, and not apply the user_context to it?
>  
> /Peter

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From peter.olsson at visionutveckling.se  Wed Apr 21 06:15:40 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 21 Apr 2010 15:15:40 +0200
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>

Thanks,

So the a way around this would be to;

1.       Exclude the phones from the acl list.

2.       Or - use another sip profile for "open" trunks, and disable the inbound-acl on the internal profile?

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 21 april 2010 15:01
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] SIP Agent (Android SIP client)

Correct.

/b

On Apr 21, 2010, at 6:23 AM, Peter Olsson wrote:


Actually, I just noticed the same behaviour today in my lab system, when dialing from a Polycom phone. Though in this case I have "apply-inbound-acl" set to allow access everything from my whole network (192.168.94.0/24), could this cause the call fr?n Polycom ext not being treated as a "authenticated" call, and not apply the user_context to it?

/Peter

!DSPAM:4bcef85e32936933977704!
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From abid_freeswitch at live.com  Wed Apr 21 06:23:56 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Wed, 21 Apr 2010 19:23:56 +0600
Subject: [Freeswitch-users] Radius/Authentication/Authorization
Message-ID: <COL109-W16741AB7D63EC18BB1996DF4090@phx.gbl>


Dear Tihomir,
It has been so many days since I did not get any response from your side. Could you please help me in this. Did you update the wiki for mod_rad_auth. If not, please help me in this regard. Thanks.
Abid Saleem

From: abid_freeswitch at live.com
To: tculjaga at gmail.com; freeswitch-users at lists.freeswitch.org
CC: neal at wanlink.com
Subject: RE: [Freeswitch-users] Radius/Authentication/Authorization
Date: Sat, 3 Apr 2010 12:02:55 +0500









Dear Tihomir,
Any update and help on the below please.
Abid
From: abid_freeswitch at live.com
To: tculjaga at gmail.com
CC: neal at wanlink.com
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization
Date: Wed, 31 Mar 2010 19:35:32 +0500








Dear Tihomir,
Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps.
Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows.
2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str**
Please advise. Thanks.
Regards-----------Abid Saleem
--Forwarded Message Attachment--
From: tculjaga at gmail.com
CC: neal at wanlink.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 30 Mar 2010 21:42:03 +0200
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization

hello,

here is an example



<configuration name="rad_auth.conf" description="radius authentification module">
  <settings>
     <!-- backward compatibility to allow radiusclient config file instead of an embedded config -->

     <!--  <param name="radius_config" value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->
  </settings>

  <client>
    <param name="authserver" value="10.1.1.10:1812:gateway"/>

    <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>
    <param name="seqfile" value="/var/run/radius.seq"/>
    <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>

    <param name="default_realm" value=""/>
    <param name="radius_timeout" value="3"/>
    <param name="radius_retries" value="2"/>
    <param name="radius_deadtime" value="0"/>

    <param name="bindaddr" value="*"/>
  </client>

  <vsas>
    <!-- 
         name:       just a description
         value:      direct input or variable 
         pec:        vendor ID (0 for default, 9 for cisco...)

         expr:       1 for channel variable, 2 for direct input
         direction:  in for radius-request, out for radius-response
    -->

    <!-- mappings for radius request message; input attributes -->

    <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/>
    <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/>

    <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/>
    <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/>

    <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/>
    <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/>

    <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>

    <!-- mappings for radius-response message; output values from returning outributes -->

    <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/>
    <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/>

    <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/>
    <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/>

    <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/>
    <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/>

    <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>
    <!-- expr param is to be ignored here-->

  </vsas>
</configuration>




in the dialplan you need to trigger auth as:

      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



there are two behaviours:

1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number
2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance







  <extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

<!--      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>  -->

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


  <extension name="unitest_rad-ANI-balance">
    <condition field="destination_number" expression="^602$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


will update the wiki by the end of the week.


you have enough information for now.

Tihomir.



On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch at live.com> wrote:






Hi Neal and other Contributors to FS,
I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration.

If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation.

Regards----------Abid SaleemSr. Product Manager
 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now. 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now. 		 	   		  
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From janvb at live.com  Wed Apr 21 06:30:47 2010
From: janvb at live.com (Jan Berger)
Date: Wed, 21 Apr 2010 15:30:47 +0200
Subject: [Freeswitch-users] Java from Lua
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>,
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
Message-ID: <SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>



hi folks,

 

I am working on a IVR demo and need to call some Java code from within a Lua. Basically i just want to run a Java app and pass some parameters forth and back.

 

I assume this might be easier using JavaScript or Java itself ???

 

Jan
 		 	   		  
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From anthony.minessale at gmail.com  Wed Apr 21 06:39:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 08:39:26 -0500
Subject: [Freeswitch-users] effective_caller_id_name not working while
	calling multiple endpoints with :_:
In-Reply-To: <66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>
References: <4BCED007.80900@gmx.net>
	<66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>
Message-ID: <l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>

when using enterprise_originate you must use the special leading <> brackets
to insert global variables meant for each tier 1 originate

<global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/
foo at bar.com,sofia/internal/foo2 at bar.com:_:sofia/internal/foo3 at bar3.com


On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I think you should first thing update to latest GIT :)
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :
>
> Setting the effective_caller_id_name when dialing multiple endpoints
> with :_: do not seem to work.
> See example:
>  <action application="set" data="effective_caller_id_name=MyName"/>
>  <action application="bridge"
>
> data="user/30 at mydomain.com:_:user/31 at mydomain.com:_:user/32 at mydomain.com
> :_:user/33 at mydomain.com:_:user/34 at mydomain.com"/>
>
> Freeswitch tries to set it:
>   EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060
> set(effective_caller_id_name=MyName)
>   2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
> sofia/external/069xxxxxxxx at 10.xx.xx.1414:5060 SET
> [effective_caller_id_name]=[MyName]
>
> But the SIP messages do not contain the effective_caller_id_name.
>
> If we change the ":_:" sperator to "," then the effective_caller_id_name
> is correctly submittted (hower I cannot call multiple-registrations on
> one number then).
>
> We are on
>  FreeSWITCH Version 1.0.head (svn-17188)
>
> Any ideas how to overcome this? Or shall I open a JIRA?
>
> Best regards
> Peter
>
> See example SIP message:
>
> U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
> INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
> Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
> Max-Forwards: 70.
> From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
> To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
> Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> CSeq: 129803060 INVITE.
> Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 920.
> X-FS-Support: update_display.
> Remote-Party-ID: "069xxxxxxxx"
> <sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
> s=FreeSWITCH.
> c=IN IP4 10.xx.xx.141.
> t=0 0.
> m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
> 101 13.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:96 AMR/8000.
> a=fmtp:96 octet-align=0.
> a=rtpmap:99 SPEEX/32000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:7 LPC/8000.
> a=rtpmap:124 G726-16/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:6 DVI4/16000.
> a=rtpmap:123 G726-24/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:10 L16/22050.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode
> #
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
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From anthony.minessale at gmail.com  Wed Apr 21 06:42:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 08:42:52 -0500
Subject: [Freeswitch-users] tiny bug in bootstrap
In-Reply-To: <o2pec2ce17e1004210454o1f6bb258z3e7da846ffdf12c8@mail.gmail.com>
References: <o2pec2ce17e1004210454o1f6bb258z3e7da846ffdf12c8@mail.gmail.com>
Message-ID: <x2z191c3a031004210642o87248908se2d1d38be120a4fd@mail.gmail.com>

got it thx

On Wed, Apr 21, 2010 at 6:54 AM, jay binks <jaybinks at gmail.com> wrote:

> just downloaded latest from GIT..
> ran bootstrap and got this :
>
> *./bootstrap.sh
> bootstrap: checking installation...
> bootstrap: autoconf not found.
>            You need autoconf version 2.59 or newer installed
>            to build FreeSWITCH from SVN.*
>
>
>
> might need to change that so its saying SVN anymore..
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
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pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 21 06:43:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 08:43:50 -0500
Subject: [Freeswitch-users] Java from Lua
In-Reply-To: <SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
	<SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>
Message-ID: <m2i191c3a031004210643h20d51b88o2d3b11401d5e6711@mail.gmail.com>

is the java app run from the shell?

On Wed, Apr 21, 2010 at 8:30 AM, Jan Berger <janvb at live.com> wrote:

>
> hi folks,
>
> I am working on a IVR demo and need to call some Java code from within a
> Lua. Basically i just want to run a Java app and pass some parameters forth
> and back.
>
> I assume this might be easier using JavaScript or Java itself ???
>
> Jan
>
> ------------------------------
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 21 06:46:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 08:46:09 -0500
Subject: [Freeswitch-users] lua api_hangup_hook missing env?
In-Reply-To: <g2m4ea6e8f21004210531w5f67d6fva20029ba1c8b8386@mail.gmail.com>
References: <E4274C77-BB6F-4B79-9208-97ADEDA67956@tlainvestments.com>
	<g2m4ea6e8f21004210531w5f67d6fva20029ba1c8b8386@mail.gmail.com>
Message-ID: <v2i191c3a031004210646i3245d153u28b1d7cadb15481d@mail.gmail.com>

yes you can't use luarun, you must use lua there.


On Wed, Apr 21, 2010 at 7:31 AM, Steven Ward <steve.d.ward at gmail.com> wrote:

> I'm invoking a lua script via api_hangup_hook and I can read the env object
> just fine.
>
> Did you try setting api_hangup_hook=*lua* recording.lua instead of luarun?
>
> The example dialplan at *
> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object*<http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object>uses
> *lua *instead of *luarun* as well.
>
> There is a brief statement on luarun vs. lua on the wiki - perhaps when
> luarun spawns its thread to run the script it doesn't get the env object?
> Just a guess...
> http://wiki.freeswitch.org/wiki/Mod_lua#luarun_at_the_CLI
>
>
>
> On Tue, Apr 20, 2010 at 5:30 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>
>> The wiki (
>> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) refers
>> to the env global in mod_lua when executing a lua script set by
>> api_hangup_hook, however, in practice, I don't see env set.  Has this been
>> changed or have I messed up somehow?  I'm using the following from within
>> the dial plan to get it going.
>>
>> <action application="set" data="api_hangup_hook=luarun recording.lua"/>
>> <action application="bridge" data="user/107@${domain_name}" />
>>
>> *Here are the global variables lua does have set:*
>>
>> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> string
>> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> xpcall
>> 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> package
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> tostring
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> print
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: os
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> unpack
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> swig_type
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> require
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> getfenv
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> setmetatable
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> next
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> freeswitch
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> assert
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> argv
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> tonumber
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: io
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> rawequal
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> collectgarbage
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> getmetatable
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> module
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> rawset
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> ipairs
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> script_name
>> 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> swig_equals
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> math
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> debug
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> pcall
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> table
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> newproxy
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> type
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> coroutine
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _G
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> select
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> gcinfo
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> pairs
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> rawget
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> loadstring
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> tellme
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> _VERSION
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> dofile
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> setfenv
>> 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> load
>> 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> error
>> 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua:
>> loadfile
>>
>> Thanks!
>> Troy
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mardy at voysys.com  Wed Apr 21 07:01:40 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Wed, 21 Apr 2010 10:01:40 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
Message-ID: <FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>

Just following up...  Does anyone have any suggestions on how to proceed with this?  I've run out of ideas.

Thanks,

-Mardy

On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:

> The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.
> 
> To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.
> 
> -Mardy
> 
> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
> 
>> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
>> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
>> 
>> 
>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
>> 
>> -Mardy
>> 
>> 
>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>> 
>>> but what is the client sending the REFER?
>>> 
>>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>>> 
>>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>>> your UA has an interop problem.
>>> 
>>> 
>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>> 
>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>> 
>>>> uuid_simplify will issue the refer...
>>> 
>>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>>> 
>>>> 
>>>> May I ask what application you are developing?
>>> 
>>> An ACD.
>>> 
>>>> 
>>>> Regards,
>>>> Jo?o Mesquita
>>>> FSComm developer
>>>> 
>>>> 
>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>>> 
>>>> -Mardy
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steve.d.ward at gmail.com  Wed Apr 21 07:09:15 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Wed, 21 Apr 2010 10:09:15 -0400
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
Message-ID: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>

Hello all,

I have a lua script running that checks the state of a call between A and B
- the call between A and B was set up through a Polycom's attended transfer
feature, so A itself didn't necessarily execute the bridge to B.

A gets early media from B.  After a certain amount of time, if A is still
getting early media, I want the script to end that call between A and B, and
send A through some specific dialplan.

How does uuid_transfer work for this goal?  I'd like to kill B and move A to
specific dialplan.  I don't want B to be in the transfer at all.

I know there may be many possibilities here; I'm just wondering if anyone
can recommend something.  Thanks.

- Steve
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From stormdemo001 at gmail.com  Tue Apr 20 04:26:22 2010
From: stormdemo001 at gmail.com (Storm Demo)
Date: Tue, 20 Apr 2010 19:26:22 +0800
Subject: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware EXSi
Message-ID: <n2gefe5e0161004200426q71077ba0qdc20c37db01ae7c9@mail.gmail.com>

Dear All,

I have installed Freesiwtch on CentOS 5.4 on VMware EXSi.

Everything works fine but I keep having voice quality issues

1. Conference call will have gradually built up delay . Up to 10s of seconds
for a conference call running for like 10 mins
2. Google talk have very choppy voice quality even I am using G711. I have
extrated the RTPs in wireshark (less that 1 % packet lost) and play it using
media player, the media is also choppy.

For guest OS clock issue. I have already set clocksource=pit in linux boot
option and NTP in linux but seems the problem still persist.

Is there any suggestion that I can get the problem solved ? Or I have to run
FS in real machine instead of visualization?

Thanks a lot for your help
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From info at evestech.com  Wed Apr 21 07:08:07 2010
From: info at evestech.com (Kashif Kahn)
Date: Wed, 21 Apr 2010 07:08:07 -0700 (PDT)
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
Message-ID: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>

Dear All,

All those who have wanted to try speech recognition with Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, I encourage you to try our Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. Of course, the engine comes with a Freeswitch connector, thereby allowing a Freeswitch user to bypass engine API and interact directly via Dialplan.

Best regards,
-Kashif

 Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615
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From anthony.minessale at gmail.com  Wed Apr 21 07:13:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 09:13:02 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
Message-ID: <j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>

instead of emailing again when impatient for an answer (something we frown
upon here in this busy list)
produce a reproducible step by step process to duplicate your issue.  We are
trying to help people but we don't have the time to do the leg work for
everyone who asks a question when we get hundreds of emails a day.



On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:

> Just following up...  Does anyone have any suggestions on how to proceed
> with this?  I've run out of ideas.
>
> Thanks,
>
> -Mardy
>
> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>
> The phones that I am using are not registered with FS.  They are registered
> with another proxy based PBX.  I am simply using FS as B2BUA which is also
> registered with the PBX.  And yes, I can successfully transfer a call to
> another phone with this setup.
>
> To simplify things I tried the same scenario using FSComm in place of my
> own FS application and tried to transfer a call to FSComm with the same
> results.  And just in case there might be a problem specific to FSComm, I
> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
> echo application with the same results.  By the way, I have no problems with
> blind transfers, only attended transfers.
>
> -Mardy
>
> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>
> did you try just setting up 2 phones on plain fresh FS install, and calling
> them normally and transferring them around?
> That description is still pretty vague? What is an Event Socket
> application, which has nothing to do with sip and sip transfers, that's a FS
> protocol.
>
>
> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> I have two phones (Polycom) and an event_socket application, all of which
>> are using a SIP proxy for call routing.  The first phone calls the second
>> phone.  The second phone then attempts to transfer the call to the
>> FS/event_socket application by first placing the call on hold and then
>> calling the FS application, followed by a consultative transfer.  The REFER
>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>  The transferred call leg appears to be answered by FS and the application
>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>> that I see is that the original call leg, created when the user called the
>> FS application to announce the transfer, does not get canceled by FS and
>> subsequently does not send the BYE back to the Polycom.  Is there something
>> that I need to do at the event_socket application to complete the transfer?
>>  I've tried killing the UUID associated with the first call leg as well as
>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>
>> -Mardy
>>
>>
>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>
>> but what is the client sending the REFER?
>>
>> FS gets refer+replaces all the time, if it's the one where the dest is on
>> another box (aka the nightmare xfer that you should see references to in the
>> debug log if so) then it will not complete until that far end call is
>> answered.
>>
>> FS handles this scenerio for us hundreds of times a day using a wide range
>> of sip devices so perhaps
>> your UA has an interop problem.
>>
>>
>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>
>>>
>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>
>>> uuid_simplify will issue the refer...
>>>
>>>
>>> I looked at uuid_simplify and if I understand it correctly it is for use
>>> when one wants to act as the transfer controller.  In my case, FS is the
>>> transfer destination.  Another phone has already generated the refer and FS
>>> has been sent an invite with replaces.
>>>
>>>
>>> May I ask what application you are developing?
>>>
>>>
>>> An ACD.
>>>
>>>
>>> Regards,
>>> Jo?o Mesquita
>>> FSComm developer
>>>
>>>
>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>
>>>> I'm having a problem with attended transfers where the destination of
>>>> the transfer is a FreeSWITCH based application such as FSComm.  (It should
>>>> be noted that in my setup the phone performing the transfer and the caller
>>>> which is being transferred are parties of another SIP server.)  What I see,
>>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>> application side, FS still thinks that both the initial call leg and
>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>> kill all legs of the call.  Is there a specific action that the application
>>>> must take in order for the transfer to complete?
>>>>
>>>> -Mardy
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From peter.olsson at visionutveckling.se  Wed Apr 21 07:18:45 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 21 Apr 2010 16:18:45 +0200
Subject: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware
 EXSi
In-Reply-To: <n2gefe5e0161004200426q71077ba0qdc20c37db01ae7c9@mail.gmail.com>
References: <n2gefe5e0161004200426q71077ba0qdc20c37db01ae7c9@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55776E77EC@cooper>

Just avoid virtualization - I think that's the only solution..

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Storm Demo
Skickat: den 20 april 2010 13:26
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware EXSi

Dear All,

I have installed Freesiwtch on CentOS 5.4 on VMware EXSi.

Everything works fine but I keep having voice quality issues

1. Conference call will have gradually built up delay . Up to 10s of seconds for a conference call running for like 10 mins
2. Google talk have very choppy voice quality even I am using G711. I have extrated the RTPs in wireshark (less that 1 % packet lost) and play it using media player, the media is also choppy.

For guest OS clock issue. I have already set clocksource=pit in linux boot option and NTP in linux but seems the problem still persist.

Is there any suggestion that I can get the problem solved ? Or I have to run FS in real machine instead of visualization?

Thanks a lot for your help

!DSPAM:4bcf090432932040348935!
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From gmaruzz at celliax.org  Wed Apr 21 07:28:47 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 21 Apr 2010 16:28:47 +0200
Subject: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware
	EXSi
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55776E77EC@cooper>
References: <n2gefe5e0161004200426q71077ba0qdc20c37db01ae7c9@mail.gmail.com> 
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E77EC@cooper>
Message-ID: <w2u7b197bef1004210728p2dcf377fid7c0aae957f9d251@mail.gmail.com>

On Wed, Apr 21, 2010 at 4:18 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Just avoid virtualization ? I think that?s the only solution..
>

And an easy one, by the way. Just quantify the amount of time you'll
use to "maybe" get it right... When virtualization will be really
ready for voip, we all we'll be aware of that ;). Don't fear you
missed that while you were in Spring holydays

-giovanni


>
>
> /Peter
>
>
>
>
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Storm Demo
> Skickat: den 20 april 2010 13:26
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware EXSi
>
>
>
> Dear All,
>
> I have installed Freesiwtch on CentOS 5.4 on VMware EXSi.
>
> Everything works fine but I keep having voice quality issues
>
> 1. Conference call will have gradually built up delay . Up to 10s of seconds
> for a conference call running for like 10 mins
> 2. Google talk have very choppy voice quality even I am using G711. I have
> extrated the RTPs in wireshark (less that 1 % packet lost) and play it using
> media player, the media is also choppy.
>
> For guest OS clock issue. I have already set clocksource=pit in linux boot
> option and NTP in linux but seems the problem still persist.
>
> Is there any suggestion that I can get the problem solved ? Or I have to run
> FS in real machine instead of visualization?
>
> Thanks a lot for your help
>
> !DSPAM:4bcf090432932040348935!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Wed Apr 21 07:33:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 09:33:51 -0500
Subject: [Freeswitch-users] Voice quality Freeswitch on Centos on VMware
	EXSi
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55776E77EC@cooper>
References: <n2gefe5e0161004200426q71077ba0qdc20c37db01ae7c9@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E77EC@cooper>
Message-ID: <k2g191c3a031004210733md04b1803l442fb97e9a6b988d@mail.gmail.com>

there is some other common vmware issue related to networking that causes
you to get 2 of every RTP packets.  if you want to use virtual I suggest
openvz as it works very well with FS.


On Wed, Apr 21, 2010 at 9:18 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Just avoid virtualization ? I think that?s the only solution..
>
>
>
> /Peter
>
>
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Storm Demo
> *Skickat:* den 20 april 2010 13:26
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* [Freeswitch-users] Voice quality Freeswitch on Centos on VMware
> EXSi
>
>
>
> Dear All,
>
> I have installed Freesiwtch on CentOS 5.4 on VMware EXSi.
>
> Everything works fine but I keep having voice quality issues
>
> 1. Conference call will have gradually built up delay . Up to 10s of
> seconds for a conference call running for like 10 mins
> 2. Google talk have very choppy voice quality even I am using G711. I have
> extrated the RTPs in wireshark (less that 1 % packet lost) and play it using
> media player, the media is also choppy.
>
> For guest OS clock issue. I have already set clocksource=pit in linux boot
> option and NTP in linux but seems the problem still persist.
>
> Is there any suggestion that I can get the problem solved ? Or I have to
> run FS in real machine instead of visualization?
>
> Thanks a lot for your help
>
> !DSPAM:4bcf090432932040348935!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From boris at tagnet.ru  Wed Apr 21 07:30:18 2010
From: boris at tagnet.ru (boris at tagnet.ru)
Date: Wed, 21 Apr 2010 20:30:18 +0600
Subject: [Freeswitch-users] mod_say or strftime problems?
Message-ID: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>

Hello!

I have an extension for the current date/time:
    <extension name="SayTime">
        <condition field="destination_number" expression="^9000$">
            <action application="answer"/>
            <action application="say" data="ru current_date_time
pronounced $strftime(%s)"/>
            <action application="sleep" data="1000"/>
            <action application="hangup"/>
        </condition>
    </extension>

The extension works fine, but time is +1 hour of current time. For example
current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6
(from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer
time in effect now, so tz=YEKST). So, my question is - something wrong
with my extension configuration or this is bug in mod_say_ru or may be
strfmt?

With respect,
Boris




From jaybinks at gmail.com  Wed Apr 21 07:39:46 2010
From: jaybinks at gmail.com (Jay Binks)
Date: Thu, 22 Apr 2010 00:39:46 +1000
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
Message-ID: <BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>

Bought the 25$ kit yesterday.

Where do I download the connector ??
Is that in fs git repo ?? Or from vestec site ??  ( if so it's not  
enabled on my account ;( )

Also you need to wiki up some mad examples...

Can't wait to play though ....

J



On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com> wrote:

> Dear All,
>
> All those who have wanted to try speech recognition with Freeswitch  
> but found the software cost too expensive or the recognition  
> accuracy unsatisfactory, I encourage you to try our Vestec Speech  
> Engine for Freeswitch at: http://www.vestec.ca/products A starter  
> kit - which is a specially priced one port (ie. one channel) license  
> for the standard engine - is available for only $25. Additional  
> ports (channels) licenses can be purchased for $99/port. Of course,  
> the engine comes with a Freeswitch connector, thereby allowing a  
> Freeswitch user to bypass engine API and interact directly via  
> Dialplan.
>
> Best regards,
> -Kashif
>
> Kashif Kahn
> VP, Business Development
> Vestec, Inc.
> Waterloo, ON Canada
> phone: (519) 885-7615
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 21 07:42:29 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 09:42:29 -0500
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
Message-ID: <2B108863-2BAD-418C-8890-B5A1F5AD6577@freeswitch.org>

Collins and I are working on examples.

/b

On Apr 21, 2010, at 9:39 AM, Jay Binks wrote:

> Bought the 25$ kit yesterday.
> 
> Where do I download the connector ??
> Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled on my account ;( )
> 
> Also you need to wiki up some mad examples...
> 
> Can't wait to play though ....  
> 
> J
> 



From info at evestech.com  Wed Apr 21 07:52:46 2010
From: info at evestech.com (Kashif Kahn)
Date: Wed, 21 Apr 2010 07:52:46 -0700 (PDT)
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
Message-ID: <949726.30191.qm@web208.biz.mail.re2.yahoo.com>

Hi Jay,

The Freeswitch connector is available for download free-of-charge from the Vestec webstore; you can see the connector download option advertized prominently on the main webstore page at: http://www.vestec.ca/products

Please contact support at vestec.ca for any questions/issues on using the engine or connector. We provide installation support free of charge.

Regards,
-Kashif



________________________________
From: Jay Binks <jaybinks at gmail.com>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Sent: Wed, April 21, 2010 10:39:46 AM
Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition


Bought the 25$ kit yesterday.

Where do I download the connector ??
Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled on my account ;( )

Also you need to wiki up some mad examples...

Can't wait to play though ....  

J



On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com> wrote:


Dear All,
>
>All those who have wanted to try speech recognition with Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, I encourage you to try our Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. Of course, the engine comes with a Freeswitch connector, thereby allowing a Freeswitch user to bypass engine API and interact directly via Dialplan.
>
>Best regards,
>-Kashif
>
> Kashif Kahn
>VP, Business Development
>Vestec,
> Inc.
>Waterloo, ON Canada
>phone: (519) 885-7615
>
>
_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.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 Apr 21 08:07:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Apr 2010 08:07:49 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call
Message-ID: <v2p87f2f3b91004210807j2fa1fedteeedbd920a4cc334@mail.gmail.com>

Hello all. Here is a reminder that the FreeSWITCH community conference call
will be starting soon. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_21

Please feel free to add your agenda items.
-MC
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From anthony.minessale at gmail.com  Wed Apr 21 08:23:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 10:23:45 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
Message-ID: <t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>

I'm trying to understand this:

If FS is acting as a phone in your scenario why are you sending a refer to
it and not the server?
In most situations there is a b2bua server who routes the calls and takes
all the REFER.
Is this one of those PROXY only sip servers?

I think you would need to produce a full debug log of this, and if you are
using some kind of proxy based setup we would need some way to easily
reproduce it or visit your lab because we do not typically use anything of
the sort.

Execute these commands and reproduce it and capture the whole log and put it
on
http://pastebin.freeswitch.org

sofia profile internal siptrace on
console loglevel debug





On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> instead of emailing again when impatient for an answer (something we frown
> upon here in this busy list)
> produce a reproducible step by step process to duplicate your issue.  We
> are trying to help people but we don't have the time to do the leg work for
> everyone who asks a question when we get hundreds of emails a day.
>
>
>
>
> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> Just following up...  Does anyone have any suggestions on how to proceed
>> with this?  I've run out of ideas.
>>
>> Thanks,
>>
>> -Mardy
>>
>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>
>> The phones that I am using are not registered with FS.  They are
>> registered with another proxy based PBX.  I am simply using FS as B2BUA
>> which is also registered with the PBX.  And yes, I can successfully transfer
>> a call to another phone with this setup.
>>
>> To simplify things I tried the same scenario using FSComm in place of my
>> own FS application and tried to transfer a call to FSComm with the same
>> results.  And just in case there might be a problem specific to FSComm, I
>> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
>> echo application with the same results.  By the way, I have no problems with
>> blind transfers, only attended transfers.
>>
>> -Mardy
>>
>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>
>> did you try just setting up 2 phones on plain fresh FS install, and
>> calling them normally and transferring them around?
>> That description is still pretty vague? What is an Event Socket
>> application, which has nothing to do with sip and sip transfers, that's a FS
>> protocol.
>>
>>
>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>
>>> I have two phones (Polycom) and an event_socket application, all of which
>>> are using a SIP proxy for call routing.  The first phone calls the second
>>> phone.  The second phone then attempts to transfer the call to the
>>> FS/event_socket application by first placing the call on hold and then
>>> calling the FS application, followed by a consultative transfer.  The REFER
>>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>>  The transferred call leg appears to be answered by FS and the application
>>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>>> that I see is that the original call leg, created when the user called the
>>> FS application to announce the transfer, does not get canceled by FS and
>>> subsequently does not send the BYE back to the Polycom.  Is there something
>>> that I need to do at the event_socket application to complete the transfer?
>>>  I've tried killing the UUID associated with the first call leg as well as
>>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>
>>> -Mardy
>>>
>>>
>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>
>>> but what is the client sending the REFER?
>>>
>>> FS gets refer+replaces all the time, if it's the one where the dest is on
>>> another box (aka the nightmare xfer that you should see references to in the
>>> debug log if so) then it will not complete until that far end call is
>>> answered.
>>>
>>> FS handles this scenerio for us hundreds of times a day using a wide
>>> range of sip devices so perhaps
>>> your UA has an interop problem.
>>>
>>>
>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>
>>>>
>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>
>>>> uuid_simplify will issue the refer...
>>>>
>>>>
>>>> I looked at uuid_simplify and if I understand it correctly it is for use
>>>> when one wants to act as the transfer controller.  In my case, FS is the
>>>> transfer destination.  Another phone has already generated the refer and FS
>>>> has been sent an invite with replaces.
>>>>
>>>>
>>>> May I ask what application you are developing?
>>>>
>>>>
>>>> An ACD.
>>>>
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>> FSComm developer
>>>>
>>>>
>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>
>>>>> I'm having a problem with attended transfers where the destination of
>>>>> the transfer is a FreeSWITCH based application such as FSComm.  (It should
>>>>> be noted that in my setup the phone performing the transfer and the caller
>>>>> which is being transferred are parties of another SIP server.)  What I see,
>>>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>>> application side, FS still thinks that both the initial call leg and
>>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>>> kill all legs of the call.  Is there a specific action that the application
>>>>> must take in order for the transfer to complete?
>>>>>
>>>>> -Mardy
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ranjtech at gmail.com  Wed Apr 21 08:34:10 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 21 Apr 2010 11:34:10 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
Message-ID: <k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>

Thanks David. We know the custom header is the way to go but didn't know how
to send that in the header. After a little research, discovered that we
could use the sip_invite_domain channel variable to forward the call with
the original "From" Header to the Asterisk Farm. Just thought it would help
someone else out in the future :)

On Tue, Apr 20, 2010 at 9:16 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Setting a customer SIP header seems a nice way to do that. And easy.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/04/2010 ? 01:24, RR a ?crit :
>
> Hello list,
>
> we have an FS server configured as a call-distributor (using
> "mod_distributor") to a bank/farm of Asterisk servers. However, since the FS
> is acting like a B2BUA, it's not passing the originating IP (network_addr)
> from the originating switch/gateway to the Asterisk farm (as expected).
> However, we need to have this information in the Asterisk servers. How can
> we achieve this? Do we need to create/insert a custom SIP header when
> passing on the call or set a variable or any other way?
>
> Any help on the issue will be much appreciated
>
> Thanks in advance,
> RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Wed Apr 21 08:43:12 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 21 Apr 2010 17:43:12 +0200
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
Message-ID: <45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>

Well, yes.
Or you use the network_addr variable, and you put it in a custom  
header X-Original-IP or whatever.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/04/2010 ? 17:34, RR a ?crit :

> Thanks David. We know the custom header is the way to go but didn't  
> know how to send that in the header. After a little research,  
> discovered that we could use the sip_invite_domain channel variable  
> to forward the call with the original "From" Header to the Asterisk  
> Farm. Just thought it would help someone else out in the future :)
>
> On Tue, Apr 20, 2010 at 9:16 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Setting a customer SIP header seems a nice way to do that. And easy.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 21/04/2010 ? 01:24, RR a ?crit :
>
>> Hello list,
>>
>> we have an FS server configured as a call-distributor (using  
>> "mod_distributor") to a bank/farm of Asterisk servers. However,  
>> since the FS is acting like a B2BUA, it's not passing the  
>> originating IP (network_addr) from the originating switch/gateway  
>> to the Asterisk farm (as expected). However, we need to have this  
>> information in the Asterisk servers. How can we achieve this? Do we  
>> need to create/insert a custom SIP header when passing on the call  
>> or set a variable or any other way?
>>
>> Any help on the issue will be much appreciated
>>
>> Thanks in advance,
>> RR
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From troy at tlainvestments.com  Wed Apr 21 08:49:00 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 21 Apr 2010 08:49:00 -0700
Subject: [Freeswitch-users] Responding to bridge after rejected attended
	transfer
In-Reply-To: <01FC915A-16BB-4EE1-BD97-83F81DD8E423@freeswitch.org>
References: <1B03EB8E-8F82-4688-9FDA-B296BFD0FEC7@tlainvestments.com>
	<01FC915A-16BB-4EE1-BD97-83F81DD8E423@freeswitch.org>
Message-ID: <4B2B7072-CB98-46BF-BE71-A5CD45DF3DE9@tlainvestments.com>

Actually, continue_on_fail is set to true - the line you saw with the list of failure codes was commented out.

> 	<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
> 	<action application="set" data="continue_on_fail=true"/>


On Apr 21, 2010, at 5:57 AM, Brian West wrote:

> You don't have CALL_REJECTED in your continue_on_fail list.
> 
> /b
> 
> On Apr 21, 2010, at 1:58 AM, Troy Anderson wrote:
> 
>> I have the latest version of FS (90913b8e26265fd381318334f40e0b1a038bb066 committed Apr 21) and am using the default config with a small change that allows me to respond to the various values originate_disposition values.
>> 
>> First, is there a better way than what I am doing?  Everything seems to work fine until I try it with an attended transfer.  I've experienced this with Polycom, Cisco, and a soft phone client, so I don't think it's the phones, but who knows?  With 3 extensions, say 1001, 1002, 1003, each on a different phone, I can call from 1001 to 1002.  1002 initiates a transfer to 1003.  1003 rejects so 1002 starts hearing voicemail.  1002 hits transfer again to "connect" 1001 to 1003's voicemail.  With the default FS dialplan, it starts 1003's voicemail over, which is good.  With this slight modification, it hangs up on 1001.
>> 
>> If the feedback I get is that this modification looks fine, I can open a jira and supply a SIP trace and fs logs for the calls.
>> 
>> Incidentally, to keep it simple, this example does the same thing regardless of the originate_disposition.  In practice, there would be different actions taken.
>> 
>> Thanks for any help!
>> 
>> -Troy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Wed Apr 21 10:25:35 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 21 Apr 2010 13:25:35 -0400
Subject: [Freeswitch-users] ODBC DB Support in Windows
Message-ID: <m2h367751821004211025s112e4fc4wc882ac4cea182b00@mail.gmail.com>

Hi there,

I am trying to set up ODBC support on Win2003. I have created and tested a
system DSN and added  this to sofia.conf.xml in the <global_settings>
section:

*<param name="odbc-dsn" value="FS:xx:password"/>*

Is there more to do?

Freeswitch starts with no problem - seemingly ignoring this request to use
ODBC.

What am I missing??


Phil
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From alexandre.mouille at gmail.com  Wed Apr 21 09:07:00 2010
From: alexandre.mouille at gmail.com (=?UTF-8?Q?Alexandre_Mouill=C3=A9?=)
Date: Wed, 21 Apr 2010 18:07:00 +0200
Subject: [Freeswitch-users] create dummy channel to record conference
Message-ID: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>

Hi,

I hope I'm not posting this in the wrong mailing list.

I'm working on a custom conference that uses the conference module. I need a
record function that keeps recording (or at list appends the same wav file)
when no one is in a given conference.
The solution that was used in the Asterisk program I'm porting was to create
a dummy channel that goes in the conference and records what it hears in a
file.

- Can I create a dummy channel with an "originate /loopback" that would go
into a conf and record what it hears (I'm not sure what loopback really is)

- Otherwise, can I use originate from the FS box to call itself, one side
connects to the conference and the other records.
(I'm not sure how to do that, and how I could transmit variables (to tell
which file to record to) to the called program)

Thanks
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From brian at freeswitch.org  Wed Apr 21 10:40:18 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 12:40:18 -0500
Subject: [Freeswitch-users] create dummy channel to record conference
In-Reply-To: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
References: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
Message-ID: <349A66AF-38C9-421B-A963-ACAB94A02600@freeswitch.org>

The conference has a native record functionality and the core knows how to append but I'm not sure the conference itself can append.  I think we would need to have a bounty posted for that.

/b

On Apr 21, 2010, at 11:07 AM, Alexandre Mouill? wrote:

> Hi,
> 
> I hope I'm not posting this in the wrong mailing list.
> 
> I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference.
> The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file.
> 
> - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is)
> 
> - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records.
> (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program)
> 
> Thanks 



From jeremy at seadragons.us  Wed Apr 21 11:03:09 2010
From: jeremy at seadragons.us (Jeremy Shaffner)
Date: Wed, 21 Apr 2010 14:03:09 -0400
Subject: [Freeswitch-users] create dummy channel to record conference
In-Reply-To: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
References: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
Message-ID: <CEB9D2B0-6F6E-4C0C-90F7-1E15EA61F44C@seadragons.us>

You want the conference always running right?  Could you originate a call between an extension running the record app and the conference?

On Apr 21, 2010, at 12:07 PM, Alexandre Mouill? wrote:

> Hi,
> 
> I hope I'm not posting this in the wrong mailing list.
> 
> I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference.
> The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file.
> 
> - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is)
> 
> - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records.
> (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program)
> 
> 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 ranjtech at gmail.com  Wed Apr 21 11:07:51 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 21 Apr 2010 14:07:51 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
Message-ID: <k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>

Cool, that works too. But I guess in Asterisk it's easier to extract the
"FROM" header using the SIP_HEADER function than some custom header
variable...although it might just be easy, I just don't know how to do it.

Thanks anyway,
RR

On Wed, Apr 21, 2010 at 11:43 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Well, yes.
> Or you use the network_addr variable, and you put it in a custom header
> X-Original-IP or whatever.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/04/2010 ? 17:34, RR a ?crit :
>
> Thanks David. We know the custom header is the way to go but didn't know
> how to send that in the header. After a little research, discovered that we
> could use the sip_invite_domain channel variable to forward the call with
> the original "From" Header to the Asterisk Farm. Just thought it would help
> someone else out in the future :)
>
> On Tue, Apr 20, 2010 at 9:16 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Setting a customer SIP header seems a nice way to do that. And easy.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/04/2010 ? 01:24, RR a ?crit :
>>
>> Hello list,
>>
>> we have an FS server configured as a call-distributor (using
>> "mod_distributor") to a bank/farm of Asterisk servers. However, since the FS
>> is acting like a B2BUA, it's not passing the originating IP (network_addr)
>> from the originating switch/gateway to the Asterisk farm (as expected).
>> However, we need to have this information in the Asterisk servers. How can
>> we achieve this? Do we need to create/insert a custom SIP header when
>> passing on the call or set a variable or any other way?
>>
>> Any help on the issue will be much appreciated
>>
>> Thanks in advance,
>> RR
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 21 11:10:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 13:10:56 -0500
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
Message-ID: <EE00F8C6-9792-46E2-8609-61E2E662072D@freeswitch.org>

Try the info app... I will bet you money the answer to your question is already there.

/b

On Apr 21, 2010, at 1:07 PM, RR wrote:

> Cool, that works too. But I guess in Asterisk it's easier to extract the "FROM" header using the SIP_HEADER function than some custom header variable...although it might just be easy, I just don't know how to do it. 
> 
> Thanks anyway,
> RR



From david.ponzone at gmail.com  Wed Apr 21 11:15:20 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 21 Apr 2010 20:15:20 +0200
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
Message-ID: <30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>

It's not the right place for this but well :)

Should be ${SIP_HEADER(X-Original-IP)}

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/04/2010 ? 20:07, RR a ?crit :

> Cool, that works too. But I guess in Asterisk it's easier to extract  
> the "FROM" header using the SIP_HEADER function than some custom  
> header variable...although it might just be easy, I just don't know  
> how to do it.
>
> Thanks anyway,
> RR
>
> On Wed, Apr 21, 2010 at 11:43 AM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Well, yes.
> Or you use the network_addr variable, and you put it in a custom  
> header X-Original-IP or whatever.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 21/04/2010 ? 17:34, RR a ?crit :
>
>> Thanks David. We know the custom header is the way to go but didn't  
>> know how to send that in the header. After a little research,  
>> discovered that we could use the sip_invite_domain channel variable  
>> to forward the call with the original "From" Header to the Asterisk  
>> Farm. Just thought it would help someone else out in the future :)
>>
>> On Tue, Apr 20, 2010 at 9:16 PM, David Ponzone <david.ponzone at gmail.com 
>> > wrote:
>> Setting a customer SIP header seems a nice way to do that. And easy.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>> Le 21/04/2010 ? 01:24, RR a ?crit :
>>
>>> Hello list,
>>>
>>> we have an FS server configured as a call-distributor (using  
>>> "mod_distributor") to a bank/farm of Asterisk servers. However,  
>>> since the FS is acting like a B2BUA, it's not passing the  
>>> originating IP (network_addr) from the originating switch/gateway  
>>> to the Asterisk farm (as expected). However, we need to have this  
>>> information in the Asterisk servers. How can we achieve this? Do  
>>> we need to create/insert a custom SIP header when passing on the  
>>> call or set a variable or any other way?
>>>
>>> Any help on the issue will be much appreciated
>>>
>>> Thanks in advance,
>>> RR
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Wed Apr 21 11:14:54 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 21 Apr 2010 14:14:54 -0400
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
Message-ID: <201004211414.54538.sos@sokhapkin.dyndns.org>

What is the best way to pass --enable-ndebug option (without editing FS files) 
to sofia-sip configure script?

I'm getting FS crashes from time to time because of assertion failure in soa 
lib.


From brian at freeswitch.org  Wed Apr 21 11:22:49 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 13:22:49 -0500
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
In-Reply-To: <201004211414.54538.sos@sokhapkin.dyndns.org>
References: <201004211414.54538.sos@sokhapkin.dyndns.org>
Message-ID: <987EEB2B-B334-4784-A139-3AD34D21417A@freeswitch.org>

Why are you doing this?

/b

On Apr 21, 2010, at 1:14 PM, Sergey Okhapkin wrote:

> What is the best way to pass --enable-ndebug option (without editing FS files) 
> to sofia-sip configure script?
> 
> I'm getting FS crashes from time to time because of assertion failure in soa 
> lib.



From brian at freeswitch.org  Wed Apr 21 11:23:16 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 13:23:16 -0500
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
Message-ID: <9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>

In freeswitch the incoming X-Headers are done with sip_h_X-Header as a variable.

/b

On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:

> It's not the right place for this but well :)
> 
> Should be ${SIP_HEADER(X-Original-IP)}
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 

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From sos at sokhapkin.dyndns.org  Wed Apr 21 11:34:56 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 21 Apr 2010 14:34:56 -0400
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
In-Reply-To: <987EEB2B-B334-4784-A139-3AD34D21417A@freeswitch.org>
References: <201004211414.54538.sos@sokhapkin.dyndns.org>
	<987EEB2B-B334-4784-A139-3AD34D21417A@freeswitch.org>
Message-ID: <201004211434.56201.sos@sokhapkin.dyndns.org>

--enable-ndebug configure option will turn off assert() macros in sofia lib.

On Wednesday 21 April 2010, Brian West wrote:
> Why are you doing this?
> 
> /b
> 
> On Apr 21, 2010, at 1:14 PM, Sergey Okhapkin wrote:
> > What is the best way to pass --enable-ndebug option (without editing FS
> > files) to sofia-sip configure script?
> >
> > I'm getting FS crashes from time to time because of assertion failure in
> > soa lib.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 21 11:43:06 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 13:43:06 -0500
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
In-Reply-To: <201004211434.56201.sos@sokhapkin.dyndns.org>
References: <201004211414.54538.sos@sokhapkin.dyndns.org>
	<987EEB2B-B334-4784-A139-3AD34D21417A@freeswitch.org>
	<201004211434.56201.sos@sokhapkin.dyndns.org>
Message-ID: <E778ADD2-38E7-447E-A68A-1C3EABE9743B@freeswitch.org>

If you're hitting an assert you're hitting it for a reason... can you elaborate what exactly you'e doing that is asserting?

/b

On Apr 21, 2010, at 1:34 PM, Sergey Okhapkin wrote:

> --enable-ndebug configure option will turn off assert() macros in sofia lib.
> 
> On Wednesday 21 April 2010, Brian West wrote:
>> Why are you doing this?
>> 
>> /b



From sos at sokhapkin.dyndns.org  Wed Apr 21 11:56:42 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 21 Apr 2010 14:56:42 -0400
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
In-Reply-To: <E778ADD2-38E7-447E-A68A-1C3EABE9743B@freeswitch.org>
References: <201004211414.54538.sos@sokhapkin.dyndns.org>
	<201004211434.56201.sos@sokhapkin.dyndns.org>
	<E778ADD2-38E7-447E-A68A-1C3EABE9743B@freeswitch.org>
Message-ID: <201004211456.42474.sos@sokhapkin.dyndns.org>

Assertion is fired in soa/soa_static.c, line 1036.

   for (j = 0, um = user->sdp_media; j != s2u[i]; um = um->m_next, j++)
      assert(um);
    if (um == NULL) {
      if (dryrun)
          return 1;

According to gdb dryrun is 1, um could be 0. Seems like happens on reinvites. 
May be on fallback from t38 back to G711u.

BTW, windows build is done with -DNDENUG, asserts are disabled.


On Wednesday 21 April 2010, Brian West wrote:
> If you're hitting an assert you're hitting it for a reason... can you
>  elaborate what exactly you'e doing that is asserting?
> 
> /b
> 
> On Apr 21, 2010, at 1:34 PM, Sergey Okhapkin wrote:
> > --enable-ndebug configure option will turn off assert() macros in sofia
> > lib.
> >
> > On Wednesday 21 April 2010, Brian West wrote:
> >> Why are you doing this?
> >>
> >> /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  Wed Apr 21 12:00:58 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 14:00:58 -0500
Subject: [Freeswitch-users] Configure sofia-sip with --enable-ndebug
In-Reply-To: <201004211456.42474.sos@sokhapkin.dyndns.org>
References: <201004211414.54538.sos@sokhapkin.dyndns.org>
	<201004211434.56201.sos@sokhapkin.dyndns.org>
	<E778ADD2-38E7-447E-A68A-1C3EABE9743B@freeswitch.org>
	<201004211456.42474.sos@sokhapkin.dyndns.org>
Message-ID: <EFB49D1D-2D91-4395-B9A4-E82CC06B00E1@freeswitch.org>

Can you get me the sip traffic that causes this?

/b

On Apr 21, 2010, at 1:56 PM, Sergey Okhapkin wrote:

> Assertion is fired in soa/soa_static.c, line 1036.
> 
>   for (j = 0, um = user->sdp_media; j != s2u[i]; um = um->m_next, j++)
>      assert(um);
>    if (um == NULL) {
>      if (dryrun)
>          return 1;
> 
> According to gdb dryrun is 1, um could be 0. Seems like happens on reinvites. 
> May be on fallback from t38 back to G711u.
> 
> BTW, windows build is done with -DNDENUG, asserts are disabled.



From emilbergg at gmail.com  Wed Apr 21 12:05:08 2010
From: emilbergg at gmail.com (Emil Berg)
Date: Wed, 21 Apr 2010 22:05:08 +0300
Subject: [Freeswitch-users] Sip over TCP issues?
Message-ID: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>

Hello,

I'm not sure that this is the right mailing list for this question, but I'll
ask anyway :)
Are there any known issues with sip over tcp?

In some scenarios, freeswitch doesn't behave in a standard way.
Attached a picture with flow graph (from wireshark), where you can see such
a problematic behavior.
When user1 invites user2 (through the server), the server routes the invite
message to user2, but when user2 responds, the server doesn't send an ack to
this user!

Looking forward to your reply,
Thanks,

Emil.
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From janvb at live.com  Wed Apr 21 12:13:12 2010
From: janvb at live.com (Jan Berger)
Date: Wed, 21 Apr 2010 21:13:12 +0200
Subject: [Freeswitch-users] Java from Lua
In-Reply-To: <m2i191c3a031004210643h20d51b88o2d3b11401d5e6711@mail.gmail.com>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>,
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>,
	<SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>,
	<m2i191c3a031004210643h20d51b88o2d3b11401d5e6711@mail.gmail.com>
Message-ID: <SNT135-w24586229340FF36D1A8AC6D2090@phx.gbl>


I can make it run from shell - but it's bascally a collection of routines that I would like to use.

 

I know javascript has a Java intereface (JNI), but might be that working on the Java module makes more sence.

 

Jan
 


Date: Wed, 21 Apr 2010 08:43:50 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Java from Lua

is the java app run from the shell? 


On Wed, Apr 21, 2010 at 8:30 AM, Jan Berger <janvb at live.com> wrote:



hi folks,
 
I am working on a IVR demo and need to call some Java code from within a Lua. Basically i just want to run a Java app and pass some parameters forth and back.
 
I assume this might be easier using JavaScript or Java itself ???
 
Jan



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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
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From brian at freeswitch.org  Wed Apr 21 12:15:43 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 14:15:43 -0500
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
Message-ID: <A679FE13-48E5-4906-BCDA-0E9031919006@freeswitch.org>

You failed to frame this misbehavior with any useful info.  What REV are you running... what scenario, what mode of operation ie proxy media bypass media ... What I would like to see are debug logs on level 8 (press F8),   I have yet to see it misbehave unless something in the packets causes it to misbehave and a ladder graph won't give us enough info to go on.

/b

On Apr 21, 2010, at 2:05 PM, Emil Berg wrote:

> Hello,
> 
> I'm not sure that this is the right mailing list for this question, but I'll ask anyway :)
> Are there any known issues with sip over tcp?
> 
> In some scenarios, freeswitch doesn't behave in a standard way.
> Attached a picture with flow graph (from wireshark), where you can see such a problematic behavior.
> When user1 invites user2 (through the server), the server routes the invite message to user2, but when user2 responds, the server doesn't send an ack to this user!
> 
> Looking forward to your reply,
> Thanks,
> 
> Emil.



From anthony.minessale at gmail.com  Wed Apr 21 12:19:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Apr 2010 14:19:54 -0500
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
Message-ID: <s2m191c3a031004211219h65e251d3p70b83f617ac1a5b3@mail.gmail.com>

are you running that on the FS box ?
run the inline sip console trace to confirm it's not sending to somewhere
else based on a changed contact addr.


On Wed, Apr 21, 2010 at 2:05 PM, Emil Berg <emilbergg at gmail.com> wrote:

> Hello,
>
> I'm not sure that this is the right mailing list for this question, but
> I'll ask anyway :)
> Are there any known issues with sip over tcp?
>
> In some scenarios, freeswitch doesn't behave in a standard way.
> Attached a picture with flow graph (from wireshark), where you can see such
> a problematic behavior.
> When user1 invites user2 (through the server), the server routes the invite
> message to user2, but when user2 responds, the server doesn't send an ack to
> this user!
>
> Looking forward to your reply,
> Thanks,
>
> Emil.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From alexandre.mouille at gmail.com  Wed Apr 21 12:23:59 2010
From: alexandre.mouille at gmail.com (=?UTF-8?Q?Alexandre_Mouill=C3=A9?=)
Date: Wed, 21 Apr 2010 21:23:59 +0200
Subject: [Freeswitch-users] create dummy channel to record conference
In-Reply-To: <CEB9D2B0-6F6E-4C0C-90F7-1E15EA61F44C@seadragons.us>
References: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
	<CEB9D2B0-6F6E-4C0C-90F7-1E15EA61F44C@seadragons.us>
Message-ID: <l2s6b2d481e1004211223o35cd2f3epa3cd86c9e8043de6@mail.gmail.com>

Is that possible? (I head Asterisk wouldn't allow that) I would need to pass
a variable to tell ne name of the file to record to.

2010/4/21 Jeremy Shaffner <jeremy at seadragons.us>

> You want the conference always running right?  Could you originate a call
> between an extension running the record app and the conference?
>
> On Apr 21, 2010, at 12:07 PM, Alexandre Mouill? wrote:
>
> > Hi,
> >
> > I hope I'm not posting this in the wrong mailing list.
> >
> > I'm working on a custom conference that uses the conference module. I
> need a record function that keeps recording (or at list appends the same wav
> file) when no one is in a given conference.
> > The solution that was used in the Asterisk program I'm porting was to
> create a dummy channel that goes in the conference and records what it hears
> in a file.
> >
> > - Can I create a dummy channel with an "originate /loopback" that would
> go into a conf and record what it hears (I'm not sure what loopback really
> is)
> >
> > - Otherwise, can I use originate from the FS box to call itself, one side
> connects to the conference and the other records.
> > (I'm not sure how to do that, and how I could transmit variables (to tell
> which file to record to) to the called program)
> >
> > 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 21 12:27:18 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Apr 2010 14:27:18 -0500
Subject: [Freeswitch-users] create dummy channel to record conference
In-Reply-To: <l2s6b2d481e1004211223o35cd2f3epa3cd86c9e8043de6@mail.gmail.com>
References: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
	<CEB9D2B0-6F6E-4C0C-90F7-1E15EA61F44C@seadragons.us>
	<l2s6b2d481e1004211223o35cd2f3epa3cd86c9e8043de6@mail.gmail.com>
Message-ID: <FDDDFFC9-A562-4E50-AA26-D851A0EB9FF5@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_conference

Search for record.

/b

On Apr 21, 2010, at 2:23 PM, Alexandre Mouill? wrote:

> Is that possible? (I head Asterisk wouldn't allow that) I would need to pass a variable to tell ne name of the file to record to.
> 
> 2010/4/21 Jeremy Shaffner <jeremy at seadragons.us>
> You want the conference always running right?  Could you originate a call between an extension running the record app and the conference?
> 

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From jeremy at seadragons.us  Wed Apr 21 12:46:59 2010
From: jeremy at seadragons.us (Jeremy Shaffner)
Date: Wed, 21 Apr 2010 15:46:59 -0400
Subject: [Freeswitch-users] create dummy channel to record conference
In-Reply-To: <FDDDFFC9-A562-4E50-AA26-D851A0EB9FF5@freeswitch.org>
References: <s2z6b2d481e1004210907p1865889jef6315437b59532f@mail.gmail.com>
	<CEB9D2B0-6F6E-4C0C-90F7-1E15EA61F44C@seadragons.us>
	<l2s6b2d481e1004211223o35cd2f3epa3cd86c9e8043de6@mail.gmail.com>
	<FDDDFFC9-A562-4E50-AA26-D851A0EB9FF5@freeswitch.org>
Message-ID: <F65CC767-B896-4AFB-BB90-C1001E8FF7B9@seadragons.us>

I think he wants a persistent conference.

On Apr 21, 2010, at 3:27 PM, Brian West wrote:

> http://wiki.freeswitch.org/wiki/Mod_conference
> 
> Search for record.
> 
> /b
> 
> On Apr 21, 2010, at 2:23 PM, Alexandre Mouill? wrote:
> 
>> Is that possible? (I head Asterisk wouldn't allow that) I would need to pass a variable to tell ne name of the file to record to.
>> 
>> 2010/4/21 Jeremy Shaffner <jeremy at seadragons.us>
>> You want the conference always running right?  Could you originate a call between an extension running the record app and the conference?
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ranjtech at gmail.com  Wed Apr 21 13:24:16 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 21 Apr 2010 16:24:16 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
	<9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
Message-ID: <w2udcde51f21004211324yce2e0861j8d5369bd4f054883@mail.gmail.com>

David,

Thanks. I was thinking the same thing but figured it's neater for me to
modify the FROM header so even the CDRs will have the correct origination IP
as the CDRs might/will not carry the customer X-Header by default and we'll
have to extract it and insert it into the Asterisk CDRs to capture the
origination IP.

Brian,

Yep got that from what David said. Like I said, I figured out how to do it,
just didn't know in Asterisk how to extract a customer header in SIP.

Thanks all
\RR

2010/4/21 Brian West <brian at freeswitch.org>

> In freeswitch the incoming X-Headers are done with sip_h_X-Header as a
> variable.
>
> /b
>
> On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:
>
> It's not the right place for this but well :)
>
> Should be ${SIP_HEADER(X-Original-IP)}
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brent at overthewire.com.au  Wed Apr 21 13:24:33 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Thu, 22 Apr 2010 06:24:33 +1000
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <949726.30191.qm@web208.biz.mail.re2.yahoo.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com> 
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
	<949726.30191.qm@web208.biz.mail.re2.yahoo.com>
Message-ID: <w2vad139bd21004211324lfbc0c191p438d971de736e9bc@mail.gmail.com>

Hi Kashif,

I couldn't find it either.  Perhaps because once I'd been into the store to
buy the $25 kit, I didn't go back in there.  Perhaps you could also think
about putting another folder in the following location called "connectors"
or something ? :

https://www.vestec.ca/items/files

That is where I had expected to see it.

<https://www.vestec.ca/items/files>Brent

On Thu, Apr 22, 2010 at 12:52 AM, Kashif Kahn <info at evestech.com> wrote:

> Hi Jay,
>
> The Freeswitch connector is available for download free-of-charge from the
> Vestec webstore; you can see the connector download option advertized
> prominently on the main webstore page at: http://www.vestec.ca/products
>
> Please contact support at vestec.ca for any questions/issues on using the
> engine or connector. We provide installation support free of charge.
>
> Regards,
> -Kashif
>
> ------------------------------
> *From:* Jay Binks <jaybinks at gmail.com>
> *To:* "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> *Sent:* Wed, April 21, 2010 10:39:46 AM
> *Subject:* Re: [Freeswitch-users] $25 - Robust Affordable Speech
> Recognition
>
> Bought the 25$ kit yesterday.
>
> Where do I download the connector ??
> Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled
> on my account ;( )
>
> Also you need to wiki up some mad examples...
>
> Can't wait to play though ....
>
> J
>
>
>
> On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com> wrote:
>
> Dear All,
>
> All those who have wanted to try speech recognition with Freeswitch but
> found the software cost too expensive or the recognition accuracy
> unsatisfactory, I encourage you to try our Vestec Speech Engine for
> Freeswitch at: <http://www.vestec.ca/products>
> http://www.vestec.ca/products A starter kit - which is a specially priced
> one port (ie. one channel) license for the standard engine - is available
> for only $25. Additional ports (channels) licenses can be purchased for
> $99/port. Of course, the engine comes with a Freeswitch connector, thereby
> allowing a Freeswitch user to bypass engine API and interact directly via
> Dialplan.
>
> Best regards,
> -Kashif
>
> Kashif Kahn
> VP, Business Development
> Vestec, Inc.
> Waterloo, ON Canada
> phone: (519) 885-7615
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From ranjtech at gmail.com  Wed Apr 21 13:28:03 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 21 Apr 2010 16:28:03 -0400
Subject: [Freeswitch-users] reloadxml not reloading gateway configuration
Message-ID: <h2ndcde51f21004211328j362b2981od4f6c06d11648304@mail.gmail.com>

Hello,

using version

freeswitch at internal> version
FreeSWITCH Version 1.0.5pre9 (hacked)

and when I change the configuration of a gateway under
conf/sip_profiles/external/gateway2.xml configuration and then do a
reloadxml, nothing changes within the freeswitch db as seen from 'sofia
status'. Is this a bug and has been fixed in newer versions?

Also, does doing an 'fs_ctl send_sighup' drop all calls from the system?
would doing that reload the gateway configurations?

Thanks

\RR
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From david.ponzone at gmail.com  Wed Apr 21 13:40:19 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 21 Apr 2010 22:40:19 +0200
Subject: [Freeswitch-users] reloadxml not reloading gateway configuration
In-Reply-To: <h2ndcde51f21004211328j362b2981od4f6c06d11648304@mail.gmail.com>
References: <h2ndcde51f21004211328j362b2981od4f6c06d11648304@mail.gmail.com>
Message-ID: <26B9CD23-4E1D-403F-B345-5B49FA7C13E1@gmail.com>

For gateways, you need to:

sofia profile external rescan reloadxml

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/04/2010 ? 22:28, RR a ?crit :

> Hello,
>
> using version
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5pre9 (hacked)
>
> and when I change the configuration of a gateway under conf/ 
> sip_profiles/external/gateway2.xml configuration and then do a  
> reloadxml, nothing changes within the freeswitch db as seen from  
> 'sofia status'. Is this a bug and has been fixed in newer versions?
>
> Also, does doing an 'fs_ctl send_sighup' drop all calls from the  
> system? would doing that reload the gateway configurations?
>
> Thanks
>
> \RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Wed Apr 21 15:07:10 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 21 Apr 2010 18:07:10 -0400
Subject: [Freeswitch-users] Windows Build of latest git version
Message-ID: <r2u367751821004211507pb122cd57obcf13f85b9d704a4@mail.gmail.com>

Hi All,

I just downloaded the lasted version from git, and I am having real problems
compiling in VS2008 for Windows. FreeSwithCoreLib, mod_PortAudio and others
are fine, but when building libsofia_sip_ua_static I get 905 errors, eg:

Error    601    error C2065: 'sip_supported_hash' : undeclared identifier
c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c
279    libsofia_sip_ua_static
Error    602    error C2099: initializer is not a constant
c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c
279    libsofia_sip_ua_static

etc etc

I have previously compiled successfully many times before, although not on
this machine.

Anyone else experience this and know how to fix?

TIA

Phil
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From neilp at cs.stanford.edu  Wed Apr 21 15:10:08 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Wed, 21 Apr 2010 15:10:08 -0700
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
Message-ID: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com>

I'm trying to make a sip call using Gizmo. I set up the profile as specified
here <http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo> and
dropped it into /conf/sip_profiles/external/. I can see that the profile is
registered when I check sofia status from CLI:
                     Name   Type                                      Data
State
=================================================================================================
                 ...
         external::gizmo gateway
sip:otalo at proxy01.sipphone.com<sip%3Aotalo at proxy01.sipphone.com>
REGED
           ...
=================================================================================================

On a call event, I invoke the lua commands:

sessiondata = "sofia/gateway/gizmo/<num>"
new_session = freeswitch.Session(sessiondata)

>From this I'm getting a NO_USER_RESPONSE error:

2010-04-21 11:31:16.169900 [DEBUG] sofia.c:4153 Channel sofia/internal/
1234 at conference.freeswitch.org entering state [terminated][480]
2010-04-21 11:31:16.169900 [NOTICE] sofia.c:4789 Hangup sofia/internal/
1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2010-04-21 11:31:16.169900 [DEBUG] switch_channel.c:2071 Send signal
sofia/internal/1234 at conference.freeswitch.org [KILL]
2010-04-21 11:31:16.169900 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/1234 at conference.freeswitch.org [BREAK]
2010-04-21 11:31:16.169900 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1234 at conference.freeswitch.org) Running State Change
CS_HANGUP
2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/1234 at conference.freeswitch.org) State HANGUP
2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:408 sofia/internal/
1234 at conference.freeswitch.org Overriding SIP cause 408 with 480 from the
other leg
2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:414 Channel sofia/internal/
1234 at conference.freeswitch.org hanging up, cause: NO_USER_RESPONSE
2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1234 at conference.freeswitch.org Standard HANGUP, cause:
NO_USER_RESPONSE
2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/1234 at conference.freeswitch.org) State HANGUP going to sleep
2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1234 at conference.freeswitch.org) State Change CS_HANGUP ->
CS_REPORTING
2010-04-21 11:31:16.170748 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/1234 at conference.freeswitch.org [BREAK]
2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1234 at conference.freeswitch.org) Running State Change
CS_REPORTING
2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/1234 at conference.freeswitch.org) State REPORTING
2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1234 at conference.freeswitch.org Standard REPORTING, cause:
NO_USER_RESPONSE
2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/1234 at conference.freeswitch.org) State REPORTING going to
sleep
2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/1234 at conference.freeswitch.org) State Change CS_REPORTING ->
CS_DESTROY
2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/1234 at conference.freeswitch.org [BREAK]
2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1161 Session 4
(sofia/internal/1234 at conference.freeswitch.org) Locked, Waiting on external
entities
2010-04-21 11:31:16.177790 [DEBUG] switch_ivr_originate.c:3228 Originate
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
*2010-04-21 11:31:16.177790 [ERR] mod_conference.c:4563 Cannot create
outgoing channel, cause: NO_USER_RESPONSE*
2010-04-21 11:31:16.177790 [NOTICE] mod_conference.c:4566 Hangup
sofia/internal/1001 at server.IP [CS_EXECUTE] [NO_USER_RESPONSE]
2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1179 Session 4
(sofia/internal/1234 at conference.freeswitch.org) Ended
2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:428
(sofia/internal/1234 at conference.freeswitch.org) Running State Change
CS_DESTROY
2010-04-21 11:31:16.177790 [DEBUG] switch_channel.c:2071 Send signal
sofia/internal/1001 at server.IP [KILL]
2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/1234 at conference.freeswitch.org) State DESTROY
2010-04-21 11:31:16.177790 [DEBUG] mod_sofia.c:341 sofia/internal/
1234 at conference.freeswitch.org SOFIA DESTROY
2010-04-21 11:31:16.177790 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/1001 at server.IP [BREAK]

Strangely, I get the same error even if I put in some arbitrary
gateway name in the lua code, so it seems like possibly the profile
isn't being properly read. I also thought that perhaps I should be
using "sofia/gateway/external::gizmo/<num>" instead of "../gizmo/..",
but that didn't seem to work. I haven't made any other changes to conf
files, though perhaps I'm missing something?

Thanks in advance,
Neil
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From info at evestech.com  Wed Apr 21 15:41:57 2010
From: info at evestech.com (Kashif Kahn)
Date: Wed, 21 Apr 2010 15:41:57 -0700 (PDT)
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <w2vad139bd21004211324lfbc0c191p438d971de736e9bc@mail.gmail.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
	<949726.30191.qm@web208.biz.mail.re2.yahoo.com>
	<w2vad139bd21004211324lfbc0c191p438d971de736e9bc@mail.gmail.com>
Message-ID: <49586.87404.qm@web206.biz.mail.re2.yahoo.com>

Hi Brent,

This is very strange. 

Respectfully, if you have already purchased a speech engine license from Vestec webstore, please follow the following steps to download the free FreeSWITCH connector:

1) Go to the main page of Vestec webstore at: http://www.vestec.ca/products

2) Login with the user name and password you used during registration (prior to purchase). If you have forgotten your password, click on "Forget Password" option and follow the directions.

3) Once you have logged into the webstore - on the main webstore page (that is clearly visible when you are logging in) - select the 5th option called "Speech Engine - FreeSWITCH Connector". (You can do this either by clicking on the dish icon to the left of the name or by pressing "Download Now")

4) Step (3) above will take you to a new page for "Speech Engine - FreeSWITCH Connector". On that page, you simply have to click on "Download Connector" icon on top right hand side. This will open a form for you where you can specify your OS and architecture for the desired connector and download it.

If you have any issues, email me your phone number at info at evestech.com and I will be happy to walk you through the steps. You are also welcome to contact support at vestec.ca with any issues regarding use of speech engine or FS connector. 

Best regards,
-Kashif





________________________________
From: Brent Paddon <brent at overthewire.com.au>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, April 21, 2010 4:24:33 PM
Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition

Hi Kashif,

I couldn't find it either.  Perhaps because once I'd been into the store to buy the $25 kit, I didn't go back in there.  Perhaps you could also think about putting another folder in the following location called "connectors" or something ? :

https://www.vestec.ca/items/files

That is where I had expected to see it.

Brent


On Thu, Apr 22, 2010 at 12:52 AM, Kashif Kahn <info at evestech.com> wrote:

>
>Hi Jay,
>
>The Freeswitch connector is available for download free-of-charge from the Vestec webstore; you can see the connector download option advertized prominently on the main webstore page at: http://www.vestec.ca/products
>
>Please contact support at vestec.ca for any questions/issues on using the engine or connector. We provide installation support free of charge.
>
>Regards,
>-Kashif
>
>
>
________________________________
From: Jay Binks <jaybinks at gmail.com>
>To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
>Sent: Wed, April 21, 2010 10:39:46 AM
>Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
>
>>
>
>
>Bought the 25$ kit yesterday.
>
>
>Where do I download the connector ??
>Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled on my account ;( )
>
>
>Also you need to wiki up some mad examples...
>
>
>Can't wait to play though ....  
>
>
>J
>
>
>
>
>
>On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com>
> wrote:
>
>
>Dear All,
>>
>>All those who have wanted to try speech recognition with Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, I encourage you to try our Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. Of course, the engine comes with a Freeswitch connector, thereby allowing a Freeswitch user to bypass engine API and interact directly via Dialplan.
>>
>>Best
>> regards,
>>-Kashif
>>
>> Kashif Kahn
>>VP, Business Development
>>Vestec,
>> Inc.
>>Waterloo, ON Canada
>>phone: (519) 885-7615
>>
>>
>_______________________________________________
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From jeff at jefflenk.com  Wed Apr 21 20:34:48 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 21 Apr 2010 19:34:48 -0800 (PST)
Subject: [Freeswitch-users] Windows Build of latest git version
In-Reply-To: <r2u367751821004211507pb122cd57obcf13f85b9d704a4@mail.gmail.com>
References: <r2u367751821004211507pb122cd57obcf13f85b9d704a4@mail.gmail.com>
Message-ID: <1271907288609-4940814.post@n2.nabble.com>


Hi Phillip,

have you set the autocrlf option in git to false?

If no delete all the working set files and recheckout. FreeSWITCH/Git under
Windows requires the files to be left as is with no modification otherwise
the autogenerate scripts fail.

-Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-Build-of-latest-git-version-tp4939766p4940814.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Wed Apr 21 20:38:07 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 22 Apr 2010 05:38:07 +0200
Subject: [Freeswitch-users] Windows Build of latest git version
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4CCF@cooper>

When using git in Windows you must make sure to set core.autocrlf=false in the git configuration. The suggested way when installing is true, and it causes problems with some LF to CRLF conversions.

/Peter


________________________________
Fr?n: Phillip Jones <pjintheusa at gmail.com>
Skickat: den 22 april 2010 00:16
Till: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] Windows Build of latest git version

Hi All,

I just downloaded the lasted version from git, and I am having real problems compiling in VS2008 for Windows. FreeSwithCoreLib, mod_PortAudio and others are fine, but when building libsofia_sip_ua_static I get 905 errors, eg:

Error    601    error C2065: 'sip_supported_hash' : undeclared identifier    c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c    279    libsofia_sip_ua_static
Error    602    error C2099: initializer is not a constant    c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c    279    libsofia_sip_ua_static

etc etc

I have previously compiled successfully many times before, although not on this machine.

Anyone else experience this and know how to fix?

TIA

Phil
!DSPAM:4bcf792732932989516006!


From lists at infosecurity.ch  Wed Apr 21 23:21:11 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 22 Apr 2010 08:21:11 +0200
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
Message-ID: <4BCFEAD7.7050902@infosecurity.ch>

Even if i have not done a so in depth analysis, i have several random 
and not reproduceable behaviour (but continuous) using FS with SIP/TLS 
(i only use it over SIP/TLS) that seems to have a representation of the 
issue equal to this one reported by Emil.

Fabio

On 21/04/10 21.05, Emil Berg wrote:
> Hello,
>
> I'm not sure that this is the right mailing list for this question, 
> but I'll ask anyway :)
> Are there any known issues with sip over tcp?



From steveayre at gmail.com  Thu Apr 22 00:25:23 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 22 Apr 2010 08:25:23 +0100
Subject: [Freeswitch-users] reloadxml not reloading gateway configuration
In-Reply-To: <26B9CD23-4E1D-403F-B345-5B49FA7C13E1@gmail.com>
References: <h2ndcde51f21004211328j362b2981od4f6c06d11648304@mail.gmail.com>
	<26B9CD23-4E1D-403F-B345-5B49FA7C13E1@gmail.com>
Message-ID: <n2ieafa772f1004220025xa5c9a2bfvf5b023821b38a9a2@mail.gmail.com>

RR,

What David's given you will work for adding/removing gateways. If
you've changed a gateway's parameters you'll want to killgw first
though because if a gateway with the same name already exists it'll
ignore it (If this is no longer the case, please let me know!). I
don't think that would drop calls, but it would stop you routing to it
between the killgw and the rescan.

Sighup won't drop any calls, it's often sent by a cronjob once an hour
to rotate log files.

-Steve


On 21 April 2010 21:40, David Ponzone <david.ponzone at gmail.com> wrote:
> For gateways, you need to:
> sofia profile external rescan reloadxml
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 21/04/2010 ? 22:28, RR a ?crit :
>
> Hello,
> using version
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5pre9 (hacked)
> and when I change the configuration of a gateway under
> conf/sip_profiles/external/gateway2.xml configuration and then do a
> reloadxml, nothing changes within the freeswitch db as seen from 'sofia
> status'. Is this a bug and has been fixed in newer versions?
> Also, does doing an 'fs_ctl?send_sighup' drop all calls from the system?
> would doing that reload the gateway configurations?
> Thanks
> \RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From noisewaterphd at gmail.com  Thu Apr 22 00:32:51 2010
From: noisewaterphd at gmail.com (Dr. Kenneth Noisewater)
Date: Thu, 22 Apr 2010 01:32:51 -0600
Subject: [Freeswitch-users] Java from Lua
In-Reply-To: <SNT135-w24586229340FF36D1A8AC6D2090@phx.gbl>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>,
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>,
	<SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>,
	<m2i191c3a031004210643h20d51b88o2d3b11401d5e6711@mail.gmail.com>
	<SNT135-w24586229340FF36D1A8AC6D2090@phx.gbl>
Message-ID: <4BCFFBA3.4040101@gmail.com>


You could expose your java funcions as a web service and then call them 
from lua.

Kenny

On 4/21/2010 1:13 PM, Jan Berger wrote:
> I can make it run from shell - but it's bascally a collection of 
> routines that I would like to use.
>
> I know javascript has a Java intereface (JNI), but might be that 
> working on the Java module makes more sence.
>
> Jan
>
> ------------------------------------------------------------------------
> Date: Wed, 21 Apr 2010 08:43:50 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Java from Lua
>
> is the java app run from the shell?
>
> On Wed, Apr 21, 2010 at 8:30 AM, Jan Berger <janvb at live.com 
> <mailto:janvb at live.com>> wrote:
>
>
>     hi folks,
>
>     I am working on a IVR demo and need to call some Java code from
>     within a Lua. Basically i just want to run a Java app and pass
>     some parameters forth and back.
>
>     I assume this might be easier using JavaScript or Java itself ???
>
>     Jan
>
>     ------------------------------------------------------------------------
>     Hotmail: Trusted email with powerful SPAM protection. Sign up now.
>     <https://signup.live.com/signup.aspx?id=60969>
>
>     _______________________________________________
>     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 <http://www.freeswitch.org/>
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN:anthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL:anthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip:888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------------------------------------------------
> Hotmail: Free, trusted and rich email service. Get it now. 
> <https://signup.live.com/signup.aspx?id=60969>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From aep.lists at it46.se  Thu Apr 22 01:20:09 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Thu, 22 Apr 2010 10:20:09 +0200
Subject: [Freeswitch-users] Stopping an audio file playing in a XML IVR
Message-ID: <67f046e1e01e5c70b45752a2083e4792.squirrel@correo.nodo50.org>

Hi,

Is there any way to stop/exit/jumpo out from the play-file function inside
of a XML IVR?

<action function="play-file" data="content.wav"/>

/aep


-- 
Stopping junk mailers is good for the environment





From aep.lists at it46.se  Thu Apr 22 01:30:24 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Thu, 22 Apr 2010 10:30:24 +0200
Subject: [Freeswitch-users] Parsing XML files from Spidermonkey
In-Reply-To: <48fade7f8a99c447d71334d2d3e589bd.squirrel@correo.nodo50.org>
References: <48fade7f8a99c447d71334d2d3e589bd.squirrel@correo.nodo50.org>
Message-ID: <14f6fde22ecd0ad9b6fdd7a7fe56e3d0.squirrel@correo.nodo50.org>

I am resending this mail, hoping that someone has managed to read a XML
file from Javascript.


-- 
Stopping junk mailers is good for the environment

> Hi,
>
> After one year using FS i am starting to like XML so i am trying to get
> a Javascript script to read local XML files.
>
> I am using the XML method and getting Syntax errors from spidermonkey
>
> While something like this works:
> xmldata = new XML("<tag>foo</tag>");
>
> I have not been able to read and parse XML local files, using File or
> FileIO
> methods
>
> A simple example like this returns Syntax error.
> var foo = apiExecute ("show", "channels as xml");
> xmldata = new XML(foo);
>
> Has anyone managed to use the XML method from spidermonkey to read a XML
> stored file?
>
> There are some E4X bugs around and i wonder if those are the cause of the
> "Syntax Error" feedback even reading a very basic XML file
> --
> Stopping junk mailers is good for the environment
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From janvb at live.com  Thu Apr 22 02:32:21 2010
From: janvb at live.com (Jan Berger)
Date: Thu, 22 Apr 2010 11:32:21 +0200
Subject: [Freeswitch-users] Java from Lua
In-Reply-To: <4BCFFBA3.4040101@gmail.com>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>, ,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>, ,
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>, ,
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>, ,
	<SNT135-w20D3010718AECB0EEC9310D2090@phx.gbl>, ,
	<m2i191c3a031004210643h20d51b88o2d3b11401d5e6711@mail.gmail.com>,
	<SNT135-w24586229340FF36D1A8AC6D2090@phx.gbl>,
	<4BCFFBA3.4040101@gmail.com>
Message-ID: <SNT135-w218FFA2BC6B1CAB8FF8691D2080@phx.gbl>


Looking into that - do you have any free tool to convert wsdl/soap into Lua code?

 

Jan 


Date: Thu, 22 Apr 2010 01:32:51 -0600
From: noisewaterphd at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Java from Lua


You could expose your java funcions as a web service and then call them from lua.

Kenny

On 4/21/2010 1:13 PM, Jan Berger wrote: 


I can make it run from shell - but it's bascally a collection of routines that I would like to use.
 
I know javascript has a Java intereface (JNI), but might be that working on the Java module makes more sence.
 
Jan
 


Date: Wed, 21 Apr 2010 08:43:50 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Java from Lua

is the java app run from the shell? 


On Wed, Apr 21, 2010 at 8:30 AM, Jan Berger <janvb at live.com> wrote:



hi folks,
 
I am working on a IVR demo and need to call some Java code from within a Lua. Basically i just want to run a Java app and pass some parameters forth and back.
 
I assume this might be easier using JavaScript or Java itself ???
 
Jan



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



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900



Hotmail: Free, trusted and rich email service. Get it now. 
_______________________________________________
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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 brent at overthewire.com.au  Thu Apr 22 03:44:41 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Thu, 22 Apr 2010 20:44:41 +1000
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <49586.87404.qm@web206.biz.mail.re2.yahoo.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com> 
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
	<949726.30191.qm@web208.biz.mail.re2.yahoo.com> 
	<w2vad139bd21004211324lfbc0c191p438d971de736e9bc@mail.gmail.com> 
	<49586.87404.qm@web206.biz.mail.re2.yahoo.com>
Message-ID: <q2yad139bd21004220344w5656ck20fd670ea608aef@mail.gmail.com>

I think you missed my point.  My point was that once I'd already purchased
the starter kit, I expected to see the connector in the files download area.
 I did not think to go back into the webstore (and hadn't noticed it there
the first time!) to look for the downloads there.

Am I making sense?

I had already figured it out by the time I emailed back, but I figured some
user feedback on it would be helpful for you.  You have two users (Jay and
I) who couldn't see it.  Just feedback for you, take it or leave it :)

Brent

On Thu, Apr 22, 2010 at 8:41 AM, Kashif Kahn <info at evestech.com> wrote:

> Hi Brent,
>
> This is very strange.
>
> Respectfully, if you have already purchased a speech engine license from
> Vestec webstore, please follow the following steps to download the free
> FreeSWITCH connector:
>
> 1) Go to the main page of Vestec webstore at:
> http://www.vestec.ca/products
>
> 2) Login with the user name and password you used during registration
> (prior to purchase). If you have forgotten your password, click on "Forget
> Password" option and follow the directions.
>
> 3) Once you have logged into the webstore - on the main webstore page (that
> is clearly visible when you are logging in) - select the 5th option called
> "Speech Engine - FreeSWITCH Connector". (You can do this either by clicking
> on the dish icon to the left of the name or by pressing "Download Now")
>
> 4) Step (3) above will take you to a new page for "Speech Engine -
> FreeSWITCH Connector". On that page, you simply have to click on "Download
> Connector" icon on top right hand side. This will open a form for you where
> you can specify your OS and architecture for the desired connector and
> download it.
>
> If you have any issues, email me your phone number at info at evestech.comand I will be happy to walk you through the steps. You are also welcome to
> contact support at vestec.ca with any issues regarding use of speech engine
> or FS connector.
>
> Best regards,
> -Kashif
>
>
> ------------------------------
> *From:* Brent Paddon <brent at overthewire.com.au>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wed, April 21, 2010 4:24:33 PM
>
> *Subject:* Re: [Freeswitch-users] $25 - Robust Affordable Speech
> Recognition
>
> Hi Kashif,
>
> I couldn't find it either.  Perhaps because once I'd been into the store to
> buy the $25 kit, I didn't go back in there.  Perhaps you could also think
> about putting another folder in the following location called "connectors"
> or something ? :
>
> https://www.vestec.ca/items/files
>
> That is where I had expected to see it.
>
>  <https://www.vestec.ca/items/files>Brent
>
> On Thu, Apr 22, 2010 at 12:52 AM, Kashif Kahn <info at evestech.com> wrote:
>
>> Hi Jay,
>>
>> The Freeswitch connector is available for download free-of-charge from the
>> Vestec webstore; you can see the connector download option advertized
>> prominently on the main webstore page at: http://www.vestec.ca/products
>>
>> Please contact support at vestec.ca for any questions/issues on using the
>> engine or connector. We provide installation support free of charge.
>>
>> Regards,
>> -Kashif
>>
>> ------------------------------
>> *From:* Jay Binks <jaybinks at gmail.com>
>> *To:* "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wed, April 21, 2010 10:39:46 AM
>> *Subject:* Re: [Freeswitch-users] $25 - Robust Affordable Speech
>> Recognition
>>
>> Bought the 25$ kit yesterday.
>>
>> Where do I download the connector ??
>> Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled
>> on my account ;( )
>>
>> Also you need to wiki up some mad examples...
>>
>> Can't wait to play though ....
>>
>> J
>>
>>
>>
>> On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com> wrote:
>>
>> Dear All,
>>
>> All those who have wanted to try speech recognition with Freeswitch but
>> found the software cost too expensive or the recognition accuracy
>> unsatisfactory, I encourage you to try our Vestec Speech Engine for
>> Freeswitch at: <http://www.vestec.ca/products>
>> http://www.vestec.ca/products A starter kit - which is a specially priced
>> one port (ie. one channel) license for the standard engine - is available
>> for only $25. Additional ports (channels) licenses can be purchased for
>> $99/port. Of course, the engine comes with a Freeswitch connector, thereby
>> allowing a Freeswitch user to bypass engine API and interact directly via
>> Dialplan.
>>
>> Best regards,
>> -Kashif
>>
>> Kashif Kahn
>> VP, Business Development
>> Vestec, Inc.
>> Waterloo, ON Canada
>> phone: (519) 885-7615
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> --
> Brent Paddon
>
> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
> www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From dome at tel.co.th  Thu Apr 22 04:18:05 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 22 Apr 2010 18:18:05 +0700
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
Message-ID: <p2t8ccbff061004220418vfe701a3ubec610a3de511811@mail.gmail.com>

What's kind of android sip client ? Is it's work with video call ? (over sip)
i found http://code.google.com/p/sipdroid/ but never test yet


Dome C.

2010/4/21 Peter Olsson <peter.olsson at visionutveckling.se>:
> Thanks,
>
>
>
> So the a way around this would be to;
>
> 1.?????? Exclude the phones from the acl list.
>
> 2.?????? Or ? use another sip profile for ?open? trunks, and disable the
> inbound-acl on the internal profile?
>
>
>
> /Peter
>
>
>
>
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
> Skickat: den 21 april 2010 15:01
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] SIP Agent (Android SIP client)
>
>
>
> Correct.
>
>
>
> /b
>
>
>
> On Apr 21, 2010, at 6:23 AM, Peter Olsson wrote:
>
> Actually, I just noticed the same behaviour today in my lab system, when
> dialing from a Polycom phone. Though in this case I have ?apply-inbound-acl?
> set to allow access everything from my whole network (192.168.94.0/24),
> could this cause the call fr?n Polycom ext not being treated as a
> ?authenticated? call, and not apply the user_context to it?
>
>
>
> /Peter
>
>
>
> !DSPAM:4bcef85e32936933977704!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yivzhenko at mksat.net  Thu Apr 22 05:12:18 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 22 Apr 2010 15:12:18 +0300
Subject: [Freeswitch-users] mod_say or strftime problems?
In-Reply-To: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>
References: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>
Message-ID: <201004221512.18576.yivzhenko@mksat.net>

I have same problem with mod_say_ru if timezone is set to 

<action application="set" data="timezone=Europe/Kiev"/>

And i don't understand why  "tm.tm_hour + 1"
In source code 
        if (say_time) {
                switch_snprintf(buf, sizeof(buf), "%d:%d:%d", tm.tm_hour + 1, 
tm.tm_min, tm.tm_sec);
                say_args->type = SST_TIME_MEASUREMENT;
                ru_say_time(session, buf, say_args, args);

As temporary resolution i just set different timezone :)


On Wednesday 21 April 2010 17:30:18 boris at tagnet.ru wrote:
> Hello!
>
> I have an extension for the current date/time:
>     <extension name="SayTime">
>         <condition field="destination_number" expression="^9000$">
>             <action application="answer"/>
>             <action application="say" data="ru current_date_time
> pronounced $strftime(%s)"/>
>             <action application="sleep" data="1000"/>
>             <action application="hangup"/>
>         </condition>
>     </extension>
>
> The extension works fine, but time is +1 hour of current time. For example
> current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6
> (from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer
> time in effect now, so tz=YEKST). So, my question is - something wrong
> with my extension configuration or this is bug in mod_say_ru or may be
> strfmt?
>
> With respect,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 22 06:09:19 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 08:09:19 -0500
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <4BCFEAD7.7050902@infosecurity.ch>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
	<4BCFEAD7.7050902@infosecurity.ch>
Message-ID: <4C83016A-98CF-40C7-BA61-9EF23BB7F5B0@freeswitch.org>

Well it saddens me that these issues are just over looked and not reported.  When reporting issues with FreeSWITCH please follow the bug reporting guidelines on the wiki.  Sitting on bugs isn't going to get them fixed.  Not collecting the right info and reporting them properly won't either.

/b

On Apr 22, 2010, at 1:21 AM, Fabio Pietrosanti (naif) wrote:

> Even if i have not done a so in depth analysis, i have several random 
> and not reproduceable behaviour (but continuous) using FS with SIP/TLS 
> (i only use it over SIP/TLS) that seems to have a representation of the 
> issue equal to this one reported by Emil.
> 
> Fabio



From info at evestech.com  Thu Apr 22 06:25:10 2010
From: info at evestech.com (Kashif Kahn)
Date: Thu, 22 Apr 2010 06:25:10 -0700 (PDT)
Subject: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
In-Reply-To: <q2yad139bd21004220344w5656ck20fd670ea608aef@mail.gmail.com>
References: <964861.64117.qm@web205.biz.mail.re2.yahoo.com>
	<BB16B325-5DB1-4C82-B261-D25A00070E97@gmail.com>
	<949726.30191.qm@web208.biz.mail.re2.yahoo.com>
	<w2vad139bd21004211324lfbc0c191p438d971de736e9bc@mail.gmail.com>
	<49586.87404.qm@web206.biz.mail.re2.yahoo.com>
	<q2yad139bd21004220344w5656ck20fd670ea608aef@mail.gmail.com>
Message-ID: <687238.33029.qm@web203.biz.mail.re2.yahoo.com>

Hi Brent,

Thanks for the clarification. We will certainly take your feedback into account.

You may not be aware of this, from a billing and accounting point of view, we need to keep track of FreeSWITCH connector downloads. That is why we have created a separate download procedure for the FreeSWITCH connector.

Regards,
-Kashif





________________________________
From: Brent Paddon <brent at overthewire.com.au>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, April 22, 2010 6:44:41 AM
Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition

I think you missed my point.  My point was that once I'd already purchased the starter kit, I expected to see the connector in the files download area.  I did not think to go back into the webstore (and hadn't noticed it there the first time!) to look for the downloads there.

Am I making sense?

I had already figured it out by the time I emailed back, but I figured some user feedback on it would be helpful for you.  You have two users (Jay and I) who couldn't see it.  Just feedback for you, take it or leave it :)

Brent


On Thu, Apr 22, 2010 at 8:41 AM, Kashif Kahn <info at evestech.com> wrote:

>
>Hi Brent,
>
>This is very strange. 
>
>Respectfully, if you have already purchased a speech engine license from Vestec webstore, please follow the following steps to download the free FreeSWITCH connector:
>
>1) Go to the main page of Vestec webstore at: http://www.vestec.ca/products
>
>2) Login with the user name and password you used during registration (prior to purchase). If you have forgotten your password, click on "Forget Password" option and follow the directions.
>
>3) Once you have logged into the webstore - on the main webstore page (that is clearly visible when you are logging in) - select the 5th option called "Speech Engine - FreeSWITCH Connector". (You can do this either by clicking on the
> dish icon to the left of the name or by pressing "Download Now")
>
>4) Step (3) above will take you to a new page for "Speech Engine - FreeSWITCH Connector". On that page, you simply have to click on "Download Connector" icon on top right hand side. This will open a form for you where you can specify your OS and architecture for the desired connector and download it.
>
>If you have any issues, email me your phone number at info at evestech.com and I will be happy to walk you through the steps. You are also welcome to contact support at vestec.ca with any issues regarding use of speech engine or FS connector. 
>
>Best regards,
>-Kashif
>
>
>
>
>
>
________________________________
 >From: Brent Paddon
> <brent at overthewire.com.au>
>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Wed, April 21, 2010 4:24:33 PM
>
>Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
>
>
>Hi Kashif,
>
>
>I couldn't find it either.  Perhaps because once I'd been into the store to buy the $25 kit, I didn't go back in there.  Perhaps you could also think about putting another folder in the following location called "connectors" or something ? :
>
>
>https://www.vestec.ca/items/files
>
>
>That is where I had expected to see it.
>
>
>>Brent
>
>
>On Thu, Apr 22, 2010 at 12:52 AM, Kashif Kahn <info at evestech.com> wrote:
>
>>>
>>
>>
>>Hi Jay,
>>
>>The Freeswitch connector is available for download free-of-charge from the Vestec webstore; you can see the connector download option advertized prominently on the main webstore page at: http://www.vestec.ca/products
>>
>>Please contact support at vestec.ca for any questions/issues on using the engine or connector. We provide installation support free of charge.
>>
>>>>
>>Regards,
>>-Kashif
>>
>>
>>
________________________________
From: Jay Binks <jaybinks at gmail.com>
>>To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
>>Sent: Wed, April 21, 2010 10:39:46 AM
>>Subject: Re: [Freeswitch-users] $25 - Robust Affordable Speech Recognition
>>
>>>>
>>
>>
>>
>>
>>Bought the 25$ kit yesterday.
>>
>>
>>Where do I download the connector ??
>>Is that in fs git repo ?? Or from vestec site ??  ( if so it's not enabled on my account ;( )
>>
>>
>>Also you need to wiki up some mad examples...
>>
>>
>>Can't wait to play though ....  
>>
>>
>>J
>>
>>
>>
>>
>>
>>On 22/04/2010, at 0:08, Kashif Kahn <info at evestech.com>
>> wrote:
>>
>>
>>Dear All,
>>>
>>>All those who have wanted to try speech recognition with Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, I encourage you to try our Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. Of course, the engine comes with a Freeswitch connector, thereby allowing a Freeswitch user to bypass engine API and interact directly via Dialplan.
>>>
>>>Best
>>> regards,
>>>-Kashif
>>>
>>> Kashif Kahn
>>>VP, Business Development
>>>Vestec,
>>> Inc.
>>>Waterloo, ON Canada
>>>phone: (519) 885-7615
>>>
>>>
>>_______________________________________________
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>-- 
>--
>Brent Paddon
>
>Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au | www.overthewire.com.au
>>
>
>
>Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From pjintheusa at gmail.com  Thu Apr 22 06:29:04 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 22 Apr 2010 09:29:04 -0400
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <p2t8ccbff061004220418vfe701a3ubec610a3de511811@mail.gmail.com>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
	<p2t8ccbff061004220418vfe701a3ubec610a3de511811@mail.gmail.com>
Message-ID: <n2y367751821004220629o223c4ebfv622800d0cbe9a8c2@mail.gmail.com>

No - it's just audio.

SipDroid is excellent - works great on my Moto Droid with FreeSWITCH. I set
the caller id to my cell number. Works over 3G so uses no carrier minutes.

On Thu, Apr 22, 2010 at 7:18 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> What's kind of android sip client ? Is it's work with video call ? (over
> sip)
> i found http://code.google.com/p/sipdroid/ but never test yet
>
>
> Dome C.
>
> 2010/4/21 Peter Olsson <peter.olsson at visionutveckling.se>:
> > Thanks,
> >
> >
> >
> > So the a way around this would be to;
> >
> > 1.       Exclude the phones from the acl list.
> >
> > 2.       Or ? use another sip profile for ?open? trunks, and disable the
> > inbound-acl on the internal profile?
> >
> >
> >
> > /Peter
> >
> >
> >
> >
> >
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
> > Skickat: den 21 april 2010 15:01
> > Till: freeswitch-users at lists.freeswitch.org
> > ?mne: Re: [Freeswitch-users] SIP Agent (Android SIP client)
> >
> >
> >
> > Correct.
> >
> >
> >
> > /b
> >
> >
> >
> > On Apr 21, 2010, at 6:23 AM, Peter Olsson wrote:
> >
> > Actually, I just noticed the same behaviour today in my lab system, when
> > dialing from a Polycom phone. Though in this case I have
> ?apply-inbound-acl?
> > set to allow access everything from my whole network (192.168.94.0/24),
> > could this cause the call fr?n Polycom ext not being treated as a
> > ?authenticated? call, and not apply the user_context to it?
> >
> >
> >
> > /Peter
> >
> >
> >
> > !DSPAM:4bcef85e32936933977704!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Thu Apr 22 06:42:26 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 22 Apr 2010 09:42:26 -0400
Subject: [Freeswitch-users] Windows Build of latest git version
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4CCF@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4CCF@cooper>
Message-ID: <q2r367751821004220642kbbc7d50cs220e2836d408b574@mail.gmail.com>

Jeff/Peter.

Thanks so much. I missed that - it is in the wiki so silly me.

Thanks again.

On Wed, Apr 21, 2010 at 11:38 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> When using git in Windows you must make sure to set core.autocrlf=false in
> the git configuration. The suggested way when installing is true, and it
> causes problems with some LF to CRLF conversions.
>
> /Peter
>
>
> ________________________________
> Fr?n: Phillip Jones <pjintheusa at gmail.com>
> Skickat: den 22 april 2010 00:16
> Till: freeswitch-users at lists.freeswitch.org <
> freeswitch-users at lists.freeswitch.org>
> ?mne: [Freeswitch-users] Windows Build of latest git version
>
> Hi All,
>
> I just downloaded the lasted version from git, and I am having real
> problems compiling in VS2008 for Windows. FreeSwithCoreLib, mod_PortAudio
> and others are fine, but when building libsofia_sip_ua_static I get 905
> errors, eg:
>
> Error    601    error C2065: 'sip_supported_hash' : undeclared identifier
>  c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c
>  279    libsofia_sip_ua_static
> Error    602    error C2099: initializer is not a constant
>  c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c
>  279    libsofia_sip_ua_static
>
> etc etc
>
> I have previously compiled successfully many times before, although not on
> this machine.
>
> Anyone else experience this and know how to fix?
>
> TIA
>
> Phil
> !DSPAM:4bcf792732932989516006!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Thu Apr 22 06:55:36 2010
From: boris at tagnet.ru (boris at tagnet.ru)
Date: Thu, 22 Apr 2010 19:55:36 +0600
Subject: [Freeswitch-users] mod_say or strftime problems?
In-Reply-To: <201004221512.18576.yivzhenko@mksat.net>
References: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>
	<201004221512.18576.yivzhenko@mksat.net>
Message-ID: <34adeff2737457cf22300f3b142145e5.squirrel@webmail.tagnet.ru>

IMHO this is bug, as of struct tm man page:
 int tm_hour;    /* hours (0 - 23) */
so tm_hour shouldn't be incremented. But looking at code I see no reason
why the behavior is changed with different timezone.

> I have same problem with mod_say_ru if timezone is set to
>
> <action application="set" data="timezone=Europe/Kiev"/>
>
> And i don't understand why  "tm.tm_hour + 1"
> In source code
>         if (say_time) {
>                 switch_snprintf(buf, sizeof(buf), "%d:%d:%d", tm.tm_hour +
> 1,
> tm.tm_min, tm.tm_sec);
>                 say_args->type = SST_TIME_MEASUREMENT;
>                 ru_say_time(session, buf, say_args, args);
>
> As temporary resolution i just set different timezone :)
>
>
> On Wednesday 21 April 2010 17:30:18 boris at tagnet.ru wrote:
>> Hello!
>>
>> I have an extension for the current date/time:
>>     <extension name="SayTime">
>>         <condition field="destination_number" expression="^9000$">
>>             <action application="answer"/>
>>             <action application="say" data="ru current_date_time
>> pronounced $strftime(%s)"/>
>>             <action application="sleep" data="1000"/>
>>             <action application="hangup"/>
>>         </condition>
>>     </extension>
>>
>> The extension works fine, but time is +1 hour of current time. For
>> example
>> current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6
>> (from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer
>> time in effect now, so tz=YEKST). So, my question is - something wrong
>> with my extension configuration or this is bug in mod_say_ru or may be
>> strfmt?
>>
>> With respect,
>> Boris
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From kenfulmer at icstechnologysolutions.com  Thu Apr 22 07:30:56 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 09:30:56 -0500
Subject: [Freeswitch-users] Different SIP profiles for different codecs?
Message-ID: <009001cae228$6c0e5bd0$442b1370$@com>

We are using the following external sip profile:

 

<profile name="external">

  <settings>

    <param name="accept-blind-auth" value="true"/>

    <param name="apply-inbound-acl" value="domains"/>

    <param name="debug" value="0"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="2000"/>

    <param name="inbound-codec-prefs" value="G729,PCMU"/>

    <param name="outbound-codec-prefs" value="G729,PCMU"/>

    <param name="hold-music" value="$${hold_music}"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="manage-presence" value="false"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="nonce-ttl" value="60"/>

    <param name="auth-calls" value="false"/>

    <param name="rtp-ip" value="10.10.3.12"/>

    <param name="sip-ip" value="10.10.3.12"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

    <param name="user-agent-string" value="INTEGRAL PARTNERS"/> 

  </settings>

</profile>

 

The dial plan "public" context has two entries, one for PSTN access and one
for call routing to an internal PBX. 

 

This is the dial plan for PSTN call routing:

 

<extension name="LCL">

   <condition field="destination_number" expression="^(\d{10})$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$1 at 172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="LD">

   <condition field="destination_number" expression="^1(\d{10})$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at 172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="911">

   <condition field="destination_number" expression="^(911)$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$1 at 172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="INTL">

   <condition field="destination_number" expression="^011(\d+)$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at 172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

 

And this is the entry for internal call routing to a PBX:

 

<extension name="PBX">

   <condition field="destination_number" expression="^(205314849[0-9])$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge"
data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true"/>

   </condition>

</extension>

 

So, here's my question:

 

We'd like to be able to lock down the codec as 711 for the internal leg
going to the PBX and 729 for the external leg to the PSTN. We have
transcoding setup and it's working fine. How can we use two SIP profiles  to
hard code the codec in each direction? I've seen an example in the dial plan
section, but didn't understand how to implement it.

 

Thanks,

 

Ken Fulmer

 

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From steveayre at gmail.com  Thu Apr 22 07:56:18 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 22 Apr 2010 15:56:18 +0100
Subject: [Freeswitch-users] Different SIP profiles for different codecs?
In-Reply-To: <009001cae228$6c0e5bd0$442b1370$@com>
References: <009001cae228$6c0e5bd0$442b1370$@com>
Message-ID: <n2ieafa772f1004220756lc51e8155jde50099b07f3fa71@mail.gmail.com>

No need to have separate profiles for it, these are the two parameters
you'd want to change:
<param name="inbound-codec-prefs" value="PCMU"/>
<param name="outbound-codec-prefs" value="G729"/>

If you want to have calls coming in one one profile (e.g. internal)
and going out on another (e.g. external) you can do so. Create both
profiles (if you haven't already) and set inbound-codec-prefs on
internal and outbound-codec-prefs on external. Have the internal
profile hit the dialplan context where you have the extensions
configured and have the dialplan bridge to sofia/external/... to send
the outgoing legs through that context.

-Steve


On 22 April 2010 15:30, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:
> We are using the following external sip profile:
>
>
>
> <profile name="external">
>
> ? <settings>
>
> ??? <param name="accept-blind-auth" value="true"/>
>
> ??? <param name="apply-inbound-acl" value="domains"/>
>
> ??? <param name="debug" value="0"/>
>
> ??? <param name="sip-trace" value="no"/>
>
> ??? <param name="rfc2833-pt" value="101"/>
>
> ??? <param name="sip-port" value="5080"/>
>
> ??? <param name="dialplan" value="XML"/>
>
> ??? <param name="context" value="public"/>
>
> ??? <param name="dtmf-duration" value="2000"/>
>
> ??? <param name="inbound-codec-prefs" value="G729,PCMU"/>
>
> ??? <param name="outbound-codec-prefs" value="G729,PCMU"/>
>
> ??? <param name="hold-music" value="$${hold_music}"/>
>
> ??? <param name="rtp-timer-name" value="soft"/>
>
> ??? <param name="manage-presence" value="false"/>
>
> ??? <param name="inbound-codec-negotiation" value="generous"/>
>
> ??? <param name="nonce-ttl" value="60"/>
>
> ??? <param name="auth-calls" value="false"/>
>
> ??? <param name="rtp-ip" value="10.10.3.12"/>
>
> ??? <param name="sip-ip" value="10.10.3.12"/>
>
> ??? <param name="rtp-timeout-sec" value="300"/>
>
> ??? <param name="rtp-hold-timeout-sec" value="1800"/>
>
> ??? <param name="user-agent-string" value="INTEGRAL PARTNERS"/>
>
> ? </settings>
>
> </profile>
>
>
>
> The dial plan ?public? context has two entries, one for PSTN access and one
> for call routing to an internal PBX.
>
>
>
> This is the dial plan for PSTN call routing:
>
>
>
> <extension name="LCL">
>
> ?? <condition field="destination_number" expression="^(\d{10})$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge" data="sofia/external/$1 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="LD">
>
> ?? <condition field="destination_number" expression="^1(\d{10})$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge" data="sofia/external/$0 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ????? ?<action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="911">
>
> ?? <condition field="destination_number" expression="^(911)$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge" data="sofia/external/$1 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="INTL">
>
> ?? <condition field="destination_number" expression="^011(\d+)$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge" data="sofia/external/$0 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
>
>
> And this is the entry for internal call routing to a PBX:
>
>
>
> <extension name="PBX">
>
> ?? <condition field="destination_number" expression="^(205314849[0-9])$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
> data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true"/>
>
> ?? </condition>
>
> </extension>
>
>
>
> So, here?s my question:
>
>
>
> We?d like to be able to lock down the codec as 711 for the internal leg
> going to the PBX and 729 for the external leg to the PSTN. We have
> transcoding setup and it?s working fine. How can we use two SIP profiles? to
> hard code the codec in each direction? I?ve seen an example in the dial plan
> section, but didn?t understand how to implement it.
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From kenfulmer at icstechnologysolutions.com  Thu Apr 22 08:14:27 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 10:14:27 -0500
Subject: [Freeswitch-users] Different SIP profiles for different codecs?
In-Reply-To: <n2ieafa772f1004220756lc51e8155jde50099b07f3fa71@mail.gmail.com>
References: <009001cae228$6c0e5bd0$442b1370$@com>
	<n2ieafa772f1004220756lc51e8155jde50099b07f3fa71@mail.gmail.com>
Message-ID: <00a101cae22e$80410c00$80c32400$@com>

I need the internal codec to be hardcoded to PCMU. I need the PSTN codec to
be 729, and I need to transcode between the two. 

So, we have PBX (711) ---> FS (729)---> PSTN and vice versa. 

If I only set the following parameters in one profile, I won't get the
necessary result:

<param name="inbound-codec-prefs" value="PCMU"/>
<param name="outbound-codec-prefs" value="G729"/>

We are routing calls through the system like a softswitch / B2BUA / SBC. We
don't have phones registering to the FS box like a PBX. We are using sipX
for that. 

So, we should be able to use two profiles and set the codecs differently in
each. This is show in the following example on the dial plan page:

Example 5

In this example we will demonstrate the use of profiles when using a
FreeSWITCH endpoint that supports profiles, like mod_sofia. Assuming that we
want to use different call settings (codecs, DTMF modes, etc) for sending
the calls to different IP addresses, we can create different profiles. For
example, in the configuration of sofia.conf, we see an example profile named
"test", which we rename to profile1 for this example, and add a profile2 for
comparison:

<profile name="profile1">
  <param name="debug" value="1"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-port" value="5060"/>
  <param name="dialplan" value="XML"/>
  <param name="dtmf-duration" value="100"/>
  <param name="codec-prefs" value="PCMU at 20i"/>
  <param name="codec-ms" value="20"/>
  <param name="use-rtp-timer" value="true"/>
</profile>
<profile name="profile2">
  <param name="debug" value="1"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-port" value="5070"/>
  <param name="dialplan" value="XML"/>
  <param name="dtmf-duration" value="100"/>
  <param name="codec-prefs" value="PCMA at 20i"/>
  <param name="codec-ms" value="20"/>
  <param name="use-rtp-timer" value="true"/>
</profile>

The difference between the two profiles are in the codecs. The first uses
G.711 uLaw and the second G711 ALaw.

Continuing the examples above, we have:

<extension name="Test5ulaw">
  <condition field="network_addr" expression="^192\.168\.1\.1$"/>
  <condition field="destination_number" expression="^1(\d+)$">
    <action application="bridge" data="sofia/profile1/$0 at 192.168.2.2"/>
  </condition>
</extension>

to send the call in G.711 uLaw and

<extension name="Test5alaw">
  <condition field="network_addr" expression="^192\.168\.1\.1$"/>
  <condition field="destination_number" expression="^1(\d+)$">
    <action application="bridge" data="sofia/profile2/$0 at 192.168.2.2"/>
  </condition>
</extension>

But when we try to setup two profiles, we always get an error on one of
them. 

Ken


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, April 22, 2010 9:56 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Different SIP profiles for different codecs?

No need to have separate profiles for it, these are the two parameters
you'd want to change:
<param name="inbound-codec-prefs" value="PCMU"/>
<param name="outbound-codec-prefs" value="G729"/>

If you want to have calls coming in one one profile (e.g. internal)
and going out on another (e.g. external) you can do so. Create both
profiles (if you haven't already) and set inbound-codec-prefs on
internal and outbound-codec-prefs on external. Have the internal
profile hit the dialplan context where you have the extensions
configured and have the dialplan bridge to sofia/external/... to send
the outgoing legs through that context.

-Steve


On 22 April 2010 15:30, Ken Fulmer <kenfulmer at icstechnologysolutions.com>
wrote:
> We are using the following external sip profile:
>
>
>
> <profile name="external">
>
> ? <settings>
>
> ??? <param name="accept-blind-auth" value="true"/>
>
> ??? <param name="apply-inbound-acl" value="domains"/>
>
> ??? <param name="debug" value="0"/>
>
> ??? <param name="sip-trace" value="no"/>
>
> ??? <param name="rfc2833-pt" value="101"/>
>
> ??? <param name="sip-port" value="5080"/>
>
> ??? <param name="dialplan" value="XML"/>
>
> ??? <param name="context" value="public"/>
>
> ??? <param name="dtmf-duration" value="2000"/>
>
> ??? <param name="inbound-codec-prefs" value="G729,PCMU"/>
>
> ??? <param name="outbound-codec-prefs" value="G729,PCMU"/>
>
> ??? <param name="hold-music" value="$${hold_music}"/>
>
> ??? <param name="rtp-timer-name" value="soft"/>
>
> ??? <param name="manage-presence" value="false"/>
>
> ??? <param name="inbound-codec-negotiation" value="generous"/>
>
> ??? <param name="nonce-ttl" value="60"/>
>
> ??? <param name="auth-calls" value="false"/>
>
> ??? <param name="rtp-ip" value="10.10.3.12"/>
>
> ??? <param name="sip-ip" value="10.10.3.12"/>
>
> ??? <param name="rtp-timeout-sec" value="300"/>
>
> ??? <param name="rtp-hold-timeout-sec" value="1800"/>
>
> ??? <param name="user-agent-string" value="INTEGRAL PARTNERS"/>
>
> ? </settings>
>
> </profile>
>
>
>
> The dial plan ?public? context has two entries, one for PSTN access and
one
> for call routing to an internal PBX.
>
>
>
> This is the dial plan for PSTN call routing:
>
>
>
> <extension name="LCL">
>
> ?? <condition field="destination_number" expression="^(\d{10})$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
data="sofia/external/$1 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="LD">
>
> ?? <condition field="destination_number" expression="^1(\d{10})$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
data="sofia/external/$0 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ????? ?<action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="911">
>
> ?? <condition field="destination_number" expression="^(911)$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
data="sofia/external/$1 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
> <extension name="INTL">
>
> ?? <condition field="destination_number" expression="^011(\d+)$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
data="sofia/external/$0 at 172.16.15.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true" />
>
> ?? </condition>
>
> </extension>
>
>
>
> And this is the entry for internal call routing to a PBX:
>
>
>
> <extension name="PBX">
>
> ?? <condition field="destination_number" expression="^(205314849[0-9])$">
>
> ?????? <action application="set" data="hangup_after_bridge=true"/>
>
> ?????? <action application="bridge"
> data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11"/>
>
> ?????? <action application="set" data="ringback=${us-ring}"/>
>
> ?????? <action application="set" data="ignore_early_media=true"/>
>
> ?? </condition>
>
> </extension>
>
>
>
> So, here?s my question:
>
>
>
> We?d like to be able to lock down the codec as 711 for the internal leg
> going to the PBX and 729 for the external leg to the PSTN. We have
> transcoding setup and it?s working fine. How can we use two SIP profiles?
to
> hard code the codec in each direction? I?ve seen an example in the dial
plan
> section, but didn?t understand how to implement it.
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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



From dome at tel.co.th  Thu Apr 22 08:14:33 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 22 Apr 2010 22:14:33 +0700
Subject: [Freeswitch-users] SIP Agent (Android SIP client)
In-Reply-To: <n2y367751821004220629o223c4ebfv622800d0cbe9a8c2@mail.gmail.com>
References: <o2t367751821004191056i308e491av17a88ef4d24492d6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E76CC@cooper>
	<A1FD3208-F126-4757-A872-71F0A0FCE545@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C55776E776A@cooper>
	<p2t8ccbff061004220418vfe701a3ubec610a3de511811@mail.gmail.com>
	<n2y367751821004220629o223c4ebfv622800d0cbe9a8c2@mail.gmail.com>
Message-ID: <g2u8ccbff061004220814h910513arf9562767c45b5d75@mail.gmail.com>

I found many issue about video in sipdroid but not clear it's ready or not.
http://code.google.com/p/sipdroid/issues/list?can=2&q=video&colspec=ID+Type+Status+Priority+Milestone+Owner+Summary&cells=tiles

If you can please test video call to x-lite.

BG

Dome C.

2010/4/22 Phillip Jones <pjintheusa at gmail.com>:
> No - it's just audio.
>
> SipDroid is excellent - works great on my Moto Droid with FreeSWITCH. I set
> the caller id to my cell number. Works over 3G so uses no carrier minutes.
>
> On Thu, Apr 22, 2010 at 7:18 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> What's kind of android sip client ? Is it's work with video call ? (over
>> sip)
>> i found http://code.google.com/p/sipdroid/ but never test yet
>>
>>
>> Dome C.
>>
>> 2010/4/21 Peter Olsson <peter.olsson at visionutveckling.se>:
>> > Thanks,
>> >
>> >
>> >
>> > So the a way around this would be to;
>> >
>> > 1.?????? Exclude the phones from the acl list.
>> >
>> > 2.?????? Or ? use another sip profile for ?open? trunks, and disable the
>> > inbound-acl on the internal profile?
>> >
>> >
>> >
>> > /Peter
>> >
>> >
>> >
>> >
>> >
>> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
>> > Skickat: den 21 april 2010 15:01
>> > Till: freeswitch-users at lists.freeswitch.org
>> > ?mne: Re: [Freeswitch-users] SIP Agent (Android SIP client)
>> >
>> >
>> >
>> > Correct.
>> >
>> >
>> >
>> > /b
>> >
>> >
>> >
>> > On Apr 21, 2010, at 6:23 AM, Peter Olsson wrote:
>> >
>> > Actually, I just noticed the same behaviour today in my lab system, when
>> > dialing from a Polycom phone. Though in this case I have
>> > ?apply-inbound-acl?
>> > set to allow access everything from my whole network (192.168.94.0/24),
>> > could this cause the call fr?n Polycom ext not being treated as a
>> > ?authenticated? call, and not apply the user_context to it?
>> >
>> >
>> >
>> > /Peter
>> >
>> >
>> >
>> > !DSPAM:4bcef85e32936933977704!
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 22 08:33:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Apr 2010 10:33:38 -0500
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <4BCFEAD7.7050902@infosecurity.ch>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
	<4BCFEAD7.7050902@infosecurity.ch>
Message-ID: <v2j191c3a031004220833r51ebad1am410665794c3ed1ac@mail.gmail.com>

Are you arrogantly refusing to collect the actual sip traces but still feel
the need to report an issue?
That is called FUD and we do not tolerate it here.  Either hold your tongues
or collect the info.  Do not just use our mailing list to say you have
"random and not reproducible problems"  It does nobody any good.  We have
now wasted 3 email exchanges simply asking for the information.  If you were
able to produce the ladder diagram surely you could have done what I asked
and did a pcap from the FS box and a console trace.  Maybe you should learn
more about network topology before you send us any more email.


On Thu, Apr 22, 2010 at 1:21 AM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> Even if i have not done a so in depth analysis, i have several random
> and not reproduceable behaviour (but continuous) using FS with SIP/TLS
> (i only use it over SIP/TLS) that seems to have a representation of the
> issue equal to this one reported by Emil.
>
> Fabio
>
> On 21/04/10 21.05, Emil Berg wrote:
> > Hello,
> >
> > I'm not sure that this is the right mailing list for this question,
> > but I'll ask anyway :)
> > Are there any known issues with sip over tcp?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Thu Apr 22 08:45:28 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 22 Apr 2010 17:45:28 +0200
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <v2j191c3a031004220833r51ebad1am410665794c3ed1ac@mail.gmail.com>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>	<4BCFEAD7.7050902@infosecurity.ch>
	<v2j191c3a031004220833r51ebad1am410665794c3ed1ac@mail.gmail.com>
Message-ID: <4BD06F18.5050808@infosecurity.ch>

Hi Anthony,
i really apologise for not being still able to provide the trace, i am 
under product release and this is an issue of the many to be managed and 
it's in my company backlog.

I did not wanted to make FUD or being arrogant, really.

It's just that i read an issue in the mailing list similar to what i 
experience.
We still not analyzed deeply and you know that a non deterministic bug 
it's something quite difficult to be traced down, require a continuous 
testing environment to catch the issue while collecting a lot of logs :(

Will do it, just is in a schedule of activities.

Fabio
On 22/04/10 17.33, Anthony Minessale wrote:
> Are you arrogantly refusing to collect the actual sip traces but still 
> feel the need to report an issue?
> That is called FUD and we do not tolerate it here.  Either hold your 
> tongues or collect the info.  Do not just use our mailing list to say 
> you have "random and not reproducible problems"  It does nobody any 
> good.  We have now wasted 3 email exchanges simply asking for the 
> information.  If you were able to produce the ladder diagram surely 
> you could have done what I asked and did a pcap from the FS box and a 
> console trace.  Maybe you should learn more about network topology 
> before you send us any more email.


From wchao at yahoo.com  Thu Apr 22 09:43:13 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Thu, 22 Apr 2010 12:43:13 -0400 (EDT)
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
Message-ID: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>

I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now 
I see a strange behavior with caller ID on Polycom phones when handling 
inbound calls.

Here is the scenario:

* Call is from 212-555-2222 (external number not on my softswitch)

* Call is to 212-555-1001 (number on my softswitch, extension 1001)

* extension 1001 is a Polycom phone (it's an IP301, but same problem
   occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
   SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.

* on phone 1001, the caller ID will display 212-555-2222 while ringing.
   The moment I pick up, the display will change to "From: 1001" (referring
   to the extension of the phone itself).

Has anyone else experienced this problem, and does anyone know how to fix 
it? It does not occur with the snom 320 (and I assume it does not occur 
with any of the snom models based on extrapolation).

While the problem only started when I updated from 1.0.4 to 1.0.6, it's 
entirely possible it's a configuration setting on the Polycom rather than 
a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to 
resolve the problem (or even how to go about investigating the root 
cause)?


From djbinter at gmail.com  Thu Apr 22 08:32:27 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Thu, 22 Apr 2010 08:32:27 -0700
Subject: [Freeswitch-users] Different SIP profiles for different codecs?
In-Reply-To: <00a101cae22e$80410c00$80c32400$@com>
References: <009001cae228$6c0e5bd0$442b1370$@com>
	<n2ieafa772f1004220756lc51e8155jde50099b07f3fa71@mail.gmail.com>
	<00a101cae22e$80410c00$80c32400$@com>
Message-ID: <y2z94f7dfb11004220832tf4c9757eif567a83e8d3b413c@mail.gmail.com>

<action application="bridge"
data="{absolute_codec_string=G729}sofia/external//$0@ <0 at 172.16.15.11>
xxx.xxx.xxx.xxx"/>

-djbinter

On Thu, Apr 22, 2010 at 8:14 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

> I need the internal codec to be hardcoded to PCMU. I need the PSTN codec to
> be 729, and I need to transcode between the two.
>
> So, we have PBX (711) ---> FS (729)---> PSTN and vice versa.
>
> If I only set the following parameters in one profile, I won't get the
> necessary result:
>
> <param name="inbound-codec-prefs" value="PCMU"/>
> <param name="outbound-codec-prefs" value="G729"/>
>
> We are routing calls through the system like a softswitch / B2BUA / SBC. We
> don't have phones registering to the FS box like a PBX. We are using sipX
> for that.
>
> So, we should be able to use two profiles and set the codecs differently in
> each. This is show in the following example on the dial plan page:
>
> Example 5
>
> In this example we will demonstrate the use of profiles when using a
> FreeSWITCH endpoint that supports profiles, like mod_sofia. Assuming that
> we
> want to use different call settings (codecs, DTMF modes, etc) for sending
> the calls to different IP addresses, we can create different profiles. For
> example, in the configuration of sofia.conf, we see an example profile
> named
> "test", which we rename to profile1 for this example, and add a profile2
> for
> comparison:
>
> <profile name="profile1">
>  <param name="debug" value="1"/>
>   <param name="rfc2833-pt" value="101"/>
>   <param name="sip-port" value="5060"/>
>   <param name="dialplan" value="XML"/>
>   <param name="dtmf-duration" value="100"/>
>  <param name="codec-prefs" value="PCMU at 20i"/>
>  <param name="codec-ms" value="20"/>
>  <param name="use-rtp-timer" value="true"/>
> </profile>
> <profile name="profile2">
>  <param name="debug" value="1"/>
>   <param name="rfc2833-pt" value="101"/>
>   <param name="sip-port" value="5070"/>
>   <param name="dialplan" value="XML"/>
>   <param name="dtmf-duration" value="100"/>
>  <param name="codec-prefs" value="PCMA at 20i"/>
>  <param name="codec-ms" value="20"/>
>  <param name="use-rtp-timer" value="true"/>
> </profile>
>
> The difference between the two profiles are in the codecs. The first uses
> G.711 uLaw and the second G711 ALaw.
>
> Continuing the examples above, we have:
>
> <extension name="Test5ulaw">
>  <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>  <condition field="destination_number" expression="^1(\d+)$">
>    <action application="bridge" data="sofia/profile1/$0 at 192.168.2.2"/>
>  </condition>
> </extension>
>
> to send the call in G.711 uLaw and
>
> <extension name="Test5alaw">
>  <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>  <condition field="destination_number" expression="^1(\d+)$">
>    <action application="bridge" data="sofia/profile2/$0 at 192.168.2.2"/>
>  </condition>
> </extension>
>
> But when we try to setup two profiles, we always get an error on one of
> them.
>
> Ken
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
> Ayre
> Sent: Thursday, April 22, 2010 9:56 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Different SIP profiles for different
> codecs?
>
> No need to have separate profiles for it, these are the two parameters
> you'd want to change:
> <param name="inbound-codec-prefs" value="PCMU"/>
> <param name="outbound-codec-prefs" value="G729"/>
>
> If you want to have calls coming in one one profile (e.g. internal)
> and going out on another (e.g. external) you can do so. Create both
> profiles (if you haven't already) and set inbound-codec-prefs on
> internal and outbound-codec-prefs on external. Have the internal
> profile hit the dialplan context where you have the extensions
> configured and have the dialplan bridge to sofia/external/... to send
> the outgoing legs through that context.
>
> -Steve
>
>
> On 22 April 2010 15:30, Ken Fulmer <kenfulmer at icstechnologysolutions.com>
> wrote:
> > We are using the following external sip profile:
> >
> >
> >
> > <profile name="external">
> >
> >   <settings>
> >
> >     <param name="accept-blind-auth" value="true"/>
> >
> >     <param name="apply-inbound-acl" value="domains"/>
> >
> >     <param name="debug" value="0"/>
> >
> >     <param name="sip-trace" value="no"/>
> >
> >     <param name="rfc2833-pt" value="101"/>
> >
> >     <param name="sip-port" value="5080"/>
> >
> >     <param name="dialplan" value="XML"/>
> >
> >     <param name="context" value="public"/>
> >
> >     <param name="dtmf-duration" value="2000"/>
> >
> >     <param name="inbound-codec-prefs" value="G729,PCMU"/>
> >
> >     <param name="outbound-codec-prefs" value="G729,PCMU"/>
> >
> >     <param name="hold-music" value="$${hold_music}"/>
> >
> >     <param name="rtp-timer-name" value="soft"/>
> >
> >     <param name="manage-presence" value="false"/>
> >
> >     <param name="inbound-codec-negotiation" value="generous"/>
> >
> >     <param name="nonce-ttl" value="60"/>
> >
> >     <param name="auth-calls" value="false"/>
> >
> >     <param name="rtp-ip" value="10.10.3.12"/>
> >
> >     <param name="sip-ip" value="10.10.3.12"/>
> >
> >     <param name="rtp-timeout-sec" value="300"/>
> >
> >     <param name="rtp-hold-timeout-sec" value="1800"/>
> >
> >     <param name="user-agent-string" value="INTEGRAL PARTNERS"/>
> >
> >   </settings>
> >
> > </profile>
> >
> >
> >
> > The dial plan ?public? context has two entries, one for PSTN access and
> one
> > for call routing to an internal PBX.
> >
> >
> >
> > This is the dial plan for PSTN call routing:
> >
> >
> >
> > <extension name="LCL">
> >
> >    <condition field="destination_number" expression="^(\d{10})$">
> >
> >        <action application="set" data="hangup_after_bridge=true"/>
> >
> >        <action application="bridge"
> data="sofia/external/$1 at 172.16.15.11"/>
> >
> >        <action application="set" data="ringback=${us-ring}"/>
> >
> >        <action application="set" data="ignore_early_media=true" />
> >
> >    </condition>
> >
> > </extension>
> >
> > <extension name="LD">
> >
> >    <condition field="destination_number" expression="^1(\d{10})$">
> >
> >        <action application="set" data="hangup_after_bridge=true"/>
> >
> >        <action application="bridge"
> data="sofia/external/$0 at 172.16.15.11"/>
> >
> >        <action application="set" data="ringback=${us-ring}"/>
> >
> >        <action application="set" data="ignore_early_media=true" />
> >
> >    </condition>
> >
> > </extension>
> >
> > <extension name="911">
> >
> >    <condition field="destination_number" expression="^(911)$">
> >
> >        <action application="set" data="hangup_after_bridge=true"/>
> >
> >        <action application="bridge"
> data="sofia/external/$1 at 172.16.15.11"/>
> >
> >        <action application="set" data="ringback=${us-ring}"/>
> >
> >        <action application="set" data="ignore_early_media=true" />
> >
> >    </condition>
> >
> > </extension>
> >
> > <extension name="INTL">
> >
> >    <condition field="destination_number" expression="^011(\d+)$">
> >
> >        <action application="set" data="hangup_after_bridge=true"/>
> >
> >        <action application="bridge"
> data="sofia/external/$0 at 172.16.15.11"/>
> >
> >        <action application="set" data="ringback=${us-ring}"/>
> >
> >        <action application="set" data="ignore_early_media=true" />
> >
> >    </condition>
> >
> > </extension>
> >
> >
> >
> > And this is the entry for internal call routing to a PBX:
> >
> >
> >
> > <extension name="PBX">
> >
> >    <condition field="destination_number" expression="^(205314849[0-9])$">
> >
> >        <action application="set" data="hangup_after_bridge=true"/>
> >
> >        <action application="bridge"
> > data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11
> "/>
> >
> >        <action application="set" data="ringback=${us-ring}"/>
> >
> >        <action application="set" data="ignore_early_media=true"/>
> >
> >    </condition>
> >
> > </extension>
> >
> >
> >
> > So, here?s my question:
> >
> >
> >
> > We?d like to be able to lock down the codec as 711 for the internal leg
> > going to the PBX and 729 for the external leg to the PSTN. We have
> > transcoding setup and it?s working fine. How can we use two SIP profiles
> to
> > hard code the codec in each direction? I?ve seen an example in the dial
> plan
> > section, but didn?t understand how to implement it.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Ken Fulmer
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jfirles at bio.es  Thu Apr 22 09:46:47 2010
From: jfirles at bio.es (Jose Fco. Irles)
Date: Thu, 22 Apr 2010 18:46:47 +0200
Subject: [Freeswitch-users] Manage call from Event Socket inbound
Message-ID: <4BD07D77.4080200@bio.es>

Hi,

I would like to manage calls from the Event Socket in inbound mode, 
connecting to freeswitch and waiting for events and sending commands.
I don't know how to sleep the call until my logic executes a action in 
this new call.

I've tried to park the call with this in the dialplan:

    <extension name="my_park">
      <condition field="destination_number" expression="^.+$">
        <action application="park"/>
      </condition>
    </extension>

but freeswitch "pre answers" the call and my sip client receives a "183 
Session progress". I want the the call to wait in "100 Trying" until my 
logic execute something for the new call.

In outbound mode this works well, when the dialplan executes the 
"socket" application, the call waits in "100 Trying".

I prefer to make this in inbound mode because it's more scalable and 
simple than build a tcp server and manage the calls in outbound mode. 
However in outbound mode I need inbound mode for some things.
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From dome at tel.co.th  Thu Apr 22 11:03:05 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 23 Apr 2010 01:03:05 +0700
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
	<b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
Message-ID: <w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>

Hi Rupa,

   any update about NoSQL module. Now i'm modify mod_limit change sql
to tokyo tyrant. it's work fine for me. now i can share limit data to
other FS server. Now i'm thinking about all fs db. if use redis
(sorted sets)  i think posible todo. but it's not easy to modify fs
code.

So i need comment about aventage if change fs backend froom SQL to NoSQL.

BG

Dome C.


2010/3/4 Rupa Schomaker <rupa at rupa.com>:
> On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman
> <freeswitch at cartissolutions.com> wrote:
>>
>>
>> Maybe it would be even more useful to provide general functionality that
>> can be shared amongst multiple components of freeswitch. ?That would
>> make it all the more useful. ?However, seeing that I'm not a big fan of
>> the NoSQL data engines, I don't know if this is possible. ?In my
>> opinion, most NoSQL is just key=>value pairs, and is basically a
>> reinvention of the wheel that RDBMS's had taken care of years ago...
>> Not meaning to open up a discussion about the virtues of NoSQL (and
>> certainly not a flamewar)...
>>
>
> I created a mod_memcache already and I intend to do a mod_redis at some
> point. ?I'm not a huge fan of NoSQL but there are definite uses for it.
>
> General functionality: ?I could be convinced to come up with a general api
> for distributed key/value store with mod_memcache and mod_redis providing
> implementation. ?The problem is that the NoSQL stuff isn't very consistent
> so even though both support key/value their behavior can significantly
> differ. ?eg: updating a value in memcached doesn't bump the expire time but
> does in redis. ?Also, redis has support for a much more robust set of value
> types and operators.
> Anyway, a generic distributed key/value api might look like:
> dhash <backend> set key value [expire]
> dhash <backend> setnx key value [expire] ?# only set if it doesn't already
> exist
> dhash <backend> get key [...] # support multiple keys
> dhash <backend> <increment|decrement> key [step]
> dhash <backend> del key
> where backend would be memcache or redis or some other implementation.
> Notice I didn't even touch things like hashing to support sharding,
> failover, etc.
>
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mrene_lists at avgs.ca  Thu Apr 22 11:06:27 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 22 Apr 2010 14:06:27 -0400
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
	<b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
	<w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
Message-ID: <247D9598-335D-4DD9-9A53-D71E8D774DEB@avgs.ca>

I have mod_redis ready, it'll go in soon.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-22, at 2:03 PM, Dome Charoenyost wrote:

> Hi Rupa,
> 
>   any update about NoSQL module. Now i'm modify mod_limit change sql
> to tokyo tyrant. it's work fine for me. now i can share limit data to
> other FS server. Now i'm thinking about all fs db. if use redis
> (sorted sets)  i think posible todo. but it's not easy to modify fs
> code.
> 
> So i need comment about aventage if change fs backend froom SQL to NoSQL.
> 
> BG
> 
> Dome C.
> 
> 
> 2010/3/4 Rupa Schomaker <rupa at rupa.com>:
>> On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman
>> <freeswitch at cartissolutions.com> wrote:
>>> 
>>> 
>>> Maybe it would be even more useful to provide general functionality that
>>> can be shared amongst multiple components of freeswitch.  That would
>>> make it all the more useful.  However, seeing that I'm not a big fan of
>>> the NoSQL data engines, I don't know if this is possible.  In my
>>> opinion, most NoSQL is just key=>value pairs, and is basically a
>>> reinvention of the wheel that RDBMS's had taken care of years ago...
>>> Not meaning to open up a discussion about the virtues of NoSQL (and
>>> certainly not a flamewar)...
>>> 
>> 
>> I created a mod_memcache already and I intend to do a mod_redis at some
>> point.  I'm not a huge fan of NoSQL but there are definite uses for it.
>> 
>> General functionality:  I could be convinced to come up with a general api
>> for distributed key/value store with mod_memcache and mod_redis providing
>> implementation.  The problem is that the NoSQL stuff isn't very consistent
>> so even though both support key/value their behavior can significantly
>> differ.  eg: updating a value in memcached doesn't bump the expire time but
>> does in redis.  Also, redis has support for a much more robust set of value
>> types and operators.
>> Anyway, a generic distributed key/value api might look like:
>> dhash <backend> set key value [expire]
>> dhash <backend> setnx key value [expire]  # only set if it doesn't already
>> exist
>> dhash <backend> get key [...] # support multiple keys
>> dhash <backend> <increment|decrement> key [step]
>> dhash <backend> del key
>> where backend would be memcache or redis or some other implementation.
>> Notice I didn't even touch things like hashing to support sharding,
>> failover, etc.
>> 
>>> 
>>> Yossi Neiman
>>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>> 
>> 
>> --
>> -Rupa
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kenfulmer at icstechnologysolutions.com  Thu Apr 22 11:08:17 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 13:08:17 -0500
Subject: [Freeswitch-users] Different SIP profiles for different codecs?
In-Reply-To: <y2z94f7dfb11004220832tf4c9757eif567a83e8d3b413c@mail.gmail.com>
References: <009001cae228$6c0e5bd0$442b1370$@com>	<n2ieafa772f1004220756lc51e8155jde50099b07f3fa71@mail.gmail.com>	<00a101cae22e$80410c00$80c32400$@com>
	<y2z94f7dfb11004220832tf4c9757eif567a83e8d3b413c@mail.gmail.com>
Message-ID: <00dd01cae246$c91a42d0$5b4ec870$@com>

Thanks, I just noticed that in the dial plan config. It works great!

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of DJB
INTERNATIONAL
Sent: Thursday, April 22, 2010 10:32 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Different SIP profiles for different codecs?

 

 

<action application="bridge"
data="{absolute_codec_string=G729}sofia/external//$0@
<mailto:0 at 172.16.15.11> xxx.xxx.xxx.xxx"/>

 

-djbinter

On Thu, Apr 22, 2010 at 8:14 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

I need the internal codec to be hardcoded to PCMU. I need the PSTN codec to
be 729, and I need to transcode between the two.

So, we have PBX (711) ---> FS (729)---> PSTN and vice versa.

If I only set the following parameters in one profile, I won't get the
necessary result:


<param name="inbound-codec-prefs" value="PCMU"/>
<param name="outbound-codec-prefs" value="G729"/>

We are routing calls through the system like a softswitch / B2BUA / SBC. We
don't have phones registering to the FS box like a PBX. We are using sipX
for that.

So, we should be able to use two profiles and set the codecs differently in
each. This is show in the following example on the dial plan page:

Example 5

In this example we will demonstrate the use of profiles when using a
FreeSWITCH endpoint that supports profiles, like mod_sofia. Assuming that we
want to use different call settings (codecs, DTMF modes, etc) for sending
the calls to different IP addresses, we can create different profiles. For
example, in the configuration of sofia.conf, we see an example profile named
"test", which we rename to profile1 for this example, and add a profile2 for
comparison:

<profile name="profile1">
 <param name="debug" value="1"/>

 <param name="rfc2833-pt" value="101"/>

 <param name="sip-port" value="5060"/>

 <param name="dialplan" value="XML"/>

 <param name="dtmf-duration" value="100"/>
 <param name="codec-prefs" value="PCMU at 20i"/>
 <param name="codec-ms" value="20"/>
 <param name="use-rtp-timer" value="true"/>
</profile>
<profile name="profile2">
 <param name="debug" value="1"/>

 <param name="rfc2833-pt" value="101"/>

 <param name="sip-port" value="5070"/>

 <param name="dialplan" value="XML"/>

 <param name="dtmf-duration" value="100"/>
 <param name="codec-prefs" value="PCMA at 20i"/>
 <param name="codec-ms" value="20"/>
 <param name="use-rtp-timer" value="true"/>
</profile>

The difference between the two profiles are in the codecs. The first uses
G.711 uLaw and the second G711 ALaw.

Continuing the examples above, we have:

<extension name="Test5ulaw">
 <condition field="network_addr" expression="^192\.168\.1\.1$"/>
 <condition field="destination_number" expression="^1(\d+)$">
   <action application="bridge" data="sofia/profile1/$0 at 192.168.2.2"/>
 </condition>
</extension>

to send the call in G.711 uLaw and

<extension name="Test5alaw">
 <condition field="network_addr" expression="^192\.168\.1\.1$"/>
 <condition field="destination_number" expression="^1(\d+)$">
   <action application="bridge" data="sofia/profile2/$0 at 192.168.2.2"/>
 </condition>
</extension>

But when we try to setup two profiles, we always get an error on one of
them.

Ken



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, April 22, 2010 9:56 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Different SIP profiles for different codecs?

No need to have separate profiles for it, these are the two parameters
you'd want to change:
<param name="inbound-codec-prefs" value="PCMU"/>
<param name="outbound-codec-prefs" value="G729"/>

If you want to have calls coming in one one profile (e.g. internal)
and going out on another (e.g. external) you can do so. Create both
profiles (if you haven't already) and set inbound-codec-prefs on
internal and outbound-codec-prefs on external. Have the internal
profile hit the dialplan context where you have the extensions
configured and have the dialplan bridge to sofia/external/... to send
the outgoing legs through that context.

-Steve


On 22 April 2010 15:30, Ken Fulmer <kenfulmer at icstechnologysolutions.com>
wrote:
> We are using the following external sip profile:
>
>
>
> <profile name="external">
>
>   <settings>
>
>     <param name="accept-blind-auth" value="true"/>
>
>     <param name="apply-inbound-acl" value="domains"/>
>
>     <param name="debug" value="0"/>
>
>     <param name="sip-trace" value="no"/>
>
>     <param name="rfc2833-pt" value="101"/>
>
>     <param name="sip-port" value="5080"/>
>
>     <param name="dialplan" value="XML"/>
>
>     <param name="context" value="public"/>
>
>     <param name="dtmf-duration" value="2000"/>
>
>     <param name="inbound-codec-prefs" value="G729,PCMU"/>
>
>     <param name="outbound-codec-prefs" value="G729,PCMU"/>
>
>     <param name="hold-music" value="$${hold_music}"/>
>
>     <param name="rtp-timer-name" value="soft"/>
>
>     <param name="manage-presence" value="false"/>
>
>     <param name="inbound-codec-negotiation" value="generous"/>
>
>     <param name="nonce-ttl" value="60"/>
>
>     <param name="auth-calls" value="false"/>
>
>     <param name="rtp-ip" value="10.10.3.12"/>
>
>     <param name="sip-ip" value="10.10.3.12"/>
>
>     <param name="rtp-timeout-sec" value="300"/>
>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>
>     <param name="user-agent-string" value="INTEGRAL PARTNERS"/>
>
>   </settings>
>
> </profile>
>
>
>
> The dial plan "public" context has two entries, one for PSTN access and
one
> for call routing to an internal PBX.
>
>
>
> This is the dial plan for PSTN call routing:
>
>
>
> <extension name="LCL">
>
>    <condition field="destination_number" expression="^(\d{10})$">
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge"
data="sofia/external/$1 at 172.16.15.11"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>        <action application="set" data="ignore_early_media=true" />
>
>    </condition>
>
> </extension>
>
> <extension name="LD">
>
>    <condition field="destination_number" expression="^1(\d{10})$">
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge"
data="sofia/external/$0 at 172.16.15.11"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>        <action application="set" data="ignore_early_media=true" />
>
>    </condition>
>
> </extension>
>
> <extension name="911">
>
>    <condition field="destination_number" expression="^(911)$">
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge"
data="sofia/external/$1 at 172.16.15.11"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>        <action application="set" data="ignore_early_media=true" />
>
>    </condition>
>
> </extension>
>
> <extension name="INTL">
>
>    <condition field="destination_number" expression="^011(\d+)$">
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge"
data="sofia/external/$0 at 172.16.15.11"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>        <action application="set" data="ignore_early_media=true" />
>
>    </condition>
>
> </extension>
>
>
>
> And this is the entry for internal call routing to a PBX:
>
>
>
> <extension name="PBX">
>
>    <condition field="destination_number" expression="^(205314849[0-9])$">
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge"
> data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>        <action application="set" data="ignore_early_media=true"/>
>
>    </condition>
>
> </extension>
>
>
>
> So, here's my question:
>
>
>
> We'd like to be able to lock down the codec as 711 for the internal leg
> going to the PBX and 729 for the external leg to the PSTN. We have
> transcoding setup and it's working fine. How can we use two SIP profiles 
to
> hard code the codec in each direction? I've seen an example in the dial
plan
> section, but didn't understand how to implement it.
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


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

 

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From kenfulmer at icstechnologysolutions.com  Thu Apr 22 11:24:05 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 13:24:05 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
Message-ID: <00eb01cae248$fe1f7e80$fa5e7b80$@com>

1.       Are these licenses additive? In other words, if I've purchased one
for a machine and need a total of five, can I purchase four more and add
them to the mix?

2.       If we have to replace a machine, how can we replace the licenses
without repurchasing them? 

 

Thanks,

 

Ken Fulmer

 

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From brian at freeswitch.org  Thu Apr 22 11:30:01 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 13:30:01 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
Message-ID: <5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>

If you have to replace the machine you'll have to send me the proof the machine is dead and I can reset them... I can only do this 3 times before you have to repurchase the license.

/b

On Apr 22, 2010, at 1:24 PM, Ken Fulmer wrote:

> 1.       Are these licenses additive? In other words, if I?ve purchased one for a machine and need a total of five, can I purchase four more and add them to the mix?
> 2.       If we have to replace a machine, how can we replace the licenses without repurchasing them?
>  
> Thanks,
>  
> Ken Fulmer
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ranjtech at gmail.com  Thu Apr 22 12:15:45 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 22 Apr 2010 15:15:45 -0400
Subject: [Freeswitch-users] reloadxml not reloading gateway configuration
In-Reply-To: <n2ieafa772f1004220025xa5c9a2bfvf5b023821b38a9a2@mail.gmail.com>
References: <h2ndcde51f21004211328j362b2981od4f6c06d11648304@mail.gmail.com>
	<26B9CD23-4E1D-403F-B345-5B49FA7C13E1@gmail.com>
	<n2ieafa772f1004220025xa5c9a2bfvf5b023821b38a9a2@mail.gmail.com>
Message-ID: <z2hdcde51f21004221215jec4677a1i91b9796860c2c18e@mail.gmail.com>

Hi Steve,

That worked beautifully. Thanks so much

Cheers
RR

On Thu, Apr 22, 2010 at 3:25 AM, Steven Ayre <steveayre at gmail.com> wrote:

> RR,
>
> What David's given you will work for adding/removing gateways. If
> you've changed a gateway's parameters you'll want to killgw first
> though because if a gateway with the same name already exists it'll
> ignore it (If this is no longer the case, please let me know!). I
> don't think that would drop calls, but it would stop you routing to it
> between the killgw and the rescan.
>
> Sighup won't drop any calls, it's often sent by a cronjob once an hour
> to rotate log files.
>
> -Steve
>
>
> On 21 April 2010 21:40, David Ponzone <david.ponzone at gmail.com> wrote:
> > For gateways, you need to:
> > sofia profile external rescan reloadxml
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 21/04/2010 ? 22:28, RR a ?crit :
> >
> > Hello,
> > using version
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.5pre9 (hacked)
> > and when I change the configuration of a gateway under
> > conf/sip_profiles/external/gateway2.xml configuration and then do a
> > reloadxml, nothing changes within the freeswitch db as seen from 'sofia
> > status'. Is this a bug and has been fixed in newer versions?
> > Also, does doing an 'fs_ctl send_sighup' drop all calls from the system?
> > would doing that reload the gateway configurations?
> > Thanks
> > \RR
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Thu Apr 22 12:17:01 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 23 Apr 2010 02:17:01 +0700
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <247D9598-335D-4DD9-9A53-D71E8D774DEB@avgs.ca>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
	<b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
	<w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
	<247D9598-335D-4DD9-9A53-D71E8D774DEB@avgs.ca>
Message-ID: <t2z8ccbff061004221217ze742eadbyf2d12b52c977ddcb@mail.gmail.com>

2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>:
> I have mod_redis ready, it'll go in soon.

very good news.

Dome C.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-22, at 2:03 PM, Dome Charoenyost wrote:
>
>> Hi Rupa,
>>
>> ? any update about NoSQL module. Now i'm modify mod_limit change sql
>> to tokyo tyrant. it's work fine for me. now i can share limit data to
>> other FS server. Now i'm thinking about all fs db. if use redis
>> (sorted sets) ?i think posible todo. but it's not easy to modify fs
>> code.
>>
>> So i need comment about aventage if change fs backend froom SQL to NoSQL.
>>
>> BG
>>
>> Dome C.
>>
>>
>> 2010/3/4 Rupa Schomaker <rupa at rupa.com>:
>>> On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman
>>> <freeswitch at cartissolutions.com> wrote:
>>>>
>>>>
>>>> Maybe it would be even more useful to provide general functionality that
>>>> can be shared amongst multiple components of freeswitch. ?That would
>>>> make it all the more useful. ?However, seeing that I'm not a big fan of
>>>> the NoSQL data engines, I don't know if this is possible. ?In my
>>>> opinion, most NoSQL is just key=>value pairs, and is basically a
>>>> reinvention of the wheel that RDBMS's had taken care of years ago...
>>>> Not meaning to open up a discussion about the virtues of NoSQL (and
>>>> certainly not a flamewar)...
>>>>
>>>
>>> I created a mod_memcache already and I intend to do a mod_redis at some
>>> point. ?I'm not a huge fan of NoSQL but there are definite uses for it.
>>>
>>> General functionality: ?I could be convinced to come up with a general api
>>> for distributed key/value store with mod_memcache and mod_redis providing
>>> implementation. ?The problem is that the NoSQL stuff isn't very consistent
>>> so even though both support key/value their behavior can significantly
>>> differ. ?eg: updating a value in memcached doesn't bump the expire time but
>>> does in redis. ?Also, redis has support for a much more robust set of value
>>> types and operators.
>>> Anyway, a generic distributed key/value api might look like:
>>> dhash <backend> set key value [expire]
>>> dhash <backend> setnx key value [expire] ?# only set if it doesn't already
>>> exist
>>> dhash <backend> get key [...] # support multiple keys
>>> dhash <backend> <increment|decrement> key [step]
>>> dhash <backend> del key
>>> where backend would be memcache or redis or some other implementation.
>>> Notice I didn't even touch things like hashing to support sharding,
>>> failover, etc.
>>>
>>>>
>>>> Yossi Neiman
>>>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From delorenzodesign at gmail.com  Thu Apr 22 12:41:15 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Thu, 22 Apr 2010 15:41:15 -0400
Subject: [Freeswitch-users] Mod xml cdr
Message-ID: <i2h5c9dcbfb1004221241l8c019517hb4d42d925634d0ec@mail.gmail.com>

I have mod_xml_cdr installed but I can't seem to get it to make the HTTP
post.  I don't see any attempts either in the console or in my Fresswitch
log about it attempting, nevermind failing.

I've tried reloading the module from the console via "reload_xml_cdr" and
then processed a call and still nothing.  Here's my xml_cdr config:

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
  <settings>
<!--     <param name="cred" value="user:pass"/> -->
     <param name="url" value="http://url_path/save_cdr" />
     <param name="retries" value="5"/>
     <param name="delay" value="120"/>

     <param name="log-dir" value="/var/log/cdr"/>
     <param name="err-log-dir" value="/var/log/cdr/errors"/>

     <param name="encode" value="base64"/>
  </settings>
</configuration>

I've also verified that the module is enabled in the modules.conf.xml.

Can anyone point me in the right direction?
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From kenfulmer at icstechnologysolutions.com  Thu Apr 22 12:51:57 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 14:51:57 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
Message-ID: <012c01cae255$44971510$cdc53f30$@com>

Ok, that's not a problem. We'll only need to do this when a customer's
server dies. 

 

How about the question about adding licenses together? If we've purchased
separate licenses, how can we add each one to a server? 

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 1:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org

 

If you have to replace the machine you'll have to send me the proof the
machine is dead and I can reset them... I can only do this 3 times before
you have to repurchase the license.

 

/b

 

On Apr 22, 2010, at 1:24 PM, Ken Fulmer wrote:





1.       Are these licenses additive? In other words, if I've purchased one
for a machine and need a total of five, can I purchase four more and add
them to the mix?

2.       If we have to replace a machine, how can we replace the licenses
without repurchasing them?

 

Thanks,

 

Ken Fulmer

 

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

 

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From ranjtech at gmail.com  Thu Apr 22 12:54:31 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 22 Apr 2010 15:54:31 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
	<9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
Message-ID: <q2udcde51f21004221254yc5f47be8o733d83b59403266c@mail.gmail.com>

Hi David,

The method below to extract the value of or access X-Original-IP didn't work
in Asterisk. I really want to avoid going to the Asterisk mailing list...so
if anyone can help figuring out how I can access custom header value in an
Asterisk Dialplan, I would be really grateful.

Thanks
RR

2010/4/21 Brian West <brian at freeswitch.org>

> In freeswitch the incoming X-Headers are done with sip_h_X-Header as a
> variable.
>
> /b
>
> On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:
>
> It's not the right place for this but well :)
>
> Should be ${SIP_HEADER(X-Original-IP)}
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Thu Apr 22 12:57:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 22 Apr 2010 22:57:53 +0300
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
Message-ID: <w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>

Try adding the following line in your diaplan before the bridge command:

<action application="set" data="ignore_display_updates=true"/>
(try true and false and see which one works better for you).

                           __Yehavi:
2010/4/22 Wellie Chao <wchao at yahoo.com>

> I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now
> I see a strange behavior with caller ID on Polycom phones when handling
> inbound calls.
>
> Here is the scenario:
>
> * Call is from 212-555-2222 (external number not on my softswitch)
>
> * Call is to 212-555-1001 (number on my softswitch, extension 1001)
>
> * extension 1001 is a Polycom phone (it's an IP301, but same problem
>   occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
>   SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.
>
> * on phone 1001, the caller ID will display 212-555-2222 while ringing.
>   The moment I pick up, the display will change to "From: 1001" (referring
>   to the extension of the phone itself).
>
> Has anyone else experienced this problem, and does anyone know how to fix
> it? It does not occur with the snom 320 (and I assume it does not occur
> with any of the snom models based on extrapolation).
>
> While the problem only started when I updated from 1.0.4 to 1.0.6, it's
> entirely possible it's a configuration setting on the Polycom rather than
> a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
> resolve the problem (or even how to go about investigating the root
> cause)?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ranjtech at gmail.com  Thu Apr 22 13:03:26 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 22 Apr 2010 16:03:26 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <q2udcde51f21004221254yc5f47be8o733d83b59403266c@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
	<9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
	<q2udcde51f21004221254yc5f47be8o733d83b59403266c@mail.gmail.com>
Message-ID: <k2mdcde51f21004221303t16af0e5w678cb532b0aab87d@mail.gmail.com>

Sorry ignore that question. I figured it out :)

Thanks

On Thu, Apr 22, 2010 at 3:54 PM, RR <ranjtech at gmail.com> wrote:

> Hi David,
>
> The method below to extract the value of or access X-Original-IP didn't
> work in Asterisk. I really want to avoid going to the Asterisk mailing
> list...so if anyone can help figuring out how I can access custom header
> value in an Asterisk Dialplan, I would be really grateful.
>
> Thanks
> RR
>
> 2010/4/21 Brian West <brian at freeswitch.org>
>
>> In freeswitch the incoming X-Headers are done with sip_h_X-Header as a
>> variable.
>>
>> /b
>>
>> On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:
>>
>> It's not the right place for this but well :)
>>
>> Should be ${SIP_HEADER(X-Original-IP)}
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Thu Apr 22 13:05:49 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 15:05:49 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <012c01cae255$44971510$cdc53f30$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
	<012c01cae255$44971510$cdc53f30$@com>
Message-ID: <ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>

Add the additional license files in and HUP the freeswitch_licence_server thats running.

/b

On Apr 22, 2010, at 2:51 PM, Ken Fulmer wrote:

> Ok, that?s not a problem. We?ll only need to do this when a customer?s server dies.
>  
> How about the question about adding licenses together? If we?ve purchased separate licenses, how can we add each one to a server?
>  
> Thanks,
>  
> Ken

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From david.ponzone at gmail.com  Thu Apr 22 13:11:52 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 22 Apr 2010 22:11:52 +0200
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <k2mdcde51f21004221303t16af0e5w678cb532b0aab87d@mail.gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
	<9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
	<q2udcde51f21004221254yc5f47be8o733d83b59403266c@mail.gmail.com>
	<k2mdcde51f21004221303t16af0e5w678cb532b0aab87d@mail.gmail.com>
Message-ID: <68E4D5C0-A17F-4196-90E0-CBA1AB48170A@gmail.com>

Can you share it, just in case ?

Even if Asterisk is obviously not the topic here, FreeSWITCH  
integration into an Asterisk-based architecture is always an  
interesting subject, as it allows smooth 100% migration to FreeSWITCH :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 22/04/2010 ? 22:03, RR a ?crit :

> Sorry ignore that question. I figured it out :)
>
> Thanks
>
> On Thu, Apr 22, 2010 at 3:54 PM, RR <ranjtech at gmail.com> wrote:
> Hi David,
>
> The method below to extract the value of or access X-Original-IP  
> didn't work in Asterisk. I really want to avoid going to the  
> Asterisk mailing list...so if anyone can help figuring out how I can  
> access custom header value in an Asterisk Dialplan, I would be  
> really grateful.
>
> Thanks
> RR
>
> 2010/4/21 Brian West <brian at freeswitch.org>
> In freeswitch the incoming X-Headers are done with sip_h_X-Header as  
> a variable.
>
> /b
>
> On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:
>
>> It's not the right place for this but well :)
>>
>> Should be ${SIP_HEADER(X-Original-IP)}
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ranjtech at gmail.com  Thu Apr 22 13:41:42 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 22 Apr 2010 16:41:42 -0400
Subject: [Freeswitch-users] passing originating IP when configured as a
	B2BUA
In-Reply-To: <68E4D5C0-A17F-4196-90E0-CBA1AB48170A@gmail.com>
References: <s2tdcde51f21004201624ve96972b6ze32440e80ed4236e@mail.gmail.com>
	<A480DA19-AE50-4099-BAA4-0F2E31574EF9@gmail.com>
	<k2ndcde51f21004210834kddb0bd3dnb7c36f41c5b88e1a@mail.gmail.com>
	<45E53021-534B-44DF-8650-ADB9E6FE5EB2@gmail.com>
	<k2hdcde51f21004211107h466f1f5cmad5524c040459410@mail.gmail.com>
	<30138921-3A27-4C0E-8021-A350B675CC59@gmail.com>
	<9E631AFC-73A4-4722-B882-A1BD5E5891F6@freeswitch.org>
	<q2udcde51f21004221254yc5f47be8o733d83b59403266c@mail.gmail.com>
	<k2mdcde51f21004221303t16af0e5w678cb532b0aab87d@mail.gmail.com>
	<68E4D5C0-A17F-4196-90E0-CBA1AB48170A@gmail.com>
Message-ID: <t2jdcde51f21004221341ya31f9011y58a804b3de728d8b@mail.gmail.com>

Oh it was exactly how you suggested it. What I figured out what why it
wasn't working for me. It's because I had originally created my Asterisk
Dialplan to handle the From field in the format <number>@<IP Address>, so I
was removing the @ and '<>' etc from it, whereas if I use the X-Original-IP
to simply be the $network_addr variable then I don't need to do any
manipulation.

Thanks
|RR

On Thu, Apr 22, 2010 at 4:11 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Can you share it, just in case ?
>
> Even if Asterisk is obviously not the topic here, FreeSWITCH integration
> into an Asterisk-based architecture is always an interesting subject, as it
> allows smooth 100% migration to FreeSWITCH :)
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 22/04/2010 ? 22:03, RR a ?crit :
>
> Sorry ignore that question. I figured it out :)
>
> Thanks
>
> On Thu, Apr 22, 2010 at 3:54 PM, RR <ranjtech at gmail.com> wrote:
>
>> Hi David,
>>
>> The method below to extract the value of or access X-Original-IP didn't
>> work in Asterisk. I really want to avoid going to the Asterisk mailing
>> list...so if anyone can help figuring out how I can access custom header
>> value in an Asterisk Dialplan, I would be really grateful.
>>
>> Thanks
>> RR
>>
>> 2010/4/21 Brian West <brian at freeswitch.org>
>>
>>> In freeswitch the incoming X-Headers are done with sip_h_X-Header as a
>>> variable.
>>>
>>> /b
>>>
>>> On Apr 21, 2010, at 1:15 PM, David Ponzone wrote:
>>>
>>> It's not the right place for this but well :)
>>>
>>> Should be ${SIP_HEADER(X-Original-IP)}
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr 22 13:51:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Apr 2010 13:51:32 -0700
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com>
Message-ID: <s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com>

On Wed, Apr 21, 2010 at 3:10 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> I'm trying to make a sip call using Gizmo. I set up the profile as
> specified here<http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo>and dropped it into /conf/sip_profiles/external/. I can see that the profile
> is registered when I check sofia status from CLI:
>                      Name   Type                                      Data
> State
>
> =================================================================================================
>                  ...
>          external::gizmo gateway            sip:otalo at proxy01.sipphone.com<sip%3Aotalo at proxy01.sipphone.com>
> REGED
>            ...
>
> =================================================================================================
>
> On a call event, I invoke the lua commands:
>
> sessiondata = "sofia/gateway/gizmo/<num>"
> new_session = freeswitch.Session(sessiondata)
>
> What happens if you do something like this at the CLI:
originate sofia/gateway/gizmo/<num> <target ext>

Be sure to turn on sip trace for the external profile to see if anything
interesting is happening.
-MC


>
> From this I'm getting a NO_USER_RESPONSE error:
>
> 2010-04-21 11:31:16.169900 [DEBUG] sofia.c:4153 Channel sofia/internal/
> 1234 at conference.freeswitch.org entering state [terminated][480]
> 2010-04-21 11:31:16.169900 [NOTICE] sofia.c:4789 Hangup sofia/internal/
> 1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
> 2010-04-21 11:31:16.169900 [DEBUG] switch_channel.c:2071 Send signal
> sofia/internal/1234 at conference.freeswitch.org [KILL]
> 2010-04-21 11:31:16.169900 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/1234 at conference.freeswitch.org [BREAK]
> 2010-04-21 11:31:16.169900 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
> CS_HANGUP
> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/1234 at conference.freeswitch.org) State HANGUP
> 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:408 sofia/internal/
> 1234 at conference.freeswitch.org Overriding SIP cause 408 with 480 from the
> other leg
> 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:414 Channel sofia/internal/
> 1234 at conference.freeswitch.org hanging up, cause: NO_USER_RESPONSE
> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1234 at conference.freeswitch.org Standard HANGUP, cause:
> NO_USER_RESPONSE
> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/1234 at conference.freeswitch.org) State HANGUP going to
> sleep
> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/1234 at conference.freeswitch.org) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/1234 at conference.freeswitch.org [BREAK]
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
> CS_REPORTING
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/1234 at conference.freeswitch.org) State REPORTING
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1234 at conference.freeswitch.org Standard REPORTING, cause:
> NO_USER_RESPONSE
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/1234 at conference.freeswitch.org) State REPORTING going to
> sleep
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/1234 at conference.freeswitch.org) State Change CS_REPORTING
> -> CS_DESTROY
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/1234 at conference.freeswitch.org [BREAK]
> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1161 Session 4
> (sofia/internal/1234 at conference.freeswitch.org) Locked, Waiting on
> external entities
> 2010-04-21 11:31:16.177790 [DEBUG] switch_ivr_originate.c:3228 Originate
> Resulted in Error Cause: 18 [NO_USER_RESPONSE]
> *2010-04-21 11:31:16.177790 [ERR] mod_conference.c:4563 Cannot create
> outgoing channel, cause: NO_USER_RESPONSE*
> 2010-04-21 11:31:16.177790 [NOTICE] mod_conference.c:4566 Hangup
> sofia/internal/1001 at server.IP [CS_EXECUTE] [NO_USER_RESPONSE]
> 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1179 Session 4
> (sofia/internal/1234 at conference.freeswitch.org) Ended
> 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:428
> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
> CS_DESTROY
> 2010-04-21 11:31:16.177790 [DEBUG] switch_channel.c:2071 Send signal
> sofia/internal/1001 at server.IP [KILL]
> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:439
> (sofia/internal/1234 at conference.freeswitch.org) State DESTROY
> 2010-04-21 11:31:16.177790 [DEBUG] mod_sofia.c:341 sofia/internal/
> 1234 at conference.freeswitch.org SOFIA DESTROY
> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/1001 at server.IP [BREAK]
>
> Strangely, I get the same error even if I put in some arbitrary gateway name in the lua code, so it seems like possibly the profile isn't being properly read. I also thought that perhaps I should be using "sofia/gateway/external::gizmo/<num>" instead of "../gizmo/..", but that didn't seem to work. I haven't made any other changes to conf files, though perhaps I'm missing something?
>
> Thanks in advance,
> Neil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kenfulmer at icstechnologysolutions.com  Thu Apr 22 14:06:35 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 22 Apr 2010 16:06:35 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>
	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>
Message-ID: <017601cae25f$b2067aa0$16136fe0$@com>

I apologize, but I'm not sure what you mean.

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 3:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org

 

Add the additional license files in and HUP the freeswitch_licence_server
thats running.

 

/b

 

On Apr 22, 2010, at 2:51 PM, Ken Fulmer wrote:





Ok, that's not a problem. We'll only need to do this when a customer's
server dies.

 

How about the question about adding licenses together? If we've purchased
separate licenses, how can we add each one to a server?

 

Thanks,

 

Ken

 

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From brian at freeswitch.org  Thu Apr 22 14:13:30 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 16:13:30 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <017601cae25f$b2067aa0$16136fe0$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>
	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>
	<017601cae25f$b2067aa0$16136fe0$@com>
Message-ID: <3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>

If you buy more licenses... activate them on the machine,, copy the .conf file to /etc/freeswitch then restart or HUP the freeswitch_licence_server that is running and it will add the licenses together into one.

/b

On Apr 22, 2010, at 4:06 PM, Ken Fulmer wrote:

> I apologize, but I?m not sure what you mean.
>  
> Thanks,
>  
> Ken
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Thursday, April 22, 2010 3:06 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org
>  
> Add the additional license files in and HUP the freeswitch_licence_server thats running.
>  
> /b

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From william.suffill at gmail.com  Thu Apr 22 14:16:09 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 22 Apr 2010 17:16:09 -0400
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <017601cae25f$b2067aa0$16136fe0$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org> 
	<012c01cae255$44971510$cdc53f30$@com>
	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org> 
	<017601cae25f$b2067aa0$16136fe0$@com>
Message-ID: <t2n6b65470d1004221416tef62c5bam3c64f05e683a9574@mail.gmail.com>

You would need to have the freeswitch_license_server re read the license
files so it knows you have added more. In linux you can kill -HUP a process
to have it re read config changes and the like.
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From neilp at cs.stanford.edu  Thu Apr 22 14:46:23 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 22 Apr 2010 14:46:23 -0700
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com> 
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com>
Message-ID: <v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com>

Hey Michael,

What happens if you do something like this at the CLI:
> originate sofia/gateway/gizmo/<num> <target ext>
>

This works! Although I got "command not found" when I did this from
fs_cli... why's that?


> Be sure to turn on sip trace for the external profile to see if anything
> interesting is happening.
>

Weird. When I execute:

sofia profile gizmo siptrace on

I get:

Invalid Profile [gizmo]

This is the contents of /freeswitch/conf/sip_profiles/external/gizmo.xml:
<include>
<gateway name="gizmo">
<param name="username" value=UNAME/>
<param name="realm" value="proxy01.sipphone.com"/>
<param name="from-user" value=UNAME/>
<param name="from-domain" value="proxy01.sipphone.com"/>
<param name="password" value=PWD/>
<param name="caller-id-in-from" value="false"/>
<param name="extension" value="cluecon"/>
<param name="proxy" value="proxy01.sipphone.com"/>
<param name="register-proxy" value="proxy01.sipphone.com"/>
<param name="expire-seconds" value="3600"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry_seconds" value="30"/>
<param name="caller-id-in-from" value="false"/>
</gateway>
</include>

-Neil

-MC
>
>
>>
>> From this I'm getting a NO_USER_RESPONSE error:
>>
>> 2010-04-21 11:31:16.169900 [DEBUG] sofia.c:4153 Channel sofia/internal/
>> 1234 at conference.freeswitch.org entering state [terminated][480]
>> 2010-04-21 11:31:16.169900 [NOTICE] sofia.c:4789 Hangup sofia/internal/
>> 1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
>> 2010-04-21 11:31:16.169900 [DEBUG] switch_channel.c:2071 Send signal
>> sofia/internal/1234 at conference.freeswitch.org [KILL]
>> 2010-04-21 11:31:16.169900 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/internal/1234 at conference.freeswitch.org [BREAK]
>> 2010-04-21 11:31:16.169900 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
>> CS_HANGUP
>> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
>> (sofia/internal/1234 at conference.freeswitch.org) State HANGUP
>> 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:408 sofia/internal/
>> 1234 at conference.freeswitch.org Overriding SIP cause 408 with 480 from the
>> other leg
>> 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:414 Channel sofia/internal/
>> 1234 at conference.freeswitch.org hanging up, cause: NO_USER_RESPONSE
>> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/1234 at conference.freeswitch.org Standard HANGUP, cause:
>> NO_USER_RESPONSE
>> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499
>> (sofia/internal/1234 at conference.freeswitch.org) State HANGUP going to
>> sleep
>> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:333
>> (sofia/internal/1234 at conference.freeswitch.org) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2010-04-21 11:31:16.170748 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/internal/1234 at conference.freeswitch.org [BREAK]
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
>> CS_REPORTING
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
>> (sofia/internal/1234 at conference.freeswitch.org) State REPORTING
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/1234 at conference.freeswitch.org Standard REPORTING, cause:
>> NO_USER_RESPONSE
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590
>> (sofia/internal/1234 at conference.freeswitch.org) State REPORTING going to
>> sleep
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/1234 at conference.freeswitch.org) State Change CS_REPORTING
>> -> CS_DESTROY
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/internal/1234 at conference.freeswitch.org [BREAK]
>> 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1161 Session 4
>> (sofia/internal/1234 at conference.freeswitch.org) Locked, Waiting on
>> external entities
>> 2010-04-21 11:31:16.177790 [DEBUG] switch_ivr_originate.c:3228 Originate
>> Resulted in Error Cause: 18 [NO_USER_RESPONSE]
>> *2010-04-21 11:31:16.177790 [ERR] mod_conference.c:4563 Cannot create
>> outgoing channel, cause: NO_USER_RESPONSE*
>> 2010-04-21 11:31:16.177790 [NOTICE] mod_conference.c:4566 Hangup
>> sofia/internal/1001 at server.IP [CS_EXECUTE] [NO_USER_RESPONSE]
>> 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1179 Session 4
>> (sofia/internal/1234 at conference.freeswitch.org) Ended
>> 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1181 Close
>> Channel sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
>> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:428
>> (sofia/internal/1234 at conference.freeswitch.org) Running State Change
>> CS_DESTROY
>> 2010-04-21 11:31:16.177790 [DEBUG] switch_channel.c:2071 Send signal
>> sofia/internal/1001 at server.IP [KILL]
>> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:439
>> (sofia/internal/1234 at conference.freeswitch.org) State DESTROY
>> 2010-04-21 11:31:16.177790 [DEBUG] mod_sofia.c:341 sofia/internal/
>> 1234 at conference.freeswitch.org SOFIA DESTROY
>> 2010-04-21 11:31:16.177790 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/internal/1001 at server.IP [BREAK]
>>
>> Strangely, I get the same error even if I put in some arbitrary gateway name in the lua code, so it seems like possibly the profile isn't being properly read. I also thought that perhaps I should be using "sofia/gateway/external::gizmo/<num>" instead of "../gizmo/..", but that didn't seem to work. I haven't made any other changes to conf files, though perhaps I'm missing something?
>>
>> Thanks in advance,
>> Neil
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu Apr 22 14:51:59 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 22 Apr 2010 18:51:59 -0300
Subject: [Freeswitch-users] Mod xml cdr
In-Reply-To: <i2h5c9dcbfb1004221241l8c019517hb4d42d925634d0ec@mail.gmail.com>
References: <i2h5c9dcbfb1004221241l8c019517hb4d42d925634d0ec@mail.gmail.com>
Message-ID: <r2xebbc2d551004221451hde1c55b2ha53638bdde6fe756@mail.gmail.com>

You mean reload mod_xml_cdr, right?

JM


On Thu, Apr 22, 2010 at 4:41 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I have mod_xml_cdr installed but I can't seem to get it to make the HTTP
> post.  I don't see any attempts either in the console or in my Fresswitch
> log about it attempting, nevermind failing.
>
> I've tried reloading the module from the console via "reload_xml_cdr" and
> then processed a call and still nothing.  Here's my xml_cdr config:
>
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>   <settings>
> <!--     <param name="cred" value="user:pass"/> -->
>      <param name="url" value="http://url_path/save_cdr" />
>      <param name="retries" value="5"/>
>      <param name="delay" value="120"/>
>
>      <param name="log-dir" value="/var/log/cdr"/>
>      <param name="err-log-dir" value="/var/log/cdr/errors"/>
>
>      <param name="encode" value="base64"/>
>   </settings>
> </configuration>
>
> I've also verified that the module is enabled in the modules.conf.xml.
>
> Can anyone point me in the right direction?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Thu Apr 22 14:53:12 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 22 Apr 2010 16:53:12 -0500
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
	<b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
	<w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
Message-ID: <n2pb39273531004221453q47f639f5g48cf37ab7cff1ee3@mail.gmail.com>

Dome,

I personally don't have any plans to add NoSQL to nibblebill.  If you have a
patch or a bounty maybe that could change,  But for now, SQL works for me.

On Thu, Apr 22, 2010 at 1:03 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Hi Rupa,
>
>   any update about NoSQL module. Now i'm modify mod_limit change sql
> to tokyo tyrant. it's work fine for me. now i can share limit data to
> other FS server. Now i'm thinking about all fs db. if use redis
> (sorted sets)  i think posible todo. but it's not easy to modify fs
> code.
>
> So i need comment about aventage if change fs backend froom SQL to NoSQL.
>
> BG
>
> Dome C.
>
>
> 2010/3/4 Rupa Schomaker <rupa at rupa.com>:
> > On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman
> > <freeswitch at cartissolutions.com> wrote:
> >>
> >>
> >> Maybe it would be even more useful to provide general functionality that
> >> can be shared amongst multiple components of freeswitch.  That would
> >> make it all the more useful.  However, seeing that I'm not a big fan of
> >> the NoSQL data engines, I don't know if this is possible.  In my
> >> opinion, most NoSQL is just key=>value pairs, and is basically a
> >> reinvention of the wheel that RDBMS's had taken care of years ago...
> >> Not meaning to open up a discussion about the virtues of NoSQL (and
> >> certainly not a flamewar)...
> >>
> >
> > I created a mod_memcache already and I intend to do a mod_redis at some
> > point.  I'm not a huge fan of NoSQL but there are definite uses for it.
> >
> > General functionality:  I could be convinced to come up with a general
> api
> > for distributed key/value store with mod_memcache and mod_redis providing
> > implementation.  The problem is that the NoSQL stuff isn't very
> consistent
> > so even though both support key/value their behavior can significantly
> > differ.  eg: updating a value in memcached doesn't bump the expire time
> but
> > does in redis.  Also, redis has support for a much more robust set of
> value
> > types and operators.
> > Anyway, a generic distributed key/value api might look like:
> > dhash <backend> set key value [expire]
> > dhash <backend> setnx key value [expire]  # only set if it doesn't
> already
> > exist
> > dhash <backend> get key [...] # support multiple keys
> > dhash <backend> <increment|decrement> key [step]
> > dhash <backend> del key
> > where backend would be memcache or redis or some other implementation.
> > Notice I didn't even touch things like hashing to support sharding,
> > failover, etc.
> >
> >>
> >> Yossi Neiman
> >> Cartis Solutions, Inc. - http://www.cartissolutions.com
> >>
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From william.suffill at gmail.com  Thu Apr 22 14:58:41 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 22 Apr 2010 17:58:41 -0400
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com> 
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com> 
	<v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com>
Message-ID: <s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com>

gizmo is your gateway.


siptrace is by profile.


sofia status in the cli will list your profiles/gateways (default cfgs use
internal/external)

so in that case sofia profile external siptrace on
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From neilp at cs.stanford.edu  Thu Apr 22 15:16:20 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 22 Apr 2010 15:16:20 -0700
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com> 
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com> 
	<v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com> 
	<s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com>
Message-ID: <n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com>

Thanks for clarifying.

Here's the output (with siptrace on) when trying to dial out over the
gateway:

2010-04-22 15:13:10.104866 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1001 at server.com [a6aac86b-4a40-41f3-8968-f9b37501483a]
2010-04-22 15:13:10.112850 [INFO] mod_dialplan_xml.c:418 Processing
neilp->5001 in context default
2010-04-22 15:13:10.120874 [INFO] switch_core_session.c:1750 Sending early
media
2010-04-22 15:13:10.122856 [NOTICE] mod_sofia.c:1907 Pre-Answer
sofia/internal/1001 at server.com!
2010-04-22 15:13:10.123810 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1234 at conference.freeswitch.org[af13cf37-eb6a-4e9f-b1cc-366b6a24aec5]
2010-04-22 15:13:10.386879 [INFO] switch_rtp.c:2049 Auto Changing port from
IP1 to IP2
2010-04-22 15:13:10.697870 [NOTICE] sofia.c:4789 Hangup sofia/internal/
1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2010-04-22 15:13:10.706880 [ERR] mod_conference.c:4563 Cannot create
outgoing channel, cause: NO_USER_RESPONSE
2010-04-22 15:13:10.706880 [NOTICE] mod_conference.c:4566 Hangup
sofia/internal/1001 at server.com [CS_EXECUTE] [NO_USER_RESPONSE]
2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1179 Session 2
(sofia/internal/1234 at conference.freeswitch.org) Ended
2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1179 Session 1
(sofia/internal/1001 at server.com) Ended
2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/internal/1001 at server.com [CS_DESTROY]


 In case it is relevant, the call is triggered in a lua script which is
executed by dialing a extension 5001. The dialplan in
freeswitch/conf/dialplan/default/5001_gizmo.xml:
 <extension name="gizmo">
      <condition field="destination_number" expression="^5001$">
      <action application="lua" data="makecall.lua"/>
    </condition>
  </extension>

makecall.lua:
function make_call()
  sessiondata = "sofia/gateway/gizmo/<num>"
  new_session = freeswitch.Session(sessiondata)
end
make_call()



On Thu, Apr 22, 2010 at 2:58 PM, William Suffill
<william.suffill at gmail.com>wrote:

> gizmo is your gateway.
>
>
> siptrace is by profile.
>
>
> sofia status in the cli will list your profiles/gateways (default cfgs use
> internal/external)
>
> so in that case sofia profile external siptrace on
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr 22 15:22:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Apr 2010 15:22:09 -0700
Subject: [Freeswitch-users] mod_say or strftime problems?
In-Reply-To: <34adeff2737457cf22300f3b142145e5.squirrel@webmail.tagnet.ru>
References: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>
	<201004221512.18576.yivzhenko@mksat.net>
	<34adeff2737457cf22300f3b142145e5.squirrel@webmail.tagnet.ru>
Message-ID: <r2m87f2f3b91004221522ie3e3f63an1cdc581558cb7f0e@mail.gmail.com>

Would you please open a JIRA on this one?
-MC

On Thu, Apr 22, 2010 at 6:55 AM, <boris at tagnet.ru> wrote:

> IMHO this is bug, as of struct tm man page:
>  int tm_hour;    /* hours (0 - 23) */
> so tm_hour shouldn't be incremented. But looking at code I see no reason
> why the behavior is changed with different timezone.
>
> > I have same problem with mod_say_ru if timezone is set to
> >
> > <action application="set" data="timezone=Europe/Kiev"/>
> >
> > And i don't understand why  "tm.tm_hour + 1"
> > In source code
> >         if (say_time) {
> >                 switch_snprintf(buf, sizeof(buf), "%d:%d:%d", tm.tm_hour
> +
> > 1,
> > tm.tm_min, tm.tm_sec);
> >                 say_args->type = SST_TIME_MEASUREMENT;
> >                 ru_say_time(session, buf, say_args, args);
> >
> > As temporary resolution i just set different timezone :)
> >
> >
> > On Wednesday 21 April 2010 17:30:18 boris at tagnet.ru wrote:
> >> Hello!
> >>
> >> I have an extension for the current date/time:
> >>     <extension name="SayTime">
> >>         <condition field="destination_number" expression="^9000$">
> >>             <action application="answer"/>
> >>             <action application="say" data="ru current_date_time
> >> pronounced $strftime(%s)"/>
> >>             <action application="sleep" data="1000"/>
> >>             <action application="hangup"/>
> >>         </condition>
> >>     </extension>
> >>
> >> The extension works fine, but time is +1 hour of current time. For
> >> example
> >> current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6
> >> (from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer
> >> time in effect now, so tz=YEKST). So, my question is - something wrong
> >> with my extension configuration or this is bug in mod_say_ru or may be
> >> strfmt?
> >>
> >> With respect,
> >> Boris
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Thu Apr 22 15:27:12 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 22 Apr 2010 15:27:12 -0700
Subject: [Freeswitch-users] Transferring to Non-Existent Extension
Message-ID: <6FD41EA060CF48528942723112A339B0@greyhawk.tonecommander.com>


Is there a "built-in" way for a transferor to recover a transferee who was
transferred to a non-existent extension?  Any examples of this?  Right now,
FS will send a BYE to the transferee with cause "NO_ROUTE_DESTINATION".

Best Regards,
Jerry



From delorenzodesign at gmail.com  Thu Apr 22 15:30:10 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Thu, 22 Apr 2010 18:30:10 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 281
In-Reply-To: <mailman.2257.1271973224.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.2257.1271973224.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <u2p5c9dcbfb1004221530lc3673630ma2143811560870ea@mail.gmail.com>

Yes I mean reload mod_xml_cdr. I get a success response when I do it.

From: Jo?o Mesquita jmesquita at freeswitch.orgTo:
freeswitch-users at lists.freeswitch.orgDate: Thu, 22 Apr 2010
18:51:59Subject: Re: [Freeswitch-users] Mod xml cdrYou mean reload
mod_xml_cdr, right? JMOn Thu, Apr 22, 2010 at 4:41 PM, Michael De
Lorenzo ?<delorenzodesign at gmail.com>? wrote:I have mod_xml_cdr
installed but I can't seem to get it to make the HTTP post. ?I don't
see any attempts either in the console or in my Fresswitch log about
it attempting, nevermind failing. I've tried reloading the module from
the console via "reload_xml_cdr" and then processed a call and still
nothing. ?Here's my xml_cdr config:<configuration name="xml_cdr.conf"
description="XML CDR CURL logger">??<settings><!-- ? ? <param
name="cred" value="user:pass"/> -->?? ? <param name="url"
value="http://url_path/save_cdr" />?? ? <param name="retries"
value="5"/>?? ? <param name="delay" value="120"/>?? ? <param
name="log-dir" value="/var/log/cdr"/>?? ? <param name="err-log-dir"
value="/var/log/cdr/errors"/>?? ? <param name="encode"
value="base64"/>??</settings></configuration>I've also verified that
the module is enabled in the modules.conf.xml.Can anyone point me in
the right direction?

On 4/22/10, freeswitch-users-request at lists.freeswitch.org
<freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>


From brian at freeswitch.org  Thu Apr 22 15:34:48 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 17:34:48 -0500
Subject: [Freeswitch-users] Transferring to Non-Existent Extension
In-Reply-To: <6FD41EA060CF48528942723112A339B0@greyhawk.tonecommander.com>
References: <6FD41EA060CF48528942723112A339B0@greyhawk.tonecommander.com>
Message-ID: <8E865498-5659-475E-84F6-3A80EDAF4E35@freeswitch.org>

the handy variable transfer_fallback_extension might help.

/b

On Apr 22, 2010, at 5:27 PM, Jerry Richards wrote:

> 
> Is there a "built-in" way for a transferor to recover a transferee who was
> transferred to a non-existent extension?  Any examples of this?  Right now,
> FS will send a BYE to the transferee with cause "NO_ROUTE_DESTINATION".
> 
> Best Regards,
> Jerry



From vipkilla at gmail.com  Thu Apr 22 14:26:41 2010
From: vipkilla at gmail.com (vip killa)
Date: Thu, 22 Apr 2010 17:26:41 -0400
Subject: [Freeswitch-users] recommended sip phone or adapter
Message-ID: <m2u957f61371004221426u95cf9c4ct724645ab91bd3997@mail.gmail.com>

I have a freeswitch server running at one location. I want to setup a sip
phone at another location (outside the local freeswitch network). can
someone recommend a sip phone or adapter that is not too expensive and will
easily connect/work with the freeswitch server at location one?
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From msc at freeswitch.org  Thu Apr 22 16:49:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Apr 2010 16:49:03 -0700
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <n2pb39273531004221453q47f639f5g48cf37ab7cff1ee3@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
	<b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>
	<w2x8ccbff061004221103x58f52e83q30590dacfa144103@mail.gmail.com>
	<n2pb39273531004221453q47f639f5g48cf37ab7cff1ee3@mail.gmail.com>
Message-ID: <q2t87f2f3b91004221649pfd7e77efr8e506c193d872c94@mail.gmail.com>

On Thu, Apr 22, 2010 at 2:53 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Dome,
>
> I personally don't have any plans to add NoSQL to nibblebill.  If you have
> a patch or a bounty maybe that could change,  But for now, SQL works for me.
>
> Is there any valid reason to go nosql-ish for nibblebill?
-MC
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From david.ponzone at gmail.com  Thu Apr 22 16:51:10 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 23 Apr 2010 01:51:10 +0200
Subject: [Freeswitch-users] recommended sip phone or adapter
In-Reply-To: <m2u957f61371004221426u95cf9c4ct724645ab91bd3997@mail.gmail.com>
References: <m2u957f61371004221426u95cf9c4ct724645ab91bd3997@mail.gmail.com>
Message-ID: <593AE1E7-20B9-4E03-B665-365EE99A9446@gmail.com>

Any of them :)

If you want to avoid issues, stay in the known area: Snom, Aastra,  
Polycom, Cisco (Linksys brand)
The Siemens Gigaset DECT/SIP family works also ok for me.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 22/04/2010 ? 23:26, vip killa a ?crit :

> I have a freeswitch server running at one location. I want to setup  
> a sip phone at another location (outside the local freeswitch  
> network). can someone recommend a sip phone or adapter that is not  
> too expensive and will easily connect/work with the freeswitch  
> server at location one?  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr 22 17:20:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Apr 2010 17:20:23 -0700
Subject: [Freeswitch-users] Transferring to Non-Existent Extension
In-Reply-To: <8E865498-5659-475E-84F6-3A80EDAF4E35@freeswitch.org>
References: <6FD41EA060CF48528942723112A339B0@greyhawk.tonecommander.com>
	<8E865498-5659-475E-84F6-3A80EDAF4E35@freeswitch.org>
Message-ID: <o2x87f2f3b91004221720k7799f72bu30e8626e8da3b928@mail.gmail.com>

On Thu, Apr 22, 2010 at 3:34 PM, Brian West <brian at freeswitch.org> wrote:

> the handy variable transfer_fallback_extension might help.
>
> FYI, this isn't in the wiki. If someone would give me a brief desc of how
it works I will add it to the wiki.
Thanks,
MC
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From woodydickson at gmail.com  Thu Apr 22 18:08:00 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 23 Apr 2010 09:08:00 +0800
Subject: [Freeswitch-users] single participant timeout for conference
Message-ID: <m2ia44688a1004221808s30e7e802t8a8c018518d8b588@mail.gmail.com>

Hi,

Is there anyway to configure conference in such a way that if there is
only one participant in the conference, that single conference
participant will be hung up after XX seconds?

Woody


From brian at freeswitch.org  Thu Apr 22 18:27:33 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Apr 2010 20:27:33 -0500
Subject: [Freeswitch-users] single participant timeout for conference
In-Reply-To: <m2ia44688a1004221808s30e7e802t8a8c018518d8b588@mail.gmail.com>
References: <m2ia44688a1004221808s30e7e802t8a8c018518d8b588@mail.gmail.com>
Message-ID: <64A1D4EC-8FEF-4DEB-9133-0EF7D45C37E8@freeswitch.org>

The mintwo flag will kick them if the conference drops below 2

/b

On Apr 22, 2010, at 8:08 PM, Woody Dickson wrote:

> Hi,
> 
> Is there anyway to configure conference in such a way that if there is
> only one participant in the conference, that single conference
> participant will be hung up after XX seconds?
> 
> Woody



From wchao at yahoo.com  Thu Apr 22 19:31:20 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Thu, 22 Apr 2010 22:31:20 -0400 (EDT)
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>

As I mentioned in my last email, the ignore_display_updates variable did 
the trick, but I am just curious: what is the purpose of the default 
behavior where the display changes to the callee name and number? I am 
guessing there must be some use case (in fact a fairly prevalent use case) 
where showing the callee is more desirable than showing the caller, but 
I'm not aware of what that might be and am curious to find out what the 
use case(s) are. Changing the display to callee also seems to affect call 
logs on the Polycom, which makes the received call log kind of useless.


Date: Thu, 22 Apr 2010 22:57:53 +0300
From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom

Try adding the following line in your diaplan before the bridge command:
?
<action application="set" data="ignore_display_updates=true"/>
(try true and false and see which one works better for you).
?
?????????????????????????? __Yehavi:
2010/4/22 Wellie Chao <wchao at yahoo.com>
      I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now
      I see a strange behavior with caller ID on Polycom phones when handling
      inbound calls.

      Here is the scenario:

      * Call is from 212-555-2222 (external number not on my softswitch)

      * Call is to 212-555-1001 (number on my softswitch, extension 1001)

      * extension 1001 is a Polycom phone (it's an IP301, but same problem
      ? occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
      ? SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.

      * on phone 1001, the caller ID will display 212-555-2222 while ringing.
      ? The moment I pick up, the display will change to "From: 1001" (referring
      ? to the extension of the phone itself).

      Has anyone else experienced this problem, and does anyone know how to fix
      it? It does not occur with the snom 320 (and I assume it does not occur
      with any of the snom models based on extrapolation).

      While the problem only started when I updated from 1.0.4 to 1.0.6, it's
      entirely possible it's a configuration setting on the Polycom rather than
      a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
      resolve the problem (or even how to go about investigating the root
      cause)?

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.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  Thu Apr 22 20:55:14 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 23 Apr 2010 05:55:14 +0200
Subject: [Freeswitch-users] Parsing XML files from Spidermonkey
In-Reply-To: <14f6fde22ecd0ad9b6fdd7a7fe56e3d0.squirrel@correo.nodo50.org>
References: <48fade7f8a99c447d71334d2d3e589bd.squirrel@correo.nodo50.org>
	<14f6fde22ecd0ad9b6fdd7a7fe56e3d0.squirrel@correo.nodo50.org>
Message-ID: <j2td021d3be1004222055xa8aa8d75odc08df002465e6b7@mail.gmail.com>

Not possible.

http://markmail.org/message/ctietfw2mucucdyq#query:spidermonkey%20domparser+page:1+mid:jjknnksvfpi4keto+state:results

http://groups.google.com/group/mozilla.dev.tech.js-engine/browse_thread/thread/86b27b854492c22a/237e4ba109d3f69c
<http://groups.google.com/group/mozilla.dev.tech.js-engine/browse_thread/thread/86b27b854492c22a/237e4ba109d3f69c>
On Thu, Apr 22, 2010 at 10:30 AM, Alberto Escudero <aep.lists at it46.se>wrote:

> I am resending this mail, hoping that someone has managed to read a XML
> file from Javascript.
>
>
> --
> Stopping junk mailers is good for the environment
>
> > Hi,
> >
> > After one year using FS i am starting to like XML so i am trying to get
> > a Javascript script to read local XML files.
> >
> > I am using the XML method and getting Syntax errors from spidermonkey
> >
> > While something like this works:
> > xmldata = new XML("<tag>foo</tag>");
> >
> > I have not been able to read and parse XML local files, using File or
> > FileIO
> > methods
> >
> > A simple example like this returns Syntax error.
> > var foo = apiExecute ("show", "channels as xml");
> > xmldata = new XML(foo);
> >
> > Has anyone managed to use the XML method from spidermonkey to read a XML
> > stored file?
> >
> > There are some E4X bugs around and i wonder if those are the cause of the
> > "Syntax Error" feedback even reading a very basic XML file
> > --
> > Stopping junk mailers is good for the environment
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jonas.gauffin at gmail.com  Thu Apr 22 20:58:11 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 23 Apr 2010 05:58:11 +0200
Subject: [Freeswitch-users] ODBC DB Support in Windows
In-Reply-To: <m2h367751821004211025s112e4fc4wc882ac4cea182b00@mail.gmail.com>
References: <m2h367751821004211025s112e4fc4wc882ac4cea182b00@mail.gmail.com>
Message-ID: <i2sd021d3be1004222058k319c8d80i34c2e6600f6d3062@mail.gmail.com>

I'm using FS with PostgreSQL on windows. Got no problems at all.

Note that you also need to setup a DSN.

Control Panel -> Administrative tools -> Data sources (ODBC)

On Wed, Apr 21, 2010 at 7:25 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I am trying to set up ODBC support on Win2003. I have created and tested a
> system DSN and added  this to sofia.conf.xml in the <global_settings>
> section:
>
> *<param name="odbc-dsn" value="FS:xx:password"/>*
>
> Is there more to do?
>
> Freeswitch starts with no problem - seemingly ignoring this request to use
> ODBC.
>
> What am I missing??
>
>
> Phil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gkuri at ieee.org  Thu Apr 22 22:30:02 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 22 Apr 2010 22:30:02 -0700
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
Message-ID: <p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>

Send proof the machine is dead? Would you like a picture of smoke pouring
out of a box in flames?

What if you want to re-allocate the machine to do something else or just
replace it with a more powerful machine? How are the licenses tied to the
machine, MAC address?

Any reason the "option" for a floating license model isn't available? It
seems given the option between the two, I'd rather have the floating license
model so I wouldn't need to prove my machine is dead before asking to have
the licenses re-issued, unless of course the license server is dead. We swap
production machines in/out all the time, particularly because we perform
maintenance on one, so we have a spare, bring it up, take the other one
down, perform maintenance, and bring it back up.

With this model, floating licenses would be our only option, I really
wouldn't want to be purchasing a bunch of extra licenses for spare machines.

I can't imagine you guys really like making us feel like criminals by tieing
the license to an actual box, what's wrong with the good 'ole "on your honor
policy"? I realize the need to pay for valid g729 licenses, but prove the
machine is dead? Is this requirement coming from Sipro?

Cheers,
Gabe

On Thu, Apr 22, 2010 at 11:30 AM, Brian West <brian at freeswitch.org> wrote:

> If you have to replace the machine you'll have to send me the proof the
> machine is dead and I can reset them... I can only do this 3 times before
> you have to repurchase the license.
>
> /b
>
> On Apr 22, 2010, at 1:24 PM, Ken Fulmer wrote:
>
> 1.       Are these licenses additive? In other words, if I?ve purchased
> one for a machine and need a total of five, can I purchase four more and add
> them to the mix?
> 2.       If we have to replace a machine, how can we replace the licenses
> without repurchasing them?
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Fri Apr 23 02:16:24 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 23 Apr 2010 17:16:24 +0800
Subject: [Freeswitch-users] Grandstream gateways
In-Reply-To: <CA993E53CEE7784F8B36147A4034E96701D661A29D@EXVMBX018-2.exch018.msoutlookonline.net>
References: <c29b930b0911251300i7098bf11y4b2e5aaedeb6f53e@mail.gmail.com>
	<771079.9902.qm@web33508.mail.mud.yahoo.com>
	<CA993E53CEE7784F8B36147A4034E96701D661A29D@EXVMBX018-2.exch018.msoutlookonline.net>
Message-ID: <p2r7d0bfd8c1004230216y1a31a842o45001138255604a2@mail.gmail.com>

@kendalll:  CallerID number is fine for PSTN application. what  model do you
have? any experience w/ their 24 port FXS?

i'm really looking for 24 ports FXS for PSTN application. aside from
audiocodes, any brand can you recommend? the FXS must generate 12/16khz
pulse metering signal. tks.

-nandy

On Thu, Dec 10, 2009 at 4:12 AM, Kendall Stauffer <ken at ksac.com> wrote:

>  Yes. I have one if anybody wants it, would let it go cheap.
>
>
>
>   Works fine, but caller id is only the number, not the name part. Other
> than that works fine with astersik
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Toro
> *Sent:* Wednesday, December 09, 2009 3:05 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Grandstream gateways
>
>
>
> I had hear about Welltech (http://www.welltech.com/default.aspx) gateways
> but I don't have any experience with them.
>
>
>
> Someone know ?, any experience...
>
>
>
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
> --- On *Wed, 11/25/09, Milena <testeador01 at gmail.com>* wrote:
>
>
> From: Milena <testeador01 at gmail.com>
> Subject: Re: [Freeswitch-users] Grandstream gateways
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, November 25, 2009, 4:00 PM
>
> Hello,
>
>
>
> Samuel: We also have some GXW4104 gateways, in small production/testing
> environments; our caller id works fine and none of them has failed in over a
> year of being used. The thing that i dislike about the GXW series is that it
> has no support for early media.
>
>
>
> Everyone: What FXO devices do you currently use / recommend?
>
>
>
> 2009/11/25 Chris Chen <chris.chen2004 at gmail.com<http://mc/compose?to=chris.chen2004 at gmail.com>
> >
>
> You haven't really put it into production for more than one year. The issue
> with GXW4108 is that after some time, say a couple of months, either all FXO
> ports not working, or worse, some FXO ports not working, but after power
> recycling, they will come back to work for some time until on strike again
> at some time you have no control.
>
> This had been reported for a couple of years without improvement. Go google
> search you will find out, this has happened to many GXW4108 users.
>
>
>  On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti <samuelmukoti at gmail.com<http://mc/compose?to=samuelmukoti at gmail.com>>
> wrote:
>
> Thank you for those tips,
>
> I do have some small setups using gxw4108 they work or, except CID
> doesn't seem to work.  I will try the channel bank route - just don't
> know too much about the setup options or how you'd purchase the
> correct config, eg. For 150 FXS channel bank, can I get a single PCI
> card for that?
>
> I may end up using the grandstream fxs gateways then use the T1
> channel bank from sangoma,
>
> Thank you all..
>
> Lastly, I know asterisk now has an offical skype_ module, Is there
> anything similar I could use?
>
>
> On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory at voipsupply.com<http://mc/compose?to=cory at voipsupply.com>>
> wrote:
>
> > Samuel - you could go with FXS gateways or channel banks.  If you go
> > the gateway route Grandstream or Audiocodes would work fine.
> > Audiocodes are a bit more telco grade.  If you have 25 POTS incoming
> > you could use a 24FXO channel bank cross connected with Rhino T1
> > cards, or individual FXO gateways but you may have a hard time
> > finding 24 ports of FXO in a single GW.  Best performing T1 cards in
> > my experience (thousands of deployments) are Sangoma.  Your server
> > configuration looks fine.
> >
> > Cory J. Andrews
> > Director New Market Initiatives
> >
> > Sayers Media Group
> > VoIP Supply, LLC
> > 454 Sonwil Drive
> > Buffalo, NY 14225
> > 716-250-3402 OFFICE
> > 716-630-1548 FAX
> > 716-601-4474 MOBILE
> > candrews at sayersmedia.com <http://mc/compose?to=candrews at sayersmedia.com>
> >
> >
> > Have I exceeded your expectations?  Please share your experience
> > with my boss,  Benjamin P. Sayers, CEO
> >
> > NOTICE: The information contained in this email and any document
> > attached hereto is intended only for the named recipient(s). It is
> > the property of the VoIP Supply, LLC and shall not be used,
> > disclosed or reproduced without the express written consent of VoIP
> > Supply, LLC. If you are not the intended recipient, nor the employee
> > or agent responsible for delivering this message in confidence to
> > the intended recipient(s), you are hereby notified that you have
> > received this transmittal in error, and any review, dissemination,
> > distribution or copying of this transmittal or its attachments is
> > strictly prohibited. If you have received this transmittal and/or
> > attachments in error, please notify me immediately by reply e-mail
> > or telephone and then delete this message, including any
> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
> > 14225 USA.
>
>
>
> >
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>]
> On Behalf Of
> > Samuel Mukoti
> > Sent: Wednesday, November 25, 2009 2:40 PM
> > To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
>
> > Subject: [Freeswitch-users] Grandstream gateways
> >
> > Hi all,
> >
> > I'm wanting to try out a my first large scale setup at the office, 200
> > extensions and 24 POTS incoming, also a T1 line once the telco guys
> > are ready.  I wanted assistance with choosing the most appropriate
> > hardware.  We already have about 150 analogue phones, and I was
> > wondering what's best? A couple of grandstream FXS GXW4024? Also for
> > my POTS lines, gxw4108  FXO gateway or is it better to buy a sangoma
> > or digium card? The best voice quality is paramount. Lastly for T1
> > what cards are recommeded,
> >
> > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM,
> > would that perform? Or do I need hardware transcoding?
> >
> > Thank you,
> >
> > Sam
> >
> > Twitter: twitter.com/samuelmukoti
> >
> >
> > On 25 Nov,2009, at 8:05 PM,
> freeswitch-users-request at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-request at lists.freeswitch.org>
> >  wrote:
> >
> >> Send FreeSWITCH-users mailing list submissions to
> >>   freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> or, via email, send a message with subject or body 'help' to
> >>   freeswitch-users-request at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-request at lists.freeswitch.org>
> >>
> >> You can reach the person managing the list at
> >>   freeswitch-users-owner at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-owner at lists.freeswitch.org>
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of FreeSWITCH-users digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >>  1. Re: mod_conference kick to abort invitations (Michael Jerris)
> >>  2. Re: Handling the 302 Moved Temporarily response    from
> >>     JavaScript (Michael Jerris)
> >>  3. Re: No NOTIFY MWI when registering via proxy. (Brian West)
> >>  4. Re: remote_media_ip variable not set (Michael Jerris)
> >>  5. Re: How to find whether the destination    extension supports
> >>     encryption (Michael Jerris)
> >>  6. Re: Bypass_media and re_invite (srinivasula reddy)
> >>  7. Re: Handling the 302 Moved Temporarily response    from
> >>     JavaScript (Stephen Crosby)
> >>  8. Re: Handling the 302 Moved Temporarily response    from
> >>     JavaScript (Tihomir Culjaga)
> >>
> >>
> >> ---
> >> -------------------------------------------------------------------
> >>
> >> Message: 1
> >> Date: Wed, 25 Nov 2009 12:44:46 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort
> >>   invitations
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com<http://mc/compose?to=1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com>
> >
> >> Content-Type: text/plain; charset="windows-1252"
> >>
> >> Its a feature we don't have, patches welcome.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
> >>
> >>> Hi members,
> >>> I?m controlling freeswitch with the conference module via xmlrpc.
> >>>
> >>> Is it desired that the kick command can only kick users that are
> >>> connected to the conference?
> >>> Is there no chance abort an  invitation?
> >>> The kick command has no effect until the person I invited with the
> >>> dial command is connected.
> >>
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 2
> >> Date: Wed, 25 Nov 2009 12:45:50 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >>   response    from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID: <A8FA625F-16D2-4A9F-B8C4-13343A488777 at jerris.com<http://mc/compose?to=A8FA625F-16D2-4A9F-B8C4-13343A488777 at jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> In trunk there is a sofia profile setting to allow dialplan
> >> processing of 302 responses.  This won't get you back into your same
> >> javascript, but you can probably do something clever from there.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>
> >>>
> >>> I have considered writing JavaScript code to bridge two calls
> >>> together. However, I would like to perform custom handling of the
> >>> 302 Moved Temporarily response. How do I handle the 302 Moved
> >>> Temporarily response if I use JavaScript?
> >>>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 3
> >> Date: Wed, 25 Nov 2009 11:46:05 -0600
> >> From: Brian West <brian at freeswitch.org<http://mc/compose?to=brian at freeswitch.org>
> >
> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via
> >>   proxy.
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org<http://mc/compose?to=0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org>
> >
> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
> >>
> >> Yes an alias will be required for every domain you run on the profile
> >> so it can find it.
> >>
> >> /b
> >>
> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
> >>
> >>> Try an alias on the sip profile.
> >>>
> >>> Mike
> >>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 4
> >> Date: Wed, 25 Nov 2009 12:47:37 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID: <DF3ECA04-0247-40BB-A810-2468F9C4D805 at jerris.com<http://mc/compose?to=DF3ECA04-0247-40BB-A810-2468F9C4D805 at jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> It's possible it does not.  I just added some code to set it on auto-
> >> adjust so it might be there sometimes now.  You might need to add
> >> some code in mod_sofia to add it other times.  Maybe it makes sense
> >> to move that var setting down to switch_rtp.c.  Patches for this
> >> would be welcome.
> >>
> >> Thanks
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote:
> >>
> >>> Hi,
> >>>
> >>> In the case of proxy_media=true, does it gets set at all then?
> >>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 5
> >> Date: Wed, 25 Nov 2009 12:48:39 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] How to find whether the destination
> >>   extension supports encryption
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com<http://mc/compose?to=38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> You can send the call with secure enabled and if it supports it it
> >> will use it.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
> >>
> >>> Hello,
> >>>
> >>> We have a mix of phones that support RTP encryption and those that
> >>> do not. I have to support both types in the meanwhile, and would
> >>> like to have encryption enabled on the relevant leg, even if the
> >>> other leg does not support it (why? one of our ATAs either must
> >>> have it unencrypted or have it encrypted, but cannot have both).
> >>>
> >>> How do I find whether the destination supports encryption? I do not
> >>> want to manage an additional table in the database...
> >>>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 6
> >> Date: Wed, 25 Nov 2009 23:25:01 +0530
> >> From: srinivasula reddy <srinivas.ksvreddy at gmail.com<http://mc/compose?to=srinivas.ksvreddy at gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID:
> >>   <f8af5740911250955x62d66f55h9584582beba76ba0 at mail.gmail.com<http://mc/compose?to=f8af5740911250955x62d66f55h9584582beba76ba0 at mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> HI,
> >> thanks for your reply, my requirement is i am doing failover stuff
> >> with
> >> freeswitch. i dont want cut the calls when freeswitch dies, when
> >> failover
> >> happens mean one freeswitch dies we are going to start the second
> >> freeswitch, i dont want close call intiated by the  first
> >> freeswtich, they
> >> are communicating with meida(bypass media). when one endpoing try to
> >> end the
> >> call at that time i want to close the call for the other end also.
> >>
> >>
> >> srinivas
> >>
> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> wrote:
> >>
> >>> FreeSWITCH will kill the calls when you shut it down, if you
> >>> intentionally
> >>> kill the network without shutting down FreeSWITCH the only thing
> >>> you can do
> >>> is enable session timers or rtp timers in the soft phones to kill
> >>> the call
> >>> when FreeSWITCH dies or when the call is over.
> >>>
> >>> Mike
> >>>
> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote:
> >>>
> >>>> Hi All,
> >>>>
> >>>> goodmorning to all, i have a scenario, two pjsua clients are
> >>>> connected
> >>> with Freeswitch and they are in call and bypass_media=true.  i
> >>> close the
> >>> Freeswitch server, still they are in call, again i started the
> >>> Freeswitch,
> >>> and registerd these two endpoints, now how can i end the call
> >>> (estabilished
> >>> by the first Freeswitch)? if i call re_invite will it estabilish
> >>> the call
> >>> between two endpoints?
> >>>> any idea?
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Srinivasula Reddy K
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 7
> >> Date: Wed, 25 Nov 2009 10:01:14 -0800
> >> From: Stephen Crosby <stevecrozz at gmail.com<http://mc/compose?to=stevecrozz at gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >>   response    from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID:
> >>   <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com<http://mc/compose?to=11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="utf-8"
> >>
> >> Surprisingly, I've found no way to access the HTTP response status
> >> code
> >> using mod_spidermonkey_curl. I'd love to see this feature added or
> >> discussed
> >> if it already exists and I'm missing it.
> >>
> >> --Stephen
> >>
> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> wrote:
> >>
> >>> In trunk there is a sofia profile setting to allow dialplan
> >>> processing of
> >>> 302 responses.  This won't get you back into your same javascript,
> >>> but you
> >>> can probably do something clever from there.
> >>>
> >>> Mike
> >>>
> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>>
> >>>>
> >>>> I have considered writing JavaScript code to bridge two calls
> >>>> together.
> >>> However, I would like to perform custom handling of the 302 Moved
> >>> Temporarily response. How do I handle the 302 Moved Temporarily
> >>> response if
> >>> I use JavaScript?
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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 --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 8
> >> Date: Wed, 25 Nov 2009 19:04:56 +0100
> >> From: Tihomir Culjaga <tculjaga at gmail.com<http://mc/compose?to=tculjaga at gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >>   response    from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >> Message-ID:
> >>   <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com<http://mc/compose?to=65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> this is how i do it from the dialplan:
> >>
> >>
> >>
> >>
> >>  <extension name="ServiceLookup">
> >>     <condition field="destination_number"
> >> expression="^(300030)(.*)|^\+(300030)(.*)">
> >>
> >>        <action application="set" data="bPfx=$1$3"/>
> >>        <action application="set" data="bNum=$2$4"/>
> >>
> >>        <action inline="true" application="set"
> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/>
> >>        <action application="set"
> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number:
> >> 1:32} :
> >> ${caller_id_number})}"/>
> >>
> >>        <action inline="true" application="set"
> >> data="aPfx=${caller_id_number:0:6}"/>
> >>        <action inline="true" application="set"
> >> data="aNum=${caller_id_number:6:16}"/>
> >>        <action inline="true" application="set"
> >> data="IP_ADDR=${network_addr}:5060"/>
> >>
> >>        <action application="lookup_service_destination" data="in $
> >> {aNum},
> >>                                                               in $
> >> {aPfx},
> >>                                                               in $
> >> {bNum},
> >>                                                               in $
> >> {bPfx},
> >>                                                               in
> >> ${IP_ADDR},
> >>                                                               out
> >> redContact,
> >>                                                               out
> >> authResult"/>
> >>
> >>        <action application="log" data="INFO ########################
> >> ServiceLookup ########################\n"/>
> >>        <action application="log" data="INFO ########################
> >> contact = '${redContact}' ##############\n"/>
> >>        <action application="log" data="INFO ########################
> >> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
> >>        <action application="log" data="INFO ########################
> >> RADIUS auth = '${authResult}' ##########\n"/>
> >>
> >>        <action application="execute_extension" data="doRedirect XML
> >> public"/>
> >>       </condition>
> >>  </extension>
> >>
> >>
> >>  <extension name="doRedirect">
> >>     <condition field="destination_number" expression="^doRedirect$"/>
> >>     <condition field="${authResult}" expression="^0$|">
> >>        <action application="log" data="INFO ########################
> >> RADIUS auth OK!!!' ##########\n"/>
> >>        <action application="redirect" data="${red_contact}"/>
> >>        <anti-action application="log" data="INFO
> >> ########################
> >> RADIUS auth NOK!! ##########\n"/>
> >>        <anti-action application="respond" data="403 Forbidden"/>
> >>     </condition>
> >>
> >>  </extension>
> >>
> >>
> >>
> >>
> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike at jerris.com>
> >
> >> wrote:
> >>
> >>> In trunk there is a sofia profile setting to allow dialplan
> >>> processing of
> >>> 302 responses.  This won't get you back into your same javascript,
> >>> but you
> >>> can probably do something clever from there.
> >>>
> >>> Mike
> >>>
> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>>
> >>>>
> >>>> I have considered writing JavaScript code to bridge two calls
> >>>> together.
> >>> However, I would like to perform custom handling of the 302 Moved
> >>> Temporarily response. How do I handle the 302 Moved Temporarily
> >>> response if
> >>> I use JavaScript?
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> >>> http://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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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> >> http://www.freeswitch.org
> >>
> >>
> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189
> >> *************************************************
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> > http://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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
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>
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>
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From Prometheus001 at gmx.net  Fri Apr 23 02:39:10 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 23 Apr 2010 11:39:10 +0200
Subject: [Freeswitch-users] effective_caller_id_name not working while
 calling multiple endpoints with :_:
In-Reply-To: <l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>
References: <4BCED007.80900@gmx.net>	<66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>
	<l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>
Message-ID: <4BD16ABE.2030506@gmx.net>

Hello Anthony,

I upgraded to newest GIT and tried it

The dialplan now contains the fowllowing:
<action application="bridge" data="<effective_caller_id_name=My
Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain"/>

When the dialplan is executed, it seems to be processed correctly:
EXECUTE sofia/local/06912345678 at 192.168.178.218:5060
bridge(<effective_caller_id_name=My
Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain)
2010-04-23 10:59:12.479598 [DEBUG] switch_ivr_originate.c:1394 variable
string 0 = [effective_caller_id_name=My Name]
2010-04-23 10:59:12.501255 [DEBUG] switch_ivr_originate.c:1885 variable
string 0 = [global_to_originate_1=true]
2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
string 0 = [presence_id=200 at my.domain]
2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
string 1 = [transfer_fallback_extension=200]

However the INVITE message does not contain the caller_id_name, see below


What am I doing wrong?

Best regards
Peter

U 192.168.178.220:5060 -> 192.168.178.50:3072
INVITE sip:200 at 192.168.178.50:3072;line=v3bii5l2 SIP/2.0.
Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bK4jUFH215p85tr.
Max-Forwards: 70.
From: "06912345678" <sip:06912345678 at 192.168.178.220>;tag=a8m8ccQcgjjUg.
To: <sip:200 at 192.168.178.50:3072;line=v3bii5l2>.
Call-ID: b6cafa23-c959-122d-4682-080027e51f59.
CSeq: 129888323 INVITE.
Contact: <sip:mod_sofia at 192.168.178.220:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 348.
X-FS-Support: update_display.
Remote-Party-ID: "06912345678"
<sip:06912345678 at 192.168.178.220>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1272001222 1272001223 IN IP4 192.168.178.220.
s=FreeSWITCH.
c=IN IP4 192.168.178.220.
t=0 0.
m=audio 12096 RTP/AVP 9 0 8 99 3 101 13.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:99 SPEEX/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.

Anthony Minessale schrieb:
> when using enterprise_originate you must use the special leading <>
> brackets to insert global variables meant for each tier 1 originate
>
> <global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/foo at bar.com
> <mailto:foo at bar.com>,sofia/internal/foo2 at bar.com:_:sofia/internal/foo3 at bar3.com
> <mailto:foo3 at bar3.com>
>
>
> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone
> <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>> wrote:
>
>     I think you should first thing update to latest GIT :)
>
>     David Ponzone  Direction Technique
>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>     tel:      01 74 03 18 97
>     gsm:   06 66 98 76 34
>
>     Service Client IPeva
>     tel:      0811 46 26 26
>     www.ipeva.fr  -   www.ipeva-studio.com
>
>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>     utilisation ou diffusion non autoris?e est interdite. Tout message
>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline
>     toute responsabilit? au titre de ce message s'il a ?t? alt?r?,
>     d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>     message, merci de le d?truire imm?diatement et d'avertir
>     l'exp?diteur./
>     /
>     /
>
>
>
>     Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :
>
>>     Setting the effective_caller_id_name when dialing multiple endpoints
>>     with :_: do not seem to work.
>>     See example:
>>      <action application="set" data="effective_caller_id_name=MyName"/>
>>      <action application="bridge"
>>
>>     data="user/30 at mydomain.com
>>     <mailto:user/30 at mydomain.com>:_:user/31 at mydomain.com
>>     <mailto:user/31 at mydomain.com>:_:user/32 at mydomain.com
>>     <mailto:user/32 at mydomain.com>:_:user/33 at mydomain.com
>>     <mailto:user/33 at mydomain.com>:_:user/34 at mydomain.com
>>     <mailto:user/34 at mydomain.com>"/>
>>
>>     Freeswitch tries to set it:
>>       EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414
>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060
>>     set(effective_caller_id_name=MyName)
>>       2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
>>     sofia/external/069xxxxxxxx at 10.xx.xx.1414
>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060 SET
>>     [effective_caller_id_name]=[MyName]
>>
>>     But the SIP messages do not contain the effective_caller_id_name.
>>
>>     If we change the ":_:" sperator to "," then the
>>     effective_caller_id_name
>>     is correctly submittted (hower I cannot call
>>     multiple-registrations on
>>     one number then).
>>
>>     We are on
>>      FreeSWITCH Version 1.0.head (svn-17188)
>>
>>     Any ideas how to overcome this? Or shall I open a JIRA?
>>
>>     Best regards
>>     Peter
>>
>>     See example SIP message:
>>
>>     U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
>>     INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
>>     Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
>>     Max-Forwards: 70.
>>     From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
>>     To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
>>     Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
>>     CSeq: 129803060 INVITE.
>>     Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>     REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
>>     Supported: timer, precondition, path, replaces.
>>     Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>     include-session-description, presence.winfo, message-summary, refer.
>>     Content-Type: application/sdp.
>>     Content-Disposition: session.
>>     Content-Length: 920.
>>     X-FS-Support: update_display.
>>     Remote-Party-ID: "069xxxxxxxx"
>>     <sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
>>     .
>>     v=0.
>>     o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
>>     s=FreeSWITCH.
>>     c=IN IP4 10.xx.xx.141.
>>     t=0 0.
>>     m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9
>>     121 3
>>     101 13.
>>     a=rtpmap:115 G7221/32000.
>>     a=fmtp:115 bitrate=48000.
>>     a=rtpmap:96 AMR/8000.
>>     a=fmtp:96 octet-align=0.
>>     a=rtpmap:99 SPEEX/32000.
>>     a=rtpmap:18 G729/8000.
>>     a=rtpmap:4 G723/8000.
>>     a=rtpmap:7 LPC/8000.
>>     a=rtpmap:124 G726-16/8000.
>>     a=rtpmap:8 PCMA/8000.
>>     a=rtpmap:6 DVI4/16000.
>>     a=rtpmap:123 G726-24/8000.
>>     a=rtpmap:0 PCMU/8000.
>>     a=rtpmap:10 L16/22050.
>>     a=rtpmap:98 iLBC/8000.
>>     a=fmtp:98 mode
>>     #
>>
>>
>>
>>     _______________________________________________
>>     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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
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> http://www.freeswitch.org
>   


From steve.d.ward at gmail.com  Fri Apr 23 05:14:52 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Fri, 23 Apr 2010 08:14:52 -0400
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
	<alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
Message-ID: <g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>

To give one example, display updates are very useful in the application I'm
using.

I have operators using speed dials for dialing out to dialplan that does a
fifo out; so I see what the speed dial is when I press it (e.g.
Operator_Call); but then when the operator is connected to the actual
caller, the display updates to the CID of the caller she's talking to - very
useful.

Also, people using a SIP phone to call into a fifo see the number they
dialed (the number that gets them into the fifo).  But when they're answered
by an actual phone, their display updates to the number of the endpoint
they're talking to.



On Thu, Apr 22, 2010 at 10:31 PM, Wellie Chao <wchao at yahoo.com> wrote:

> As I mentioned in my last email, the ignore_display_updates variable did
> the trick, but I am just curious: what is the purpose of the default
> behavior where the display changes to the callee name and number? I am
> guessing there must be some use case (in fact a fairly prevalent use case)
> where showing the callee is more desirable than showing the caller, but
> I'm not aware of what that might be and am curious to find out what the
> use case(s) are. Changing the display to callee also seems to affect call
> logs on the Polycom, which makes the received call log kind of useless.
>
>
> Date: Thu, 22 Apr 2010 22:57:53 +0300
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
>
> Try adding the following line in your diaplan before the bridge command:
>
> <action application="set" data="ignore_display_updates=true"/>
> (try true and false and see which one works better for you).
>
>                            __Yehavi:
> 2010/4/22 Wellie Chao <wchao at yahoo.com>
>      I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and
> now
>      I see a strange behavior with caller ID on Polycom phones when
> handling
>      inbound calls.
>
>      Here is the scenario:
>
>      * Call is from 212-555-2222 (external number not on my softswitch)
>
>      * Call is to 212-555-1001 (number on my softswitch, extension 1001)
>
>      * extension 1001 is a Polycom phone (it's an IP301, but same problem
>        occurs on other Polycom models such as the 501 and 601). SIP 3.1.4
> and
>        SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.
>
>      * on phone 1001, the caller ID will display 212-555-2222 while
> ringing.
>        The moment I pick up, the display will change to "From: 1001"
> (referring
>        to the extension of the phone itself).
>
>      Has anyone else experienced this problem, and does anyone know how to
> fix
>      it? It does not occur with the snom 320 (and I assume it does not
> occur
>      with any of the snom models based on extrapolation).
>
>      While the problem only started when I updated from 1.0.4 to 1.0.6,
> it's
>      entirely possible it's a configuration setting on the Polycom rather
> than
>      a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
>      resolve the problem (or even how to go about investigating the root
>      cause)?
>
>      _______________________________________________
>      FreeSWITCH-users mailing list
>      FreeSWITCH-users at lists.freeswitch.org
>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/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 wchao at yahoo.com  Fri Apr 23 05:49:16 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Fri, 23 Apr 2010 08:49:16 -0400 (EDT)
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
	<alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
	<g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1004230845540.21297@osgood.microoffice.com>

Yes, I see how display updates are useful on outbound calls now based on 
your examples, but what about use cases for inbound calls? It doesn't seem 
that useful to have the display update to the extension that is called 
after the call is answered (everybody knows their own extension, after 
all). Just wondering why that default behavior was selected. Is there a 
way to leave display updates turned on for the outbound leg of the call 
and to turn them off for the inbound leg of the call?



Date: Fri, 23 Apr 2010 08:14:52 -0400
From: Steven Ward <steve.d.ward at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom

To give one example, display updates are very useful in the application I'm using.

I have operators using speed dials for dialing out to dialplan that does a fifo out; so I see what the speed dial is when I
press it (e.g. Operator_Call); but then when the operator is connected to the actual caller, the display updates to the CID of
the caller she's talking to - very useful.

Also, people using a SIP phone to call into a fifo see the number they dialed (the number that gets them into the fifo).? But
when they're answered by an actual phone, their display updates to the number of the endpoint they're talking to.



On Thu, Apr 22, 2010 at 10:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
      As I mentioned in my last email, the ignore_display_updates variable did
      the trick, but I am just curious: what is the purpose of the default
      behavior where the display changes to the callee name and number? I am
      guessing there must be some use case (in fact a fairly prevalent use case)
      where showing the callee is more desirable than showing the caller, but
      I'm not aware of what that might be and am curious to find out what the
      use case(s) are. Changing the display to callee also seems to affect call
      logs on the Polycom, which makes the received call log kind of useless.


      Date: Thu, 22 Apr 2010 22:57:53 +0300
      From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
      Reply-To: freeswitch-users at lists.freeswitch.org
      To: freeswitch-users at lists.freeswitch.org
      Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom

      Try adding the following line in your diaplan before the bridge command:
      ?
      <action application="set" data="ignore_display_updates=true"/>
      (try true and false and see which one works better for you).
      ?
      ?????????????????????????? __Yehavi:
      2010/4/22 Wellie Chao <wchao at yahoo.com>
      ? ? ?I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now
      ? ? ?I see a strange behavior with caller ID on Polycom phones when handling
      ? ? ?inbound calls.

      ? ? ?Here is the scenario:

      ? ? ?* Call is from 212-555-2222 (external number not on my softswitch)

      ? ? ?* Call is to 212-555-1001 (number on my softswitch, extension 1001)

      ? ? ?* extension 1001 is a Polycom phone (it's an IP301, but same problem
      ? ? ?? occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
      ? ? ?? SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.

      ? ? ?* on phone 1001, the caller ID will display 212-555-2222 while ringing.
      ? ? ?? The moment I pick up, the display will change to "From: 1001" (referring
      ? ? ?? to the extension of the phone itself).

      ? ? ?Has anyone else experienced this problem, and does anyone know how to fix
      ? ? ?it? It does not occur with the snom 320 (and I assume it does not occur
      ? ? ?with any of the snom models based on extrapolation).

      ? ? ?While the problem only started when I updated from 1.0.4 to 1.0.6, it's
      ? ? ?entirely possible it's a configuration setting on the Polycom rather than
      ? ? ?a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
      ? ? ?resolve the problem (or even how to go about investigating the root
      ? ? ?cause)?

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


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.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 Apr 23 06:34:45 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 23 Apr 2010 08:34:45 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
	<p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
Message-ID: <55FBE664-764D-4EA1-9B48-B8FB8949D85C@freeswitch.org>


On Apr 23, 2010, at 12:30 AM, Gabriel Kuri wrote:

> Send proof the machine is dead? Would you like a picture of smoke pouring out of a box in flames? 

A signed paper stating that its failed is usually good enough.  Again right now I'm letting you get away with it three times.  If you have it happen 10 times a month for 3 months in a row thats a suspect.... but if you email me and its happened 12 months or more part then i'm more than willing to let it slide more than 3 times.  I just won't let it be abused.

> What if you want to re-allocate the machine to do something else or just replace it with a more powerful machine? How are the licenses tied to the machine, MAC address? 

See above.

> Any reason the "option" for a floating license model isn't available? It seems given the option between the two, I'd rather have the floating license model so I wouldn't need to prove my machine is dead before asking to have the licenses re-issued, unless of course the license server is dead. We swap production machines in/out all the time, particularly because we perform maintenance on one, so we have a spare, bring it up, take the other one down, perform maintenance, and bring it back up.

Floating license server isn't out there because you have more chances for things to go wrong and calls to be dropped due to issues related to reaching the server.  The decision was made to not do that.  Have the calls just work is more critical to our carrier customers.

> With this model, floating licenses would be our only option, I really wouldn't want to be purchasing a bunch of extra licenses for spare machines.
> 
> I can't imagine you guys really like making us feel like criminals by tieing the license to an actual box, what's wrong with the good 'ole "on your honor policy"? I realize the need to pay for valid g729 licenses, but prove the machine is dead? Is this requirement coming from Sipro?

We have to make reasonable efforts, its a requirement.

> 
> Cheers,
> Gabe



From steveayre at gmail.com  Fri Apr 23 07:01:34 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Apr 2010 15:01:34 +0100
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com>
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com>
	<v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com>
	<s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com>
	<n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com>
Message-ID: <o2qeafa772f1004230701w8cb58413na7400a0436b17878@mail.gmail.com>

There's no siptrace in that...

On 22 April 2010 23:16, Neil Patel <neilp at cs.stanford.edu> wrote:
> Thanks for clarifying.
> Here's the output (with siptrace on) when trying to dial out over the
> gateway:
> 2010-04-22 15:13:10.104866 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1001 at server.com [a6aac86b-4a40-41f3-8968-f9b37501483a]
> 2010-04-22 15:13:10.112850 [INFO] mod_dialplan_xml.c:418 Processing
> neilp->5001 in context default
> 2010-04-22 15:13:10.120874 [INFO] switch_core_session.c:1750 Sending early
> media
> 2010-04-22 15:13:10.122856 [NOTICE] mod_sofia.c:1907 Pre-Answer
> sofia/internal/1001 at server.com!
> 2010-04-22 15:13:10.123810 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1234 at conference.freeswitch.org
> [af13cf37-eb6a-4e9f-b1cc-366b6a24aec5]
> 2010-04-22 15:13:10.386879 [INFO] switch_rtp.c:2049 Auto Changing port from
> IP1 to IP2
> 2010-04-22 15:13:10.697870 [NOTICE] sofia.c:4789 Hangup
> sofia/internal/1234 at conference.freeswitch.org [CS_CONSUME_MEDIA]
> [NO_USER_RESPONSE]
> 2010-04-22 15:13:10.706880 [ERR] mod_conference.c:4563 Cannot create
> outgoing channel, cause: NO_USER_RESPONSE
> 2010-04-22 15:13:10.706880 [NOTICE] mod_conference.c:4566 Hangup
> sofia/internal/1001 at server.com [CS_EXECUTE] [NO_USER_RESPONSE]
> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1179 Session 2
> (sofia/internal/1234 at conference.freeswitch.org) Ended
> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1179 Session 1
> (sofia/internal/1001 at server.com) Ended
> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/internal/1001 at server.com [CS_DESTROY]
>
> ?In case it is relevant, the call is triggered in a lua script which is
> executed by dialing a extension 5001. The dialplan in
> freeswitch/conf/dialplan/default/5001_gizmo.xml:
> ?<extension name="gizmo">
> ?? ? ?<condition field="destination_number" expression="^5001$">
> ?? ? ?<action application="lua" data="makecall.lua"/>
> ?? ?</condition>
> ??</extension>
> makecall.lua:
> function make_call()
> ??sessiondata = "sofia/gateway/gizmo/<num>"
> ??new_session = freeswitch.Session(sessiondata)
> end
> make_call()
>
>
> On Thu, Apr 22, 2010 at 2:58 PM, William Suffill <william.suffill at gmail.com>
> wrote:
>>
>> gizmo is your gateway.
>>
>>
>> siptrace is by profile.
>>
>>
>> sofia status in the cli will list your profiles/gateways (default cfgs use
>> internal/external)
>>
>> so in that case sofia profile external siptrace on
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From btsteve at yahoo.com  Thu Apr 22 22:30:00 2010
From: btsteve at yahoo.com (Travis Stevens)
Date: Thu, 22 Apr 2010 22:30:00 -0700 (PDT)
Subject: [Freeswitch-users] One Way Audio on Some calls
Message-ID: <413285.30918.qm@web30208.mail.mud.yahoo.com>

Has any one seen an issue where an endpoint answers and it take literally 10 seconds before the server forwards the signaling to the next server upstream. It does not happen every call, but the ip that the server is communicating to is the same if the call is good or bad. If the call is bad after the 10 seconds the remote end of connected and there is one way audio.

The Only difference i have been able to see in the logs is that the call that completes correctly returns a lines stating that stun is not needed because the ports match. 

I am running FreeSWITCH Version 1.0.head (svn-17188M)

Freeswitch has public IP addresses and the Endpoint are behind nat. I have tested multiple endpoint with the same results.

Thanks in advance for any help.


      


From patrick at speechpro.com  Fri Apr 23 02:41:38 2010
From: patrick at speechpro.com (patrick)
Date: Fri, 23 Apr 2010 13:41:38 +0400
Subject: [Freeswitch-users] Where i can get help with freeswitch?
Message-ID: <4BD16B52.9060800@speechpro.com>

Hello from St.Petersburg!
My name is Patrick.
I try to realise IVR with ASR & TTS.
Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local 
product "Voicenavigator" (mrcp server, ASR, TTS).

I have 2 problems:

1. How to realise "barge in" for playback and TTS? Does freeswitch allow 
that?
I need to start playback, or TTS and break it when some speech is 
detected (by ASR)...

2. When I call from Asterisk to Freeswitch and bridge my call back to 
Asterisk (to another number), there only noise in first asterisk 
abonent's phone...

I would be grateful for any help ;-)

-- 
? ?????????,

?????????? ?????? ??????????
??????? ?? ????????????
??? ?????? ??????? ???????????
???.: (812) 325-8848, ???. 6225
????: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru



From pjintheusa at gmail.com  Fri Apr 23 10:55:04 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 23 Apr 2010 13:55:04 -0400
Subject: [Freeswitch-users] Any breaking change on register with 1.0.6
Message-ID: <w2m367751821004231055h8b2ea90cxe0046f3c61ab59f3@mail.gmail.com>

Hi there,

I have been using my Cisco phones registered with FreeSWITCH (1.0.4) for
months now. Today I upgraded to 1.0.6 and they no longer registers.

This what configured to get them to work.

1) Add

 <list name="lan" default="allow">
        <node type="allow" cidr="71.**5.60.***/32"/>

to acl.conf.xml

2) Add the appropriate directory entry.


That worked (may be it shouldn't have done) - but on 1.0.6 (today's trunk)
is fails.

Is something else required in 1.0.6 do you know?


Thanks


Phil
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From cmrienzo at gmail.com  Fri Apr 23 11:59:33 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 23 Apr 2010 14:59:33 -0400
Subject: [Freeswitch-users] Where i can get help with freeswitch?
In-Reply-To: <4BD16B52.9060800@speechpro.com>
References: <4BD16B52.9060800@speechpro.com>
Message-ID: <k2n3687b0701004231159ib220cea6laa0f69708bd9ff10@mail.gmail.com>

Barge-in will work out of the box for digits... to make it work for ASR is a
bit more complicated.

I don't know what method you are using to do TTS, but it this is the general
idea:

   1. set up a handler to deal with input callbacks
   2. on DTMF or start of speech, return "break" to cause barge-in.

My help can be more specific if you tell me more about what method
(dialplan, Lua, javascript, etc) you are using to execute TTS and ASR.

Can't help you on the noise issue... someone else needs to chime in.



On Fri, Apr 23, 2010 at 5:41 AM, patrick <patrick at speechpro.com> wrote:

> Hello from St.Petersburg!
> My name is Patrick.
> I try to realise IVR with ASR & TTS.
> Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local
> product "Voicenavigator" (mrcp server, ASR, TTS).
>
> I have 2 problems:
>
> 1. How to realise "barge in" for playback and TTS? Does freeswitch allow
> that?
> I need to start playback, or TTS and break it when some speech is
> detected (by ASR)...
>
> 2. When I call from Asterisk to Freeswitch and bridge my call back to
> Asterisk (to another number), there only noise in first asterisk
> abonent's phone...
>
> I would be grateful for any help ;-)
>
> --
> ? ?????????,
>
> ?????????? ?????? ??????????
> ??????? ?? ????????????
> ??? <<????? ??????? ??????????>>
> ???.: (812) 325-8848, ???. 6225
> ????: (812) 327-9297
> E-mail: patrick at speechpro.com
> http://www.speechpro.ru
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stevendt at primrosebank.net  Fri Apr 23 13:23:25 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 23 Apr 2010 21:23:25 +0100
Subject: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
References: <4B281EC072E8405A9B2ACF2925901D65@bp1.ad.bp.com>
Message-ID: <0405D805B8EA40CD9288FF4AF9974963@bp1.ad.bp.com>

Hi,

I'm still struggling to get the VS2008 Express Build working, but there seems to be another problem with mod_skinny

Mathieu,

mod_skinny.c has a #include looking for skinny_api.h, the file does not seem to be there ?


regards
Dave
  ----- Original Message ----- 
  From: Dave Stevenson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, April 15, 2010 5:32 PM
  Subject: [Freeswitch-users] mod_skinny - Building with VS 2008 Express


  Hi,

  I'm trying to use mod_skinny under windows but can't get the Visual Studio 2008 Express build configuration working. 

  I know that Mathieu does not run Windows, but can anyone else modify the VS build solution in source to include building mod_skinny please ?

  regards
  Dave


------------------------------------------------------------------------------


  _______________________________________________
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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 nandy1925 at gmail.com  Fri Apr 23 16:27:43 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sat, 24 Apr 2010 07:27:43 +0800
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com>
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com>
	<v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com>
	<s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com>
	<n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com>
Message-ID: <j2j7d0bfd8c1004231627sbdfe60c5o8c474c7bcfb197b7@mail.gmail.com>

neil,

in makecall.lua, you're creating a new session. don't. when makecall.lua is
called, it's already in a session.  try this one:
    session:execute("bridge","sessiondata")

for flexibility,  it's better to pass the URI to the lua script at the
dialplan.

-nandy

On Fri, Apr 23, 2010 at 6:16 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks for clarifying.
>
> Here's the output (with siptrace on) when trying to dial out over the
> gateway:
>
> 2010-04-22 15:13:10.104866 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1001 at server.com [a6aac86b-4a40-41f3-8968-f9b37501483a]
> 2010-04-22 15:13:10.112850 [INFO] mod_dialplan_xml.c:418 Processing
> neilp->5001 in context default
> 2010-04-22 15:13:10.120874 [INFO] switch_core_session.c:1750 Sending early
> media
> 2010-04-22 15:13:10.122856 [NOTICE] mod_sofia.c:1907 Pre-Answer
> sofia/internal/1001 at server.com!
> 2010-04-22 15:13:10.123810 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1234 at conference.freeswitch.org[af13cf37-eb6a-4e9f-b1cc-366b6a24aec5]
> 2010-04-22 15:13:10.386879 [INFO] switch_rtp.c:2049 Auto Changing port from
> IP1 to IP2
> 2010-04-22 15:13:10.697870 [NOTICE] sofia.c:4789 Hangup sofia/internal/
> 1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
> 2010-04-22 15:13:10.706880 [ERR] mod_conference.c:4563 Cannot create
> outgoing channel, cause: NO_USER_RESPONSE
> 2010-04-22 15:13:10.706880 [NOTICE] mod_conference.c:4566 Hangup
> sofia/internal/1001 at server.com [CS_EXECUTE] [NO_USER_RESPONSE]
> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1179 Session 2
> (sofia/internal/1234 at conference.freeswitch.org) Ended
> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1179 Session 1
> (sofia/internal/1001 at server.com) Ended
> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/internal/1001 at server.com [CS_DESTROY]
>
>
>  In case it is relevant, the call is triggered in a lua script which is
> executed by dialing a extension 5001. The dialplan in
> freeswitch/conf/dialplan/default/5001_gizmo.xml:
>  <extension name="gizmo">
>       <condition field="destination_number" expression="^5001$">
>       <action application="lua" data="makecall.lua"/>
>     </condition>
>   </extension>
>
> makecall.lua:
> function make_call()
>   sessiondata = "sofia/gateway/gizmo/<num>"
>   new_session = freeswitch.Session(sessiondata)
> end
> make_call()
>
>
>
> On Thu, Apr 22, 2010 at 2:58 PM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> gizmo is your gateway.
>>
>>
>> siptrace is by profile.
>>
>>
>> sofia status in the cli will list your profiles/gateways (default cfgs use
>> internal/external)
>>
>> so in that case sofia profile external siptrace on
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Fri Apr 23 20:04:47 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Fri, 23 Apr 2010 20:04:47 -0700
Subject: [Freeswitch-users] outbound sip call: NO_USER_RESPONSE
In-Reply-To: <j2j7d0bfd8c1004231627sbdfe60c5o8c474c7bcfb197b7@mail.gmail.com>
References: <i2me5503bdb1004211510m37a8840fl36d35d08dae9b1d1@mail.gmail.com> 
	<s2w87f2f3b91004221351p33c7b44bl4ed90a1bf019496b@mail.gmail.com> 
	<v2re5503bdb1004221446k291751e6hecabe6660a059c3a@mail.gmail.com> 
	<s2m6b65470d1004221458ud0aaab34mcb45fd61e7e65cfa@mail.gmail.com> 
	<n2ye5503bdb1004221516tdcd3742dof25f8fa8d09e824c@mail.gmail.com> 
	<j2j7d0bfd8c1004231627sbdfe60c5o8c474c7bcfb197b7@mail.gmail.com>
Message-ID: <h2ke5503bdb1004232004q8db9ce44y3bc647c437a6a135@mail.gmail.com>

As I suspected, the problem was a silly one. Extension 5001 is reserved by
dynamic_conference in the default dialplan. Once I changed to an unused
extension it works fine.

Nandy, once I bridge a call, how do I stream audio to the external number?
When I try session:streamfile (after checking session:ready() == true), the
file plays to the initiating FS extension, but not out to the external
number. Any suggestions?

Thanks,
Neil

On Fri, Apr 23, 2010 at 4:27 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> neil,
>
> in makecall.lua, you're creating a new session. don't. when makecall.lua is
> called, it's already in a session.  try this one:
>     session:execute("bridge","sessiondata")
>
> for flexibility,  it's better to pass the URI to the lua script at the
> dialplan.
>
> -nandy
>
>
> On Fri, Apr 23, 2010 at 6:16 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Thanks for clarifying.
>>
>> Here's the output (with siptrace on) when trying to dial out over the
>> gateway:
>>
>> 2010-04-22 15:13:10.104866 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1001 at server.com [a6aac86b-4a40-41f3-8968-f9b37501483a]
>> 2010-04-22 15:13:10.112850 [INFO] mod_dialplan_xml.c:418 Processing
>> neilp->5001 in context default
>> 2010-04-22 15:13:10.120874 [INFO] switch_core_session.c:1750 Sending early
>> media
>> 2010-04-22 15:13:10.122856 [NOTICE] mod_sofia.c:1907 Pre-Answer
>> sofia/internal/1001 at server.com!
>> 2010-04-22 15:13:10.123810 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1234 at conference.freeswitch.org[af13cf37-eb6a-4e9f-b1cc-366b6a24aec5]
>>  2010-04-22 15:13:10.386879 [INFO] switch_rtp.c:2049 Auto Changing port
>> from IP1 to IP2
>> 2010-04-22 15:13:10.697870 [NOTICE] sofia.c:4789 Hangup sofia/internal/
>> 1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
>> 2010-04-22 15:13:10.706880 [ERR] mod_conference.c:4563 Cannot create
>> outgoing channel, cause: NO_USER_RESPONSE
>> 2010-04-22 15:13:10.706880 [NOTICE] mod_conference.c:4566 Hangup
>> sofia/internal/1001 at server.com [CS_EXECUTE] [NO_USER_RESPONSE]
>> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1179 Session 2
>> (sofia/internal/1234 at conference.freeswitch.org) Ended
>> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1181 Close
>> Channel sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
>> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1179 Session 1
>> (sofia/internal/1001 at server.com) Ended
>> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1181 Close
>> Channel sofia/internal/1001 at server.com [CS_DESTROY]
>>
>>
>>  In case it is relevant, the call is triggered in a lua script which is
>> executed by dialing a extension 5001. The dialplan in
>> freeswitch/conf/dialplan/default/5001_gizmo.xml:
>>  <extension name="gizmo">
>>       <condition field="destination_number" expression="^5001$">
>>       <action application="lua" data="makecall.lua"/>
>>     </condition>
>>   </extension>
>>
>> makecall.lua:
>> function make_call()
>>   sessiondata = "sofia/gateway/gizmo/<num>"
>>    new_session = freeswitch.Session(sessiondata)
>> end
>> make_call()
>>
>>
>>
>> On Thu, Apr 22, 2010 at 2:58 PM, William Suffill <
>> william.suffill at gmail.com> wrote:
>>
>>> gizmo is your gateway.
>>>
>>>
>>> siptrace is by profile.
>>>
>>>
>>> sofia status in the cli will list your profiles/gateways (default cfgs
>>> use internal/external)
>>>
>>> so in that case sofia profile external siptrace on
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 23 20:34:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 23 Apr 2010 23:34:12 -0400
Subject: [Freeswitch-users] Transferring to Non-Existent Extension
In-Reply-To: <o2x87f2f3b91004221720k7799f72bu30e8626e8da3b928@mail.gmail.com>
References: <6FD41EA060CF48528942723112A339B0@greyhawk.tonecommander.com>
	<8E865498-5659-475E-84F6-3A80EDAF4E35@freeswitch.org>
	<o2x87f2f3b91004221720k7799f72bu30e8626e8da3b928@mail.gmail.com>
Message-ID: <4C075AB6-20CE-45B1-AE3D-2521B543AA0B@jerris.com>

its an extension the channel falls back to on failed transfer.  Set it before the transfer, transfer to some invalid or unavailable ext, and it will "fall back" to the ext set in the var.

Mike

On Apr 22, 2010, at 8:20 PM, Michael Collins wrote:

> 
> 
> On Thu, Apr 22, 2010 at 3:34 PM, Brian West <brian at freeswitch.org> wrote:
> the handy variable transfer_fallback_extension might help.
> 
> FYI, this isn't in the wiki. If someone would give me a brief desc of how it works I will add it to the wiki.
> Thanks,
> MC

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From mike at jerris.com  Fri Apr 23 20:36:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 23 Apr 2010 23:36:23 -0400
Subject: [Freeswitch-users] Parsing XML files from Spidermonkey
In-Reply-To: <j2td021d3be1004222055xa8aa8d75odc08df002465e6b7@mail.gmail.com>
References: <48fade7f8a99c447d71334d2d3e589bd.squirrel@correo.nodo50.org>
	<14f6fde22ecd0ad9b6fdd7a7fe56e3d0.squirrel@correo.nodo50.org>
	<j2td021d3be1004222055xa8aa8d75odc08df002465e6b7@mail.gmail.com>
Message-ID: <8374F99F-9C0C-4700-8E89-5BFBF337B28B@jerris.com>

actually, spidermonkey has an xml parser, its very poorly documented:

http://wiki.freeswitch.org/wiki/Javascript_XML

On Apr 22, 2010, at 11:55 PM, Jonas Gauffin wrote:

> Not possible.
> 
> http://markmail.org/message/ctietfw2mucucdyq#query:spidermonkey%20domparser+page:1+mid:jjknnksvfpi4keto+state:results
> 
> http://groups.google.com/group/mozilla.dev.tech.js-engine/browse_thread/thread/86b27b854492c22a/237e4ba109d3f69c
> 
> On Thu, Apr 22, 2010 at 10:30 AM, Alberto Escudero <aep.lists at it46.se> wrote:
> I am resending this mail, hoping that someone has managed to read a XML
> file from Javascript.
> 
> 
> --
> Stopping junk mailers is good for the environment
> 
> > Hi,
> >
> > After one year using FS i am starting to like XML so i am trying to get
> > a Javascript script to read local XML files.
> >
> > I am using the XML method and getting Syntax errors from spidermonkey
> >
> > While something like this works:
> > xmldata = new XML("<tag>foo</tag>");
> >
> > I have not been able to read and parse XML local files, using File or
> > FileIO
> > methods
> >
> > A simple example like this returns Syntax error.
> > var foo = apiExecute ("show", "channels as xml");
> > xmldata = new XML(foo);
> >
> > Has anyone managed to use the XML method from spidermonkey to read a XML
> > stored file?
> >
> > There are some E4X bugs around and i wonder if those are the cause of the
> > "Syntax Error" feedback even reading a very basic XML file
> > --
> > Stopping junk mailers is good for the environment
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at googlemail.com  Sat Apr 24 04:53:44 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 24 Apr 2010 12:53:44 +0100
Subject: [Freeswitch-users] How to set maximum ring duration if call is not
	answered and fake ringing tone
Message-ID: <h2u7abab2411004240453md4ef284czc21e431b03024545@mail.gmail.com>

How do you set the maximum ring duration for a call if it is not anwered?

How do you set a fake ring tone, if there is a delay before call begins to
ring?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From sos at sokhapkin.dyndns.org  Sat Apr 24 05:08:49 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 24 Apr 2010 08:08:49 -0400
Subject: [Freeswitch-users] How to set maximum ring duration if call is
	not answered and fake ringing tone
In-Reply-To: <h2u7abab2411004240453md4ef284czc21e431b03024545@mail.gmail.com>
References: <h2u7abab2411004240453md4ef284czc21e431b03024545@mail.gmail.com>
Message-ID: <201004240808.49465.sos@sokhapkin.dyndns.org>

On Saturday 24 April 2010, Frank Church wrote:
> How do you set the maximum ring duration for a call if it is not anwered?
> 

<action application="set" data="call_timeout=120"/>


> How do you set a fake ring tone, if there is a delay before call begins to
> ring?
> 

It can be done easy, but I suggest to not do it...

<action application="set" data="ringback=$${us-ring}"/>


From mike at jerris.com  Sat Apr 24 11:08:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 14:08:26 -0400
Subject: [Freeswitch-users] need help on IVR
In-Reply-To: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
Message-ID: <7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>

what is a sip server?  Yes, you can do pretty much anything like this.

On Apr 19, 2010, at 12:08 AM, false wrote:

> Hi all
> 
> my network topology:
> 
> endpoint 1(100)-----sip server ---IVR(Freeswitch)
> 					|
> 					|	
> 				endpoint2(101)
> 
> endpoint1 + endpoint2 are registered to sip server
> Freeswitch is regsitered to sip server with 103
> 
> my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
> and RTP from endpoint 1 --> media proxy---> FS
> then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
> and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
> 
> can FS do it
> 
> Thank you
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> __________________________________________________
> B?n C? S? D?ng Yahoo! Kh?ng?
> M?t m?i v? th? r?c? Yahoo! Th? c? ch??ng tr?nh b?o v? ch?ng th? r?c h?u hi?u nh?t tr?n m?ng 
> http://vn.mail.yahoo.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 mike at jerris.com  Sat Apr 24 11:17:24 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 14:17:24 -0400
Subject: [Freeswitch-users] Can FreeSWITCH act as a SIP Proxy with
	transcoding?
In-Reply-To: <4BCC490F.5010701@infosecurity.ch>
References: <4BCC490F.5010701@infosecurity.ch>
Message-ID: <60EAF70A-64A6-45A4-8630-7E9F23AD1CCA@jerris.com>

FreeSWITCH is not a proxy, and proxies don't transcode, they only affect signaling.  You seem to be describing a back to back user agent, which both FreeSWITCH and cudatel are.  All of this functionality is currently available in FreeSWITCH.  We have been working on adding provisioning and configuration support for SRTP and TLS to cudatel, but it is not yet available.  

Mike


On Apr 19, 2010, at 8:14 AM, Fabio Pietrosanti (naif) wrote:

> Hi all,
> 
> i would like to know if FreeSWITCH could be used as a simple SIP Proxy 
> in order to:
> - receive SIP/TLS connections
> - receive SRTP secured connection
> - forward it in "unsecured" way to a backend PBX (plain RTP, plain SIP/UDP)
> - if required configured do simply transcoding before forwarding calls 
> to backend PBX
> 
> So to run it as a "SIP Security Proxy" in front of an existing SIP PBX 
> doing also transcoding if required (in case backend SIP PBX does not 
> support SIP UA codecs) .
> 
> Is this setup feasible with a FreeSWITCH?
> 
> Does anyone know if cudatel (https://www.cudatel.com/) support such kind 
> of configuration scenario?
> 
> Fabio
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 24 11:19:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 14:19:40 -0400
Subject: [Freeswitch-users] Users : Alphanumeric mapping
In-Reply-To: <w2o9d8570731004190620z2484c46ci3105d044f8481da0@mail.gmail.com>
References: <w2o9d8570731004190620z2484c46ci3105d044f8481da0@mail.gmail.com>
Message-ID: <21CC0743-0A40-4E90-90A0-B55ACCC5D511@jerris.com>

you can use the user endpoint and dial-string param on the user.

Mike

On Apr 19, 2010, at 9:20 AM, J?r?me M. wrote:

> Hi everyone,
> 
> I'm new to Freeswitch and just started to configure it.
> 
> I have read the wiki, and my first problem is the following :
> 
> - I want to use alphanumeric id for my users. In other words, they can register to the PBX with a id like firstname.lastname at domain.com and can be reached in this way, no need to know the phone number (300-399 in my case).
> 
> So, i declared my users as mentionned on the wiki, using number-alias. In my dialplan, every call to 300-399 numbers are automatically associated to alphanumeric id's, everything is working fine.
> 
> But in a real context, every user will only know the alphanumeric-id (it's easier to remember) of another users. I thought that the mapping was bidirectionnal, but it seems that I have to match every firstname.lastname at domain.com in my dialplan. 
> 
> I tried, but apparently destination_number contains only firstname.lastname ( no SIP URI, maybe because I use the local domain).
> 
> I' m sure I could do it using a "dirty trick" but i really want to do it in a good way...
> 
> Do you have any ideas ?
> 
> Thank you so much for your help, sorry for my approximative english ;) !

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From mike at jerris.com  Sat Apr 24 11:40:24 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 14:40:24 -0400
Subject: [Freeswitch-users] where file "xmlrpc.inc" gone?
In-Reply-To: <4BCEA13C.5060707@gmail.com>
References: <4BCEA13C.5060707@gmail.com>
Message-ID: <891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>

I just realized I don't think I ever rolled a contrib tarball for 1.0.6.  We split the contrib out of the base repo before the 1.0.6 release.  I will try to get this corrected this weekend and put a traball out on files.freeswitch.org.

Mike



On Apr 21, 2010, at 2:54 AM, Eli Hayun wrote:

> Hi
> 
> I am trying to add some xmlprc functionality to FS. I want to use it
> with PHP. The php example has the following line:
> 
> 
> include("xmlrpc.inc");
> 
> I noticed that in version 1.0.5-xx the file included but not with 1.0.6
> 
> Is there other way to use it with php or the file is wrongly missing ?



From mike at jerris.com  Sat Apr 24 11:47:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 14:47:12 -0400
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
In-Reply-To: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
References: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
Message-ID: <EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>

just uuid_transfer the a leg.

On Apr 21, 2010, at 10:09 AM, Steven Ward wrote:

> Hello all,
> 
> I have a lua script running that checks the state of a call between A and B - the call between A and B was set up through a Polycom's attended transfer feature, so A itself didn't necessarily execute the bridge to B.
> 
> A gets early media from B.  After a certain amount of time, if A is still getting early media, I want the script to end that call between A and B, and send A through some specific dialplan.
> 
> How does uuid_transfer work for this goal?  I'd like to kill B and move A to specific dialplan.  I don't want B to be in the transfer at all.
> 
> I know there may be many possibilities here; I'm just wondering if anyone can recommend something.  Thanks.
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From mike at jerris.com  Sat Apr 24 12:01:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:01:19 -0400
Subject: [Freeswitch-users] Manage call from Event Socket inbound
In-Reply-To: <4BD07D77.4080200@bio.es>
References: <4BD07D77.4080200@bio.es>
Message-ID: <4B7EEA82-6319-4B24-8440-7B1CBCDF3F6E@jerris.com>

Try changing the following line (3119 in head) from mod_dptools.c

SWITCH_ADD_APP(app_interface, "park", "Park", "Park", park_function, "", SAF_NONE); 

to

SWITCH_ADD_APP(app_interface, "park", "Park", "Park", park_function, "", SAF_SUPPORT_NOMEDIA);

I glanced at the park code and it seems to handle no media okay, but you will need to test it out to be sure there are not any edge cases where this is not the case or look closer through switch_ivr_park in switch_ivr.c to confirm.  If this all seems to be good, feel free to send us a patch to jira.freeswitch.org to make this change.

Mike
 

On Apr 22, 2010, at 12:46 PM, Jose Fco. Irles wrote:

> Hi,
> 
> I would like to manage calls from the Event Socket in inbound mode, connecting to freeswitch and waiting for events and sending commands.
> I don't know how to sleep the call until my logic executes a action in this new call.
> 
> I've tried to park the call with this in the dialplan:
> 
>   <extension name="my_park">
>     <condition field="destination_number" expression="^.+$">
>       <action application="park"/>
>     </condition>
>   </extension>
> 
> but freeswitch "pre answers" the call and my sip client receives a "183 Session progress". I want the the call to wait in "100 Trying" until my logic execute something for the new call.
> 
> In outbound mode this works well, when the dialplan executes the "socket" application, the call waits in "100 Trying".
> 
> I prefer to make this in inbound mode because it's more scalable and simple than build a tcp server and manage the calls in outbound mode. However in outbound mode I need inbound mode for some things.



From mike at jerris.com  Sat Apr 24 12:04:44 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:04:44 -0400
Subject: [Freeswitch-users] mod_say or strftime problems?
In-Reply-To: <r2m87f2f3b91004221522ie3e3f63an1cdc581558cb7f0e@mail.gmail.com>
References: <32a6e7fd9231a21d7ac5e57c76e37763.squirrel@webmail.tagnet.ru>
	<201004221512.18576.yivzhenko@mksat.net>
	<34adeff2737457cf22300f3b142145e5.squirrel@webmail.tagnet.ru>
	<r2m87f2f3b91004221522ie3e3f63an1cdc581558cb7f0e@mail.gmail.com>
Message-ID: <E94DB936-E0A0-4AFD-95A5-81F57293018F@jerris.com>

Boris opened http://jira.freeswitch.org/browse/MODAPP-421 .  I'll take a look into this one.

Thanks for the report.

Mike

On Apr 22, 2010, at 6:22 PM, Michael Collins wrote:

> Would you please open a JIRA on this one?
> -MC
> 
> On Thu, Apr 22, 2010 at 6:55 AM, <boris at tagnet.ru> wrote:
> IMHO this is bug, as of struct tm man page:
>  int tm_hour;    /* hours (0 - 23) */
> so tm_hour shouldn't be incremented. But looking at code I see no reason
> why the behavior is changed with different timezone.
> 
> > I have same problem with mod_say_ru if timezone is set to
> >
> > <action application="set" data="timezone=Europe/Kiev"/>
> >
> > And i don't understand why  "tm.tm_hour + 1"
> > In source code
> >         if (say_time) {
> >                 switch_snprintf(buf, sizeof(buf), "%d:%d:%d", tm.tm_hour +
> > 1,
> > tm.tm_min, tm.tm_sec);
> >                 say_args->type = SST_TIME_MEASUREMENT;
> >                 ru_say_time(session, buf, say_args, args);
> >
> > As temporary resolution i just set different timezone :)
> >
> >
> > On Wednesday 21 April 2010 17:30:18 boris at tagnet.ru wrote:
> >> Hello!
> >>
> >> I have an extension for the current date/time:
> >>     <extension name="SayTime">
> >>         <condition field="destination_number" expression="^9000$">
> >>             <action application="answer"/>
> >>             <action application="say" data="ru current_date_time
> >> pronounced $strftime(%s)"/>
> >>             <action application="sleep" data="1000"/>
> >>             <action application="hangup"/>
> >>         </condition>
> >>     </extension>
> >>
> >> The extension works fine, but time is +1 hour of current time. For
> >> example
> >> current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6
> >> (from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer
> >> time in effect now, so tz=YEKST). So, my question is - something wrong
> >> with my extension configuration or this is bug in mod_say_ru or may be
> >> strfmt?
> >>
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From mike at jerris.com  Sat Apr 24 12:07:58 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:07:58 -0400
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <alpine.LRH.2.00.1004230845540.21297@osgood.microoffice.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
	<alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
	<g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>
	<alpine.LRH.2.00.1004230845540.21297@osgood.microoffice.com>
Message-ID: <5A73BA42-86F8-4936-9355-81885F6425F4@jerris.com>

Can you confirm this is still the case in current git head and if so open me a bug on jira about this issue please.

Mike

On Apr 23, 2010, at 8:49 AM, Wellie Chao wrote:

> Yes, I see how display updates are useful on outbound calls now based on 
> your examples, but what about use cases for inbound calls? It doesn't seem 
> that useful to have the display update to the extension that is called 
> after the call is answered (everybody knows their own extension, after 
> all). Just wondering why that default behavior was selected. Is there a 
> way to leave display updates turned on for the outbound leg of the call 
> and to turn them off for the inbound leg of the call?
> 
> 
> 
> Date: Fri, 23 Apr 2010 08:14:52 -0400
> From: Steven Ward <steve.d.ward at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
> 
> To give one example, display updates are very useful in the application I'm using.
> 
> I have operators using speed dials for dialing out to dialplan that does a fifo out; so I see what the speed dial is when I
> press it (e.g. Operator_Call); but then when the operator is connected to the actual caller, the display updates to the CID of
> the caller she's talking to - very useful.
> 
> Also, people using a SIP phone to call into a fifo see the number they dialed (the number that gets them into the fifo).  But
> when they're answered by an actual phone, their display updates to the number of the endpoint they're talking to.
> 
> 
> 
> On Thu, Apr 22, 2010 at 10:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
>      As I mentioned in my last email, the ignore_display_updates variable did
>      the trick, but I am just curious: what is the purpose of the default
>      behavior where the display changes to the callee name and number? I am
>      guessing there must be some use case (in fact a fairly prevalent use case)
>      where showing the callee is more desirable than showing the caller, but
>      I'm not aware of what that might be and am curious to find out what the
>      use case(s) are. Changing the display to callee also seems to affect call
>      logs on the Polycom, which makes the received call log kind of useless.
> 
> 
>      Date: Thu, 22 Apr 2010 22:57:53 +0300
>      From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
>      Reply-To: freeswitch-users at lists.freeswitch.org
>      To: freeswitch-users at lists.freeswitch.org
>      Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
> 
>      Try adding the following line in your diaplan before the bridge command:
>       
>      <action application="set" data="ignore_display_updates=true"/>
>      (try true and false and see which one works better for you).
>       
>                                 __Yehavi:
>      2010/4/22 Wellie Chao <wchao at yahoo.com>
>           I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now
>           I see a strange behavior with caller ID on Polycom phones when handling
>           inbound calls.
> 
>           Here is the scenario:
> 
>           * Call is from 212-555-2222 (external number not on my softswitch)
> 
>           * Call is to 212-555-1001 (number on my softswitch, extension 1001)
> 
>           * extension 1001 is a Polycom phone (it's an IP301, but same problem
>             occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
>             SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.
> 
>           * on phone 1001, the caller ID will display 212-555-2222 while ringing.
>             The moment I pick up, the display will change to "From: 1001" (referring
>             to the extension of the phone itself).
> 
>           Has anyone else experienced this problem, and does anyone know how to fix
>           it? It does not occur with the snom 320 (and I assume it does not occur
>           with any of the snom models based on extrapolation).
> 
>           While the problem only started when I updated from 1.0.4 to 1.0.6, it's
>           entirely possible it's a configuration setting on the Polycom rather than
>           a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
>           resolve the problem (or even how to go about investigating the root
>           cause)?



From mike at jerris.com  Sat Apr 24 12:19:57 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:19:57 -0400
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <55FBE664-764D-4EA1-9B48-B8FB8949D85C@freeswitch.org>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
	<p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
	<55FBE664-764D-4EA1-9B48-B8FB8949D85C@freeswitch.org>
Message-ID: <F9931EC7-904C-41D0-8185-16C9AC075530@jerris.com>


On Apr 23, 2010, at 9:34 AM, Brian West wrote:

> 
> On Apr 23, 2010, at 12:30 AM, Gabriel Kuri wrote:
> 
>> Send proof the machine is dead? Would you like a picture of smoke pouring out of a box in flames? 
> 
> A signed paper stating that its failed is usually good enough.  Again right now I'm letting you get away with it three times.  If you have it happen 10 times a month for 3 months in a row thats a suspect.... but if you email me and its happened 12 months or more part then i'm more than willing to let it slide more than 3 times.  I just won't let it be abused.
> 
>> What if you want to re-allocate the machine to do something else or just replace it with a more powerful machine? How are the licenses tied to the machine, MAC address? 
> 
> See above.
> 
>> Any reason the "option" for a floating license model isn't available? It seems given the option between the two, I'd rather have the floating license model so I wouldn't need to prove my machine is dead before asking to have the licenses re-issued, unless of course the license server is dead. We swap production machines in/out all the time, particularly because we perform maintenance on one, so we have a spare, bring it up, take the other one down, perform maintenance, and bring it back up.
> 
> Floating license server isn't out there because you have more chances for things to go wrong and calls to be dropped due to issues related to reaching the server.  The decision was made to not do that.  Have the calls just work is more critical to our carrier customers.

To follow up on this.  This is something that I might want to do in the future, but there are a good number of non-trivial technical challenges to do this in a way that can be stable.  For example, you might want to reserve x licenses for each server, and just set a timeout if they don't check in to release the licenses so they can be assigned to another server.  All of this logic would require a lot more time, work and testing.  Our priority was to get a working stable and efficient codec out the door so people could use it, not to have every bell and whistle available.  The downsides of a floating license server outweigh the upsides at least for now.

>> With this model, floating licenses would be our only option, I really wouldn't want to be purchasing a bunch of extra licenses for spare machines.
>> 
>> I can't imagine you guys really like making us feel like criminals by tieing the license to an actual box, what's wrong with the good 'ole "on your honor policy"? I realize the need to pay for valid g729 licenses, but prove the machine is dead? Is this requirement coming from Sipro?
> 
> We have to make reasonable efforts, its a requirement.
> 

To re-iterate.  We are required by agreements to make reasonable efforts to assure compliance.  This is not something I like or want to do, this is a requirement placed on us.  You should be finding similar clauses in any other implementation out there.  Even a floating license implementation will still have a license server that is somehow locked to something, to keep you from running multiple copies of it.

Mike




From mike at jerris.com  Sat Apr 24 12:21:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:21:40 -0400
Subject: [Freeswitch-users] One Way Audio on Some calls
In-Reply-To: <413285.30918.qm@web30208.mail.mud.yahoo.com>
References: <413285.30918.qm@web30208.mail.mud.yahoo.com>
Message-ID: <AEEE2596-BD11-4403-A7B6-C0773CF5FDBB@jerris.com>

This ssems strange.  Are you able to capture a debug log with sip trace of just one of these broken calls that I can take a look at?

Mike

On Apr 23, 2010, at 1:30 AM, Travis Stevens wrote:

> Has any one seen an issue where an endpoint answers and it take literally 10 seconds before the server forwards the signaling to the next server upstream. It does not happen every call, but the ip that the server is communicating to is the same if the call is good or bad. If the call is bad after the 10 seconds the remote end of connected and there is one way audio.
> 
> The Only difference i have been able to see in the logs is that the call that completes correctly returns a lines stating that stun is not needed because the ports match. 
> 
> I am running FreeSWITCH Version 1.0.head (svn-17188M)
> 
> Freeswitch has public IP addresses and the Endpoint are behind nat. I have tested multiple endpoint with the same results.
> 
> Thanks in advance for any help.
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 24 12:23:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Apr 2010 15:23:34 -0400
Subject: [Freeswitch-users] Any breaking change on register with 1.0.6
In-Reply-To: <w2m367751821004231055h8b2ea90cxe0046f3c61ab59f3@mail.gmail.com>
References: <w2m367751821004231055h8b2ea90cxe0046f3c61ab59f3@mail.gmail.com>
Message-ID: <BFFF601E-68E0-4AEF-A78B-4ADC03C97F9D@jerris.com>

can you provide any logs of the failure?

On Apr 23, 2010, at 1:55 PM, Phillip Jones wrote:

> Hi there,
> 
> I have been using my Cisco phones registered with FreeSWITCH (1.0.4) for months now. Today I upgraded to 1.0.6 and they no longer registers. 
> 
> This what configured to get them to work.
> 
> 1) Add 
> 
>  <list name="lan" default="allow">
>         <node type="allow" cidr="71.**5.60.***/32"/>
> 
> to acl.conf.xml
> 
> 2) Add the appropriate directory entry. 
> 
> 
> That worked (may be it shouldn't have done) - but on 1.0.6 (today's trunk) is fails.
> 
> Is something else required in 1.0.6 do you know?
> 



From btsteve at yahoo.com  Sat Apr 24 12:28:40 2010
From: btsteve at yahoo.com (Travis Stevens)
Date: Sat, 24 Apr 2010 12:28:40 -0700 (PDT)
Subject: [Freeswitch-users] One Way Audio on Some calls
Message-ID: <409029.55781.qm@web30206.mail.mud.yahoo.com>

I solved the issue. There was still a default setting for trying to use stun on the external rtp and sip addresses. 

Sent from my iPhone

On Apr 24, 2010, at 3:21 PM, Michael Jerris <mike at jerris.com> wrote:

This ssems strange.  Are you able to capture a debug log with sip trace of just one of these broken calls that I can take a look at?

Mike

On Apr 23, 2010, at 1:30 AM, Travis Stevens wrote:

Has any one seen an issue where an endpoint answers and it take literally 10 seconds before the server forwards the signaling to the next server upstream. It does not happen every call, but the ip that the server is communicating to is the same if the call is good or bad. If the call is bad after the 10 seconds the remote end of connected and there is one way audio.

The Only difference i have been able to see in the logs is that the call that completes correctly returns a lines stating that stun is not needed because the ports match. 

I am running FreeSWITCH Version 1.0.head (svn-17188M)

Freeswitch has public IP addresses and the Endpoint are behind nat. I have tested multiple endpoint with the same results.

Thanks in advance for any help.




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


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



      


From vfclists at googlemail.com  Sat Apr 24 12:37:08 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 24 Apr 2010 20:37:08 +0100
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
Message-ID: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>

I have a simple Asterisk AGI based to translate to Freeswitch and want
to use mod_event_socket to implement.

The first part is whether the extension making the call is active. In
the AGI it checks the database whether the extension is supposed to be
on, and skips to the end of the dial plan if it is not.

The second part is to check the destination, select the gateway to
call, and dial through with an API command or whatever is best suited.

If I use an outbound event socket which events should I filter for?

>From reading the docs it looks once the call is parked and the event
shows up, I either hangup the call, or unpark it and bridge via a
gateway.

The wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
gives an example like this.

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808
event-lock: true

The api command list produced by api show does not contain a bridge
command, but has a uuid_bridge command. Is that the command to use?

Another wiiki example uses this: api originate
sofia/mydomain.com/ext at yourvsp.com 1000 , which appears to be another
method.

I just need some info about the options available and how to implement them.


--
Frank Church

=======================
http://devblog.brahmancreations.com


From d at d-man.org  Sat Apr 24 12:41:00 2010
From: d at d-man.org (Darren Schreiber)
Date: Sat, 24 Apr 2010 12:41:00 -0700
Subject: [Freeswitch-users] LAST REMINDER: FreeSWITCH Users Group & Install
 Fest is tomorrow in San Francisco!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F17DA22F84D@EXVMBX020-3.exch020.serverdata.net>

Hi everyone,
                A final reminder that we'll be doing a FreeSWITCH users group meet-up and install-fest tomorrow in San Francisco at Borders Books. It will be at 4pm. If you plan to attend please R.S.V.P. via the MeetUp website so I have an accurate headcount before the meeting.

                Look forward to seeing you Bay Area folks tomorrow!

http://www.meetup.com/fsusers/calendar/13009468/ for details, location & RSVP ability.

Sincerely,
                Darren Schreiber

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From msc at freeswitch.org  Sat Apr 24 13:44:10 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 24 Apr 2010 13:44:10 -0700
Subject: [Freeswitch-users] LAST REMINDER: FreeSWITCH Users Group &
	Install Fest is tomorrow in San Francisco!
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F17DA22F84D@EXVMBX020-3.exch020.serverdata.net>
References: <8A034A3098ED3C4990F7D9DE40F5585F17DA22F84D@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <76DC3A9C-321F-4FB8-8FBE-07365BACFEC0@freeswitch.org>

I'm so there! I've got a few phones we can tinker with. Be sure to  
bring some Ethernet ports! :)

-MC

Sent from my iPhone

On Apr 24, 2010, at 12:41 PM, Darren Schreiber <d at d-man.org> wrote:

> Hi everyone,
>
>                 A final reminder that we?ll be doing a FreeSWITCH us 
> ers group meet-up and install-fest tomorrow in San Francisco at Bord 
> ers Books. It will be at 4pm. If you plan to attend please R.S.V.P.  
> via the MeetUp website so I have an accurate headcount before the me 
> eting.
>
>
>
>                 Look forward to seeing you Bay Area folks tomorrow!
>
>
>
> http://www.meetup.com/fsusers/calendar/13009468/ for details,  
> location & RSVP ability.
>
>
>
> Sincerely,
>
>                 Darren Schreiber
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From dujinfang at gmail.com  Sat Apr 24 17:10:01 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 25 Apr 2010 08:10:01 +0800
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
Message-ID: <u2s23f91031004241710r737ed899p3e1fafbcc7abdf6b@mail.gmail.com>

All the uuid_xxx api are from mod_commands and execute *OUT* of a channel.

What you are looking for called app (v.s. api) which execute "IN" a
channel(think about, the b-leg)

http://wiki.freeswitch.org/wiki/Mod_dptools


2010/4/25 Frank Church <vfclists at googlemail.com>:
> I have a simple Asterisk AGI based to translate to Freeswitch and want
> to use mod_event_socket to implement.
>
> The first part is whether the extension making the call is active. In
> the AGI it checks the database whether the extension is supposed to be
> on, and skips to the end of the dial plan if it is not.
>
> The second part is to check the destination, select the gateway to
> call, and dial through with an API command or whatever is best suited.
>
> If I use an outbound event socket which events should I filter for?
>
> >From reading the docs it looks once the call is parked and the event
> shows up, I either hangup the call, or unpark it and bridge via a
> gateway.
>
> The wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
> gives an example like this.
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808
> event-lock: true
>
> The api command list produced by api show does not contain a bridge
> command, but has a uuid_bridge command. Is that the command to use?
>
> Another wiiki example uses this: api originate
> sofia/mydomain.com/ext at yourvsp.com 1000 , which appears to be another
> method.
>
> I just need some info about the options available and how to implement them.
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>



-- 
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn


From elihayun at gmail.com  Sat Apr 24 21:41:22 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Sun, 25 Apr 2010 07:41:22 +0300
Subject: [Freeswitch-users] where file "xmlrpc.inc" gone?
In-Reply-To: <891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
References: <4BCEA13C.5060707@gmail.com>
	<891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
Message-ID: <4BD3C7F2.2050808@savion.huji.ac.il>

On 04/24/2010 09:40 PM, Michael Jerris wrote:
> I just realized I don't think I ever rolled a contrib tarball for 1.0.6.  We split the contrib out of the base repo before the 1.0.6 release.  I will try to get this corrected this weekend and put a traball out on files.freeswitch.org.
>
> Mike
>
>
>
> On Apr 21, 2010, at 2:54 AM, Eli Hayun wrote:
>
>   
>> Hi
>>
>> I am trying to add some xmlprc functionality to FS. I want to use it
>> with PHP. The php example has the following line:
>>
>>
>> include("xmlrpc.inc");
>>
>> I noticed that in version 1.0.5-xx the file included but not with 1.0.6
>>
>> Is there other way to use it with php or the file is wrongly missing ?
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 now I used the file from the old tarball.

Eli


From elihayun at gmail.com  Sat Apr 24 21:41:22 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Sun, 25 Apr 2010 07:41:22 +0300
Subject: [Freeswitch-users] where file "xmlrpc.inc" gone?
In-Reply-To: <891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
References: <4BCEA13C.5060707@gmail.com>
	<891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
Message-ID: <4BD3C7F2.2050808@savion.huji.ac.il>

On 04/24/2010 09:40 PM, Michael Jerris wrote:
> I just realized I don't think I ever rolled a contrib tarball for 1.0.6.  We split the contrib out of the base repo before the 1.0.6 release.  I will try to get this corrected this weekend and put a traball out on files.freeswitch.org.
>
> Mike
>
>
>
> On Apr 21, 2010, at 2:54 AM, Eli Hayun wrote:
>
>   
>> Hi
>>
>> I am trying to add some xmlprc functionality to FS. I want to use it
>> with PHP. The php example has the following line:
>>
>>
>> include("xmlrpc.inc");
>>
>> I noticed that in version 1.0.5-xx the file included but not with 1.0.6
>>
>> Is there other way to use it with php or the file is wrongly missing ?
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 now I used the file from the old tarball.

Eli


From elihayun at gmail.com  Sat Apr 24 21:41:22 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Sun, 25 Apr 2010 07:41:22 +0300
Subject: [Freeswitch-users] where file "xmlrpc.inc" gone?
In-Reply-To: <891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
References: <4BCEA13C.5060707@gmail.com>
	<891BC13E-37F4-4BE0-93C7-3A265C0CB890@jerris.com>
Message-ID: <4BD3C7F2.2050808@savion.huji.ac.il>

On 04/24/2010 09:40 PM, Michael Jerris wrote:
> I just realized I don't think I ever rolled a contrib tarball for 1.0.6.  We split the contrib out of the base repo before the 1.0.6 release.  I will try to get this corrected this weekend and put a traball out on files.freeswitch.org.
>
> Mike
>
>
>
> On Apr 21, 2010, at 2:54 AM, Eli Hayun wrote:
>
>   
>> Hi
>>
>> I am trying to add some xmlprc functionality to FS. I want to use it
>> with PHP. The php example has the following line:
>>
>>
>> include("xmlrpc.inc");
>>
>> I noticed that in version 1.0.5-xx the file included but not with 1.0.6
>>
>> Is there other way to use it with php or the file is wrongly missing ?
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 now I used the file from the old tarball.

Eli


From msc at freeswitch.org  Sat Apr 24 23:02:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 24 Apr 2010 23:02:56 -0700
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
Message-ID: <r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com>

Throw a copy of that AGI script up on pastebin and let us take a look at it.
We should be able to assist you in finding the right event_socket stuff to
get you going.
-MC

On Sat, Apr 24, 2010 at 12:37 PM, Frank Church <vfclists at googlemail.com>wrote:

> I have a simple Asterisk AGI based to translate to Freeswitch and want
> to use mod_event_socket to implement.
>
> The first part is whether the extension making the call is active. In
> the AGI it checks the database whether the extension is supposed to be
> on, and skips to the end of the dial plan if it is not.
>
> The second part is to check the destination, select the gateway to
> call, and dial through with an API command or whatever is best suited.
>
> If I use an outbound event socket which events should I filter for?
>
> >From reading the docs it looks once the call is parked and the event
> shows up, I either hangup the call, or unpark it and bridge via a
> gateway.
>
> The wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
> gives an example like this.
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808
> event-lock: true
>
> The api command list produced by api show does not contain a bridge
> command, but has a uuid_bridge command. Is that the command to use?
>
> Another wiiki example uses this: api originate
> sofia/mydomain.com/ext at yourvsp.com 1000 , which appears to be another
> method.
>
> I just need some info about the options available and how to implement
> them.
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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 emilbergg at gmail.com  Sat Apr 24 23:50:54 2010
From: emilbergg at gmail.com (Emil Berg)
Date: Sun, 25 Apr 2010 09:50:54 +0300
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <4BD06F18.5050808@infosecurity.ch>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
	<4BCFEAD7.7050902@infosecurity.ch>
	<v2j191c3a031004220833r51ebad1am410665794c3ed1ac@mail.gmail.com>
	<4BD06F18.5050808@infosecurity.ch>
Message-ID: <q2u8817c1b51004242350r6cbee4c3u5a0735b402a57029@mail.gmail.com>

Hello everybody,

Sorry for not answering so long. There were holidays here and only now I got
back to office.
I'm a beginner and I just installed freeswitch and started playing with it
and see if it works with my client.
This is why I didn't provide any info. I just wanted to know if this is a
known issue.
If someone could tell me please how can I produce these logs, I will attach
them.

Thank you,
Emil.

On Thu, Apr 22, 2010 at 6:45 PM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> Hi Anthony,
> i really apologise for not being still able to provide the trace, i am
> under product release and this is an issue of the many to be managed and
> it's in my company backlog.
>
> I did not wanted to make FUD or being arrogant, really.
>
> It's just that i read an issue in the mailing list similar to what i
> experience.
> We still not analyzed deeply and you know that a non deterministic bug
> it's something quite difficult to be traced down, require a continuous
> testing environment to catch the issue while collecting a lot of logs :(
>
> Will do it, just is in a schedule of activities.
>
> Fabio
> On 22/04/10 17.33, Anthony Minessale wrote:
> > Are you arrogantly refusing to collect the actual sip traces but still
> > feel the need to report an issue?
> > That is called FUD and we do not tolerate it here.  Either hold your
> > tongues or collect the info.  Do not just use our mailing list to say
> > you have "random and not reproducible problems"  It does nobody any
> > good.  We have now wasted 3 email exchanges simply asking for the
> > information.  If you were able to produce the ladder diagram surely
> > you could have done what I asked and did a pcap from the FS box and a
> > console trace.  Maybe you should learn more about network topology
> > before you send us any more email.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vfclists at googlemail.com  Sun Apr 25 00:21:38 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 25 Apr 2010 08:21:38 +0100
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
	<r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com>
Message-ID: <p2y7abab2411004250021peb141351kac7aa310a2407ce@mail.gmail.com>

http://pastebin.freeswitch.org/12801

The AGI first tests if the management system is running checking the
uptime table, then it checks whether the extension making the call is
permitted, then allows the call to proceed into the dial plan.

This version is setup for a single outbound provider. I want to extend
the freeswitch version dial multiple providers in case of failure.

Thanks

Frank.

On 25 April 2010 07:02, Michael Collins <msc at freeswitch.org> wrote:
> Throw a copy of that AGI script up on pastebin and let us take a look at it.
> We should be able to assist you in finding the right event_socket stuff to
> get you going.
> -MC
>
> On Sat, Apr 24, 2010 at 12:37 PM, Frank Church <vfclists at googlemail.com>
> wrote:
>>
>> I have a simple Asterisk AGI based to translate to Freeswitch and want
>> to use mod_event_socket to implement.
>>
>> The first part is whether the extension making the call is active. In
>> the AGI it checks the database whether the extension is supposed to be
>> on, and skips to the end of the dial plan if it is not.
>>
>> The second part is to check the destination, select the gateway to
>> call, and dial through with an API command or whatever is best suited.
>>
>> If I use an outbound event socket which events should I filter for?
>>
>> >From reading the docs it looks once the call is parked and the event
>> shows up, I either hangup the call, or unpark it and bridge via a
>> gateway.
>>
>> The wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
>> gives an example like this.
>>
>> sendmsg
>> call-command: execute
>> execute-app-name: bridge
>> execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808
>> event-lock: true
>>
>> The api command list produced by api show does not contain a bridge
>> command, but has a uuid_bridge command. Is that the command to use?
>>
>> Another wiiki example uses this: api originate
>> sofia/mydomain.com/ext at yourvsp.com 1000 , which appears to be another
>> method.
>>
>> I just need some info about the options available and how to implement
>> them.
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From william.suffill at gmail.com  Sun Apr 25 00:39:02 2010
From: william.suffill at gmail.com (William Suffill)
Date: Sun, 25 Apr 2010 03:39:02 -0400
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <p2y7abab2411004250021peb141351kac7aa310a2407ce@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com> 
	<r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com> 
	<p2y7abab2411004250021peb141351kac7aa310a2407ce@mail.gmail.com>
Message-ID: <g2h6b65470d1004250039h72d181e0z4c41dba324f168d@mail.gmail.com>

Looks pretty straight forward. The biggest hurtle is design. From the posted
code it's invoked on a per call basis. Quicker to write but not ideal if you
want to scale it up. You could do similar in FS too by embed scripting or a
daemon that catches any call to a certain extension regex and applies your
logic.

outbound event socket would work too if you wanted to stick with PHP and
reuse most of what you have.
http://wiki.freeswitch.org/wiki/PHP_ESL (I wrote that page so any
questions/suggestions are welcome.)

-- W

PS: sorry for any typos but it's getting a bit late (3:38am)
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From vfclists at googlemail.com  Sun Apr 25 01:43:37 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 25 Apr 2010 09:43:37 +0100
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <g2h6b65470d1004250039h72d181e0z4c41dba324f168d@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
	<r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com>
	<p2y7abab2411004250021peb141351kac7aa310a2407ce@mail.gmail.com>
	<g2h6b65470d1004250039h72d181e0z4c41dba324f168d@mail.gmail.com>
Message-ID: <n2o7abab2411004250143md27ec6c9xc02a21c45be4f05d@mail.gmail.com>

On 25 April 2010 08:39, William Suffill <william.suffill at gmail.com> wrote:
> Looks pretty straight forward. The biggest hurtle is design. From the posted
> code it's invoked on a per call basis. Quicker to write but not ideal if you
> want to scale it up. You could do similar in FS too by embed scripting or a
> daemon that catches any call to a certain extension regex and applies your
> logic.

After some reading outbound sockets seem to be the way to go.

Can an outbound socket return to the dial plan after setting some
variables during its execution, so that the dialplan can work as
normal? It is what I prefer.

The call volumes are not that high and the controlling program is
written in ObjectPascal, making performance even less of a problem

The original deployment run on an Asterisk VM but switching to
Freeswitch means having the VoIP engine a Windows one as well.

>
> outbound event socket would work too if you wanted to stick with PHP and
> reuse most of what you have.
> http://wiki.freeswitch.org/wiki/PHP_ESL (I wrote that page so any
> questions/suggestions are welcome.)
>
> -- W
>
> PS: sorry for any typos but it's getting a bit late (3:38am)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From dujinfang at gmail.com  Sun Apr 25 03:52:39 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 25 Apr 2010 18:52:39 +0800
Subject: [Freeswitch-users] Controlling calls with mod_event_socket
In-Reply-To: <n2o7abab2411004250143md27ec6c9xc02a21c45be4f05d@mail.gmail.com>
References: <p2m7abab2411004241237tf4991ad8if4fe09cd4f4bde54@mail.gmail.com>
	<r2j87f2f3b91004242302t93421a2bs5d22fa45a76d3a4@mail.gmail.com>
	<p2y7abab2411004250021peb141351kac7aa310a2407ce@mail.gmail.com>
	<g2h6b65470d1004250039h72d181e0z4c41dba324f168d@mail.gmail.com>
	<n2o7abab2411004250143md27ec6c9xc02a21c45be4f05d@mail.gmail.com>
Message-ID: <t2r23f91031004250352zb707f9f7y3564f091aa0c6fb@mail.gmail.com>

2010/4/25 Frank Church <vfclists at googlemail.com>:
> On 25 April 2010 08:39, William Suffill <william.suffill at gmail.com> wrote:
>> Looks pretty straight forward. The biggest hurtle is design. From the posted
>> code it's invoked on a per call basis. Quicker to write but not ideal if you
>> want to scale it up. You could do similar in FS too by embed scripting or a
>> daemon that catches any call to a certain extension regex and applies your
>> logic.
>
> After some reading outbound sockets seem to be the way to go.
>
> Can an outbound socket return to the dial plan after setting some
> variables during its execution, so that the dialplan can work as
> normal? It is what I prefer.
>

Yes, you can transfer to a dialplan extension

> The call volumes are not that high and the controlling program is
> written in ObjectPascal, making performance even less of a problem
>
> The original deployment run on an Asterisk VM but switching to
> Freeswitch means having the VoIP engine a Windows one as well.
>
>>
>> outbound event socket would work too if you wanted to stick with PHP and
>> reuse most of what you have.
>> http://wiki.freeswitch.org/wiki/PHP_ESL (I wrote that page so any
>> questions/suggestions are welcome.)
>>
>> -- W
>>
>> PS: sorry for any typos but it's getting a bit late (3:38am)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>



-- 
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn


From sashker at sashker.ru  Sun Apr 25 10:37:19 2010
From: sashker at sashker.ru (=?KOI8-R?B?4czFy9PBzsTSIPzE1cHSxM/Xyd4=?=)
Date: Sun, 25 Apr 2010 23:37:19 +0600
Subject: [Freeswitch-users] Problem with multiple concurrent incoming calls
	to the same Skype username and another names (transfering)
Message-ID: <z2naa322b471004251037i1395e1ffvab5e5bad182b62f8@mail.gmail.com>

Hi!

I've a FS 1.0.6  rev svn-17188M. And with multiple concurrent calls I've
some crashes, but 1 call working fine :)).

http://pastebin.freeswitch.org/12802 - this report about my problem

Best regards,
Alexander aka Sashker
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From mike at jerris.com  Sun Apr 25 12:45:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 25 Apr 2010 15:45:53 -0400
Subject: [Freeswitch-users] Problem with multiple concurrent incoming
	calls to the same Skype username and another names (transfering)
In-Reply-To: <z2naa322b471004251037i1395e1ffvab5e5bad182b62f8@mail.gmail.com>
References: <z2naa322b471004251037i1395e1ffvab5e5bad182b62f8@mail.gmail.com>
Message-ID: <9F9DB223-72C1-4135-9749-130CDA84A847@jerris.com>

Bug reports should go to http://jira.freeswitch.org

Mike

On Apr 25, 2010, at 1:37 PM, ????????? ?????????? wrote:

> Hi!
> 
> I've a FS 1.0.6  rev svn-17188M. And with multiple concurrent calls I've some crashes, but 1 call working fine :)).
> 
> http://pastebin.freeswitch.org/12802 - this report about my problem

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From wchao at yahoo.com  Sun Apr 25 19:01:45 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Sun, 25 Apr 2010 22:01:45 -0400 (EDT)
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <5A73BA42-86F8-4936-9355-81885F6425F4@jerris.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
	<alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
	<g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>
	<alpine.LRH.2.00.1004230845540.21297@osgood.microoffice.com>
	<5A73BA42-86F8-4936-9355-81885F6425F4@jerris.com>
Message-ID: <alpine.LRH.2.00.1004252200540.21297@osgood.microoffice.com>

Yep, still happens in the git head. This is the version reported by 
FreeSWITCH:

FreeSWITCH Version 1.0.head (svn-17188M)

I'll open a bug on jira as you request.



Date: Sat, 24 Apr 2010 15:07:58 -0400
From: Michael Jerris <mike at jerris.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom

Can you confirm this is still the case in current git head and if so open me a bug on jira about this issue please.

Mike

On Apr 23, 2010, at 8:49 AM, Wellie Chao wrote:

> Yes, I see how display updates are useful on outbound calls now based on 
> your examples, but what about use cases for inbound calls? It doesn't seem 
> that useful to have the display update to the extension that is called 
> after the call is answered (everybody knows their own extension, after 
> all). Just wondering why that default behavior was selected. Is there a 
> way to leave display updates turned on for the outbound leg of the call 
> and to turn them off for the inbound leg of the call?
> 
> 
> 
> Date: Fri, 23 Apr 2010 08:14:52 -0400
> From: Steven Ward <steve.d.ward at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
> 
> To give one example, display updates are very useful in the application I'm using.
> 
> I have operators using speed dials for dialing out to dialplan that does a fifo out; so I see what the speed dial is when I
> press it (e.g. Operator_Call); but then when the operator is connected to the actual caller, the display updates to the CID of
> the caller she's talking to - very useful.
> 
> Also, people using a SIP phone to call into a fifo see the number they dialed (the number that gets them into the fifo).  But
> when they're answered by an actual phone, their display updates to the number of the endpoint they're talking to.
> 
> 
> 
> On Thu, Apr 22, 2010 at 10:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
>      As I mentioned in my last email, the ignore_display_updates variable did
>      the trick, but I am just curious: what is the purpose of the default
>      behavior where the display changes to the callee name and number? I am
>      guessing there must be some use case (in fact a fairly prevalent use case)
>      where showing the callee is more desirable than showing the caller, but
>      I'm not aware of what that might be and am curious to find out what the
>      use case(s) are. Changing the display to callee also seems to affect call
>      logs on the Polycom, which makes the received call log kind of useless.
> 
> 
>      Date: Thu, 22 Apr 2010 22:57:53 +0300
>      From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
>      Reply-To: freeswitch-users at lists.freeswitch.org
>      To: freeswitch-users at lists.freeswitch.org
>      Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
> 
>      Try adding the following line in your diaplan before the bridge command:
>       
>      <action application="set" data="ignore_display_updates=true"/>
>      (try true and false and see which one works better for you).
>       
>                                 __Yehavi:
>      2010/4/22 Wellie Chao <wchao at yahoo.com>
>           I recently updated my FreeSWITCH installation from 1.0.4 to 1.0.6 and now
>           I see a strange behavior with caller ID on Polycom phones when handling
>           inbound calls.
> 
>           Here is the scenario:
> 
>           * Call is from 212-555-2222 (external number not on my softswitch)
> 
>           * Call is to 212-555-1001 (number on my softswitch, extension 1001)
> 
>           * extension 1001 is a Polycom phone (it's an IP301, but same problem
>             occurs on other Polycom models such as the 501 and 601). SIP 3.1.4 and
>             SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the IP301/501/601.
> 
>           * on phone 1001, the caller ID will display 212-555-2222 while ringing.
>             The moment I pick up, the display will change to "From: 1001" (referring
>             to the extension of the phone itself).
> 
>           Has anyone else experienced this problem, and does anyone know how to fix
>           it? It does not occur with the snom 320 (and I assume it does not occur
>           with any of the snom models based on extrapolation).
> 
>           While the problem only started when I updated from 1.0.4 to 1.0.6, it's
>           entirely possible it's a configuration setting on the Polycom rather than
>           a FreeSWITCH issue. I'm not sure. Anyone have pointers about how to
>           resolve the problem (or even how to go about investigating the root
>           cause)?


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


From cucku.cucku at yahoo.com.vn  Sun Apr 25 19:44:31 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Mon, 26 Apr 2010 10:44:31 +0800 (SGT)
Subject: [Freeswitch-users]  need help on IVR
In-Reply-To: <7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
Message-ID: <911182.21146.qm@web76210.mail.sg1.yahoo.com>

Hi Michael Jerris

my sip server is cisco sip server
could you guide me some clue or show the same config for the case

Thank you
Ha`


________________________________
T?: Michael Jerris <mike at jerris.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] need help on IVR

what is a sip server?  Yes, you can do pretty much anything like this.


On Apr 19, 2010, at 12:08 AM, false wrote:

Hi all
>
>
>my network topology:
>
>
>endpoint 1(100)-----sip server ---IVR(Freeswitch)
>|
>|
>endpoint2(101)
>
>
>endpoint1 + endpoint2 are registered to sip server
>Freeswitch is regsitered to sip server with 103
>
>
>my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
>and RTP from endpoint 1 --> media proxy---> FS
>then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
>and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
>
>
>can FS do it
>
>
>Thank you
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>__________________________________________________
>B?n C? S? D?ng Yahoo! Kh?ng?
>M?t m?i v? th? r?c? Yahoo! Th? c? ch??ng tr?nh b?o v? ch?ng th? r?c h?u hi?u nh?t tr?n m?ng 
>http://vn.mail.yahoo.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 mike at jerris.com  Sun Apr 25 21:23:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 26 Apr 2010 00:23:25 -0400
Subject: [Freeswitch-users] need help on IVR
In-Reply-To: <911182.21146.qm@web76210.mail.sg1.yahoo.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>
Message-ID: <1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>

I meant this literally.  People always use the term "sip server" and I never understand what they are trying to say.  What exactly is a sip server?  I think you have your terms all completely confused in your explanation below or I am just being really dense.  This cisco server is a sip server and a media proxy?  Can you try to be very clear on what exactly these things are and what you are trying to do?

Mike

On Apr 25, 2010, at 10:44 PM, false wrote:

> Hi Michael Jerris
> 
> my sip server is cisco sip server
> could you guide me some clue or show the same config for the case
> 
> Thank you
> Ha`
> T?: Michael Jerris <mike at jerris.com>
> ??n: freeswitch-users at lists.freeswitch.org
> G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
> Ch? ??: Re: [Freeswitch-users] need help on IVR
> 
> what is a sip server?  Yes, you can do pretty much anything like this.
> 
> On Apr 19, 2010, at 12:08 AM, false wrote:
> 
>> Hi all
>> 
>> my network topology:
>> 
>> endpoint 1(100)-----sip server ---IVR(Freeswitch)
>> 					|
>> 					|	
>> 				endpoint2(101)
>> 
>> endpoint1 + endpoint2 are registered to sip server
>> Freeswitch is regsitered to sip server with 103
>> 
>> my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
>> and RTP from endpoint 1 --> media proxy---> FS
>> then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
>> and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
>> 
>> can FS do it

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From patrick at speechpro.com  Sun Apr 25 23:16:04 2010
From: patrick at speechpro.com (patrick)
Date: Mon, 26 Apr 2010 10:16:04 +0400
Subject: [Freeswitch-users] Where i can get help with freeswitch?
In-Reply-To: <k2n3687b0701004231159ib220cea6laa0f69708bd9ff10@mail.gmail.com>
References: <4BD16B52.9060800@speechpro.com>
	<k2n3687b0701004231159ib220cea6laa0f69708bd9ff10@mail.gmail.com>
Message-ID: <4BD52FA4.5080101@speechpro.com>


I use dialplan and this extensions:

<extension name="my_ivr_01">
<condition field="destination_number" expression="^777$">
<action application="answer"/>

<action application="set" data="tts_engine=unimrcp:vn-mrcp-v1"/>
<action application="set" data="tts_voice=Anna8000"/>
<action application="speak" data="Here is some text for synthesis"/>

<action application="detect_speech" data="unimrcp 
http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/>
</condition>
</extension>


<extension name="my_ivr_02">
<condition field="destination_number" expression="^778$">
<action application="answer"/>

<action application="set" data="playback_terminators=#"/>
<action application="playback" data="ponce-preludio-in-e-major.wav"/>

<action application="detect_speech" data="unimrcp 
http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/>
</condition>
</extension>



And I need to barge-in and start "detect_speech" in both extensions, 
when synthesis or playback is going on.

P.S. Thank you for answer!


Christopher Rienzo ?????:
> Barge-in will work out of the box for digits... to make it work for 
> ASR is a bit more complicated.
>
> I don't know what method you are using to do TTS, but it this is the 
> general idea:
>
> 1. set up a handler to deal with input callbacks
> 2. on DTMF or start of speech, return "break" to cause barge-in.
>
> My help can be more specific if you tell me more about what method 
> (dialplan, Lua, javascript, etc) you are using to execute TTS and ASR.
>
> Can't help you on the noise issue... someone else needs to chime in.
>
>
>
> On Fri, Apr 23, 2010 at 5:41 AM, patrick <patrick at speechpro.com 
> <mailto:patrick at speechpro.com>> wrote:
>
>     Hello from St.Petersburg!
>     My name is Patrick.
>     I try to realise IVR with ASR & TTS.
>     Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local
>     product "Voicenavigator" (mrcp server, ASR, TTS).
>
>     I have 2 problems:
>
>     1. How to realise "barge in" for playback and TTS? Does freeswitch
>     allow
>     that?
>     I need to start playback, or TTS and break it when some speech is
>     detected (by ASR)...
>
>     2. When I call from Asterisk to Freeswitch and bridge my call back to
>     Asterisk (to another number), there only noise in first asterisk
>     abonent's phone...
>
>     I would be grateful for any help ;-)
>
>     --
>     ? ?????????,
>
>     ?????????? ?????? ??????????
>     ??????? ?? ????????????
>     ??? ?????? ??????? ???????????
>     ???.: (812) 325-8848, ???. 6225
>     ????: (812) 327-9297
>     E-mail: patrick at speechpro.com <mailto:patrick at speechpro.com>
>     http://www.speechpro.ru
>
>
>     _______________________________________________
>     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
>   

-- 
? ?????????,

?????????? ?????? ??????????
??????? ?? ????????????
??? ?????? ??????? ???????????
???.: (812) 325-8848, ???. 6225
????: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru



From steveayre at gmail.com  Mon Apr 26 01:54:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Apr 2010 09:54:59 +0100
Subject: [Freeswitch-users] Sip over TCP issues?
In-Reply-To: <q2u8817c1b51004242350r6cbee4c3u5a0735b402a57029@mail.gmail.com>
References: <n2h8817c1b51004211205s55dab612u846a05d904972ed@mail.gmail.com>
	<4BCFEAD7.7050902@infosecurity.ch>
	<v2j191c3a031004220833r51ebad1am410665794c3ed1ac@mail.gmail.com>
	<4BD06F18.5050808@infosecurity.ch>
	<q2u8817c1b51004242350r6cbee4c3u5a0735b402a57029@mail.gmail.com>
Message-ID: <v2oeafa772f1004260154l58daf9det72b13194f4f86ff1@mail.gmail.com>

Hi Emil,

There's a page on the wiki which describes how to report bugs and how
to collect the relevant information.
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-Steve


On 25 April 2010 07:50, Emil Berg <emilbergg at gmail.com> wrote:
> Hello everybody,
>
> Sorry for not answering so long. There were holidays here and only now I got
> back to office.
> I'm a beginner and I just installed freeswitch and started playing with it
> and see if it works with my client.
> This is why I didn't provide any info. I just wanted to know if this is a
> known issue.
> If someone could tell me please how can I produce these logs, I will attach
> them.
>
> Thank you,
> Emil.
>
> On Thu, Apr 22, 2010 at 6:45 PM, Fabio Pietrosanti (naif)
> <lists at infosecurity.ch> wrote:
>>
>> Hi Anthony,
>> i really apologise for not being still able to provide the trace, i am
>> under product release and this is an issue of the many to be managed and
>> it's in my company backlog.
>>
>> I did not wanted to make FUD or being arrogant, really.
>>
>> It's just that i read an issue in the mailing list similar to what i
>> experience.
>> We still not analyzed deeply and you know that a non deterministic bug
>> it's something quite difficult to be traced down, require a continuous
>> testing environment to catch the issue while collecting a lot of logs :(
>>
>> Will do it, just is in a schedule of activities.
>>
>> Fabio
>> On 22/04/10 17.33, Anthony Minessale wrote:
>> > Are you arrogantly refusing to collect the actual sip traces but still
>> > feel the need to report an issue?
>> > That is called FUD and we do not tolerate it here. ?Either hold your
>> > tongues or collect the info. ?Do not just use our mailing list to say
>> > you have "random and not reproducible problems" ?It does nobody any
>> > good. ?We have now wasted 3 email exchanges simply asking for the
>> > information. ?If you were able to produce the ladder diagram surely
>> > you could have done what I asked and did a pcap from the FS box and a
>> > console trace. ?Maybe you should learn more about network topology
>> > before you send us any more email.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From angels_ldc at 163.com  Mon Apr 26 02:20:04 2010
From: angels_ldc at 163.com (angels_ldc)
Date: Mon, 26 Apr 2010 17:20:04 +0800
Subject: [Freeswitch-users] how i get configskypenew.tgz
Message-ID: <201004261720018596253@163.com>



I want install  Skypopen 
cp /mnt/root/configskypenew.tgz ./
tar xzf configskypenew.tgz
chown root.root .Skype
But where are configskypenew.tgz ??
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From gmaruzz at celliax.org  Mon Apr 26 03:58:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 26 Apr 2010 12:58:00 +0200
Subject: [Freeswitch-users] how i get configskypenew.tgz
In-Reply-To: <201004261720018596253@163.com>
References: <201004261720018596253@163.com>
Message-ID: <j2z7b197bef1004260358w2d731cf1h256ab531254f1852@mail.gmail.com>

On Mon, Apr 26, 2010 at 11:20 AM, angels_ldc <angels_ldc at 163.com> wrote:
>
>
> I want?install ?Skypopen
>
> cp /mnt/root/configskypenew.tgz ./
> tar xzf configskypenew.tgz
> chown root.root .Skype
> But where are configskypenew.tgz ??

why don't you read the line that precede the ones you copied from the wiki page?
So difficult to read?
It reads:

on "How to prepare the configuration directory of Skype clients on
Linux", see http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth


Also, have a read of *all* the page, 10min reading will spare you lot
of time after.

-giovanni


>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rdenert at tng.de  Mon Apr 26 04:06:44 2010
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 26 Apr 2010 13:06:44 +0200 (CEST)
Subject: [Freeswitch-users] Conferene under Xen
In-Reply-To: <33247780.115362.1272033935443.JavaMail.root@zimbra.tng.de>
Message-ID: <7756100.116871.1272280004606.JavaMail.root@zimbra.tng.de>

Hallo everybody,

I have one question. Who does freeswitch generate the nessessary clock for real-time applications like conferencing, moh, ...

The asterisk needs ztdummy. But there is a problem with running astersik on it. FS don?t have this problem on xen.

Best regards


From patrick at speechpro.com  Mon Apr 26 04:19:47 2010
From: patrick at speechpro.com (patrick)
Date: Mon, 26 Apr 2010 15:19:47 +0400
Subject: [Freeswitch-users] How to handle and parse "detect speech" answer?
Message-ID: <4BD576D3.50905@speechpro.com>


  detect speech


described here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech

I have some recog result in freeswitch console:

.................

Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Content-Length: 165

<?xml version="1.0"?>
<result grammar="holdr">
  <interpretation grammar="yesno" confidence="98">
    <input mode="speech">YES</input>
  </interpretation>
</result>

....................

They say: The XML body at the end there with our result has a 
Content-Length of 165. That is included as part of the overall count of 
1791 so don't get tripped up parsing that.


1. How can I get this answer from console to any variable in my dialplan.?
2. How and when the best way to parse it?




-- 
? ?????????,

?????????? ?????? ??????????
??????? ?? ????????????
??? ?????? ??????? ???????????
???.: (812) 325-8848, ???. 6225
????: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru



From david.ponzone at gmail.com  Mon Apr 26 04:24:49 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 26 Apr 2010 13:24:49 +0200
Subject: [Freeswitch-users] Conferene under Xen
In-Reply-To: <7756100.116871.1272280004606.JavaMail.root@zimbra.tng.de>
References: <7756100.116871.1272280004606.JavaMail.root@zimbra.tng.de>
Message-ID: <0F81B7C9-5C92-4A8F-BDC2-1877F5B75065@gmail.com>

FS very smart.
FS has its own internal super-accurate clock.

Be aware that despite its great quality, this internal clock may have  
issues running in a VM guest, one day or another.
It is not a validated architecture, and probably won't be until  
someone releases a VM product with real-time capabilities.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 26/04/2010 ? 13:06, Rudolf Denert a ?crit :

> Hallo everybody,
>
> I have one question. Who does freeswitch generate the nessessary  
> clock for real-time applications like conferencing, moh, ...
>
> The asterisk needs ztdummy. But there is a problem with running  
> astersik on it. FS don?t have this problem on xen.
>
> 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 hungngm at bkav.com.vn  Mon Apr 26 04:37:29 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Mon, 26 Apr 2010 18:37:29 +0700
Subject: [Freeswitch-users] some questions about cluster FS by using ODBC in
	the core.
Message-ID: <329861D8B5881E6C7DC0B78F46F163AA78962224@hungngm>



Hi list.

I try to deploy cluster FS by using "ODBC in the core". But i have some problem
when deploy:

1. I deploy 2 FS share registration ( by config odbc-dsn in internal profile).
With default config:

User 1001 registers in fs1: 10.2.48.194, user 1003 registers in fs2:
10.2.48.243.

I show sofia status profile internal and see to registration:

Call-ID: d3298837aa18ac2bOGYyNTYyZGJjNzcyNzNhNDA3MTI0MmY4MmMzOWFkZDU.
User: 1001 at 10.2.48.194
Contact: "1001" <sip:1001 at 10.2.48.184:42064;rinstance=22b332e0ef7c7fc1>
Agent: eyeBeam release 1003s stamp 31159
Status: Registered(UDP)(unknown) EXP(2010-04-16 11:26:47)
Host: FS1.voip.vn
IP: 10.2.48.184
Port: 42064
Auth-User: 1001
Auth-Realm: fs.vn
MWI-Account: 1001 at 10.2.48.194

Call-ID: NzFiZmQxM2FlZDdmZjU3NTk4NjY4NmIwOWQ4NTcwZDI.
User: 1003 at 10.2.48.243
Contact: "100311" <sip:1003 at 10.2.48.184:19374;rinstance=a7557f050b39a086>
Agent: X-Lite release 1104o stamp 56125
Status: Registered(UDP)(unknown) EXP(2010-04-27 12:18:37)
Host: localhost.localdomain
IP: 10.2.48.184
Port: 19374
Auth-User: 1003
Auth-Realm: fs.vn
MWI-Account: 1003 at 10.2.48.243

user 1001 can't dial user 1003 and vice-versa. It likes because user 1001 calls
1003 at 10.2.48.194 while user 1003 is registers 10.2.48.243.

I read in list that can call between FS with odbc in the core. Do i miss config
something ?

2. I have read some comments with 2 FS, we need to config DNS-SRV for load
balance between them. But I wonder how gateway connects with fs by load-balance
SIP trunk?

Tks for help.

 

 

Nguy??n M???nh H??g

Team Member - Telecom Security
Bkis Telecom Dept.

Bkis Internet Security

 

Office: HH1 Building - Yen Hoa, Cau Giay, Hanoi
Tel: (84 4) 37677090 ext 101
Mobile: (84) 1656 722 375
Email: HungNgM at bkav.com.vn
Website: www.bkav.com.vn

____________________________________________

Do your best, the rest will come !

H??? l?? vi???c h???t m???h, nh???ng ??i???u t???t
?????p s??? ?????n v???i b???n !

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From red.rain.seven at gmail.com  Mon Apr 26 05:31:40 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Mon, 26 Apr 2010 20:31:40 +0800
Subject: [Freeswitch-users] some questions about cluster FS by using
	ODBC in the core.
In-Reply-To: <329861D8B5881E6C7DC0B78F46F163AA78962224@hungngm>
References: <329861D8B5881E6C7DC0B78F46F163AA78962224@hungngm>
Message-ID: <g2o59ad9ca11004260531q95785278z5976d4dae1d93578@mail.gmail.com>

Even though the 2 FS server shares the same db. If you look closely, the
record still knows which user is registered to which FS server, thus you
can't just try to dial a user that is registered on another server. The
proper way of doing this is to find out if the user you are dialing is
registered to the same server, if not, then you will have to bridge
sofia/internal/1003 at 10.2.48.243  instead of just bridging user/1003


Henry Huang



2010/4/26 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi list.
>
> I try to deploy cluster FS by using  "ODBC in the core".  But i have some
> problem when deploy:
>
> 1. I deploy 2 FS share registration ( by config odbc-dsn in internal
> profile). With default config:
> User 1001 registers in fs1: 10.2.48.194, user 1003 registers in fs2:
> 10.2.48.243.
> I show sofia status profile internal and see to registration:
>
> Call-ID:
> d3298837aa18ac2bOGYyNTYyZGJjNzcyNzNhNDA3MTI0MmY4MmMzOWFkZDU.
> User:           1001 at 10.2.48.194
> Contact:        "1001" <sip:1001 at 10.2.48.184:42064
> ;rinstance=22b332e0ef7c7fc1>
> Agent:          eyeBeam release 1003s stamp 31159
> Status:         Registered(UDP)(unknown) EXP(2010-04-16 11:26:47)
> Host:           FS1.voip.vn
> IP:             10.2.48.184
> Port:           42064
> Auth-User:      1001
> Auth-Realm:     fs.vn
> MWI-Account:    1001 at 10.2.48.194
>
> Call-ID:        NzFiZmQxM2FlZDdmZjU3NTk4NjY4NmIwOWQ4NTcwZDI.
> User:           1003 at 10.2.48.243
> Contact:        "100311" <sip:1003 at 10.2.48.184:19374
> ;rinstance=a7557f050b39a086>
> Agent:          X-Lite release 1104o stamp 56125
> Status:         Registered(UDP)(unknown) EXP(2010-04-27 12:18:37)
> Host:           localhost.localdomain
> IP:             10.2.48.184
> Port:           19374
> Auth-User:      1003
> Auth-Realm:     fs.vn
> MWI-Account:    1003 at 10.2.48.243
>  <1001 at fs.vn>
>  user 1001 can't dial user 1003 and vice-versa. It likes because user 1001
> calls 1003 at 10.2.48.194 while user 1003 is registers 10.2.48.243.
> I read in list that can call between FS with odbc in the core. Do i miss
> config something ?
>
> 2. I have read some comments with 2 FS, we need to config DNS-SRV for load
> balance between them. But I wonder how gateway connects with fs by
> load-balance SIP trunk?
>
> Tks for help.
>
>
>
> Nguy?n M?nh H?g
>
> Team Member - Telecom Security
> Bkis Telecom Dept.
>
> Bkis Internet Security
>   Office:  HH1 Building - Yen Hoa, Cau Giay, Hanoi
>  Tel:      (84 4) 37677090 ext 101
>  Mobile: (84) 1656 722 375
>  Email:  HungNgM at bkav.com.vn
> Website: www.bkav.com.vn
>  ____________________________________________
>  Do your best, the rest will come !
> H? l? vi?c h?t m?h, nh?ng ?i?u t?t ??p s? ??n v?i b?n !
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 26 05:45:30 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Mon, 26 Apr 2010 08:45:30 -0400
Subject: [Freeswitch-users] session.steamFile misses DTMF event for first
	few seconds
Message-ID: <3F7821C562CD4EC6A7C5C342BB7077A7@ws4>

I recently upgraded from FS 12790M to svn 17188.  When I did, I noticed
that session.streamFile behaved differently and I started having
problems with my IVR app.  
 
With the upgraded FS, I have a problem with streamFile no firing on the
DTMF and calling the callback function for the first few seconds of the
wav file playback.  It behaves as though it does not hear the DTMFs.  If
I wait for 2 seconds or so of the wav file and then DTMF, streamFile
catches the DTMF and all is well.  If I key as soon as I hear the wav
file start, streamFile just keeps playing the wav and does not call the
callback function.
 
When I revert back to the previous version of FS, streamFile always
fires the callback right away no matter how quickly I press the first
DTMF as the wav file starts to stream out.
 
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
16 bit, mono 8000 Hz
 
The snippet of js code I am using is as follows.
 
if(session.ready()) {
    session.answer();
    session.sleep(750);
    while(session.ready()) {
        session.sleep(500);
        session.flushDigits(); // clear out input buffers

 
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) {
            pin=session.getDigits(pinmax,pinterm,pinwait);
        } else {
            pin+=session.getDigits(pinmax-1,pinterm,pinwait);
        }
        // more code here..
    }
 
Do I need to change the way I use streamFile in the later release?
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From rupa at rupa.com  Mon Apr 26 06:39:11 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 26 Apr 2010 08:39:11 -0500
Subject: [Freeswitch-users] Caller ID on inbound calls on Polycom
In-Reply-To: <alpine.LRH.2.00.1004252200540.21297@osgood.microoffice.com>
References: <alpine.LRH.2.00.1004221217420.21297@osgood.microoffice.com>
	<w2od361ca7b1004221257tc778c2ccobbdc3f5a6e5022ac@mail.gmail.com>
	<alpine.LRH.2.00.1004222227590.21297@osgood.microoffice.com>
	<g2x4ea6e8f21004230514t698db469t29703aa5a8241a40@mail.gmail.com>
	<alpine.LRH.2.00.1004230845540.21297@osgood.microoffice.com>
	<5A73BA42-86F8-4936-9355-81885F6425F4@jerris.com>
	<alpine.LRH.2.00.1004252200540.21297@osgood.microoffice.com>
Message-ID: <p2vb39273531004260639n47d38a87lf50fb9c11ff96f6e@mail.gmail.com>

I don't experience this behavior (internal # showing up as callee number for
incoming calls).  Perhaps it is an artifact of how you are doing your
dialplan?  When you open the ticket, please ensure that a copy of
the relevant portions of your dialplan are included.

On Sun, Apr 25, 2010 at 9:01 PM, Wellie Chao <wchao at yahoo.com> wrote:

> Yep, still happens in the git head. This is the version reported by
> FreeSWITCH:
>
> FreeSWITCH Version 1.0.head (svn-17188M)
>
> I'll open a bug on jira as you request.
>
>
>
> Date: Sat, 24 Apr 2010 15:07:58 -0400
> From: Michael Jerris <mike at jerris.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
>
> Can you confirm this is still the case in current git head and if so open
> me a bug on jira about this issue please.
>
> Mike
>
> On Apr 23, 2010, at 8:49 AM, Wellie Chao wrote:
>
> > Yes, I see how display updates are useful on outbound calls now based on
> > your examples, but what about use cases for inbound calls? It doesn't
> seem
> > that useful to have the display update to the extension that is called
> > after the call is answered (everybody knows their own extension, after
> > all). Just wondering why that default behavior was selected. Is there a
> > way to leave display updates turned on for the outbound leg of the call
> > and to turn them off for the inbound leg of the call?
> >
> >
> >
> > Date: Fri, 23 Apr 2010 08:14:52 -0400
> > From: Steven Ward <steve.d.ward at gmail.com>
> > Reply-To: freeswitch-users at lists.freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Caller ID on inbound calls on Polycom
> >
> > To give one example, display updates are very useful in the application
> I'm using.
> >
> > I have operators using speed dials for dialing out to dialplan that does
> a fifo out; so I see what the speed dial is when I
> > press it (e.g. Operator_Call); but then when the operator is connected to
> the actual caller, the display updates to the CID of
> > the caller she's talking to - very useful.
> >
> > Also, people using a SIP phone to call into a fifo see the number they
> dialed (the number that gets them into the fifo).  But
> > when they're answered by an actual phone, their display updates to the
> number of the endpoint they're talking to.
> >
> >
> >
> > On Thu, Apr 22, 2010 at 10:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
> >      As I mentioned in my last email, the ignore_display_updates variable
> did
> >      the trick, but I am just curious: what is the purpose of the default
> >      behavior where the display changes to the callee name and number? I
> am
> >      guessing there must be some use case (in fact a fairly prevalent use
> case)
> >      where showing the callee is more desirable than showing the caller,
> but
> >      I'm not aware of what that might be and am curious to find out what
> the
> >      use case(s) are. Changing the display to callee also seems to affect
> call
> >      logs on the Polycom, which makes the received call log kind of
> useless.
> >
> >
> >      Date: Thu, 22 Apr 2010 22:57:53 +0300
> >      From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> >      Reply-To: freeswitch-users at lists.freeswitch.org
> >      To: freeswitch-users at lists.freeswitch.org
> >      Subject: Re: [Freeswitch-users] Caller ID on inbound calls on
> Polycom
> >
> >      Try adding the following line in your diaplan before the bridge
> command:
> >
> >      <action application="set" data="ignore_display_updates=true"/>
> >      (try true and false and see which one works better for you).
> >
> >                                 __Yehavi:
> >      2010/4/22 Wellie Chao <wchao at yahoo.com>
> >           I recently updated my FreeSWITCH installation from 1.0.4 to
> 1.0.6 and now
> >           I see a strange behavior with caller ID on Polycom phones when
> handling
> >           inbound calls.
> >
> >           Here is the scenario:
> >
> >           * Call is from 212-555-2222 (external number not on my
> softswitch)
> >
> >           * Call is to 212-555-1001 (number on my softswitch, extension
> 1001)
> >
> >           * extension 1001 is a Polycom phone (it's an IP301, but same
> problem
> >             occurs on other Polycom models such as the 501 and 601). SIP
> 3.1.4 and
> >             SIP 3.1.6 both affected. SIP 3.2.3 doesn't run on the
> IP301/501/601.
> >
> >           * on phone 1001, the caller ID will display 212-555-2222 while
> ringing.
> >             The moment I pick up, the display will change to "From: 1001"
> (referring
> >             to the extension of the phone itself).
> >
> >           Has anyone else experienced this problem, and does anyone know
> how to fix
> >           it? It does not occur with the snom 320 (and I assume it does
> not occur
> >           with any of the snom models based on extrapolation).
> >
> >           While the problem only started when I updated from 1.0.4 to
> 1.0.6, it's
> >           entirely possible it's a configuration setting on the Polycom
> rather than
> >           a FreeSWITCH issue. I'm not sure. Anyone have pointers about
> how to
> >           resolve the problem (or even how to go about investigating the
> root
> >           cause)?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From anthony.minessale at gmail.com  Mon Apr 26 07:46:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Apr 2010 09:46:36 -0500
Subject: [Freeswitch-users] some questions about cluster FS by using
	ODBC in the core.
In-Reply-To: <329861D8B5881E6C7DC0B78F46F163AA78962224@hungngm>
References: <329861D8B5881E6C7DC0B78F46F163AA78962224@hungngm>
Message-ID: <s2l191c3a031004260746u75c41c10n83a0acf3d10299ac@mail.gmail.com>

make up domain like internal-cluster.com
set both force-reg-domain and force-reg-domain-db to this same value
set you domain to that and make an alias for i in the profile on both boxes.
then when you call 1234 at internal-cluster.com it will call on either box.

Works even better if you use a real fqhn in place internal-cluster.com


2010/4/26 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi list.
>
> I try to deploy cluster FS by using  "ODBC in the core".  But i have some
> problem when deploy:
>
> 1. I deploy 2 FS share registration ( by config odbc-dsn in internal
> profile). With default config:
> User 1001 registers in fs1: 10.2.48.194, user 1003 registers in fs2:
> 10.2.48.243.
> I show sofia status profile internal and see to registration:
>
> Call-ID:
> d3298837aa18ac2bOGYyNTYyZGJjNzcyNzNhNDA3MTI0MmY4MmMzOWFkZDU.
> User:           1001 at 10.2.48.194
> Contact:        "1001" <sip:1001 at 10.2.48.184:42064
> ;rinstance=22b332e0ef7c7fc1>
> Agent:          eyeBeam release 1003s stamp 31159
> Status:         Registered(UDP)(unknown) EXP(2010-04-16 11:26:47)
> Host:           FS1.voip.vn
> IP:             10.2.48.184
> Port:           42064
> Auth-User:      1001
> Auth-Realm:     fs.vn
> MWI-Account:    1001 at 10.2.48.194
>
> Call-ID:        NzFiZmQxM2FlZDdmZjU3NTk4NjY4NmIwOWQ4NTcwZDI.
> User:           1003 at 10.2.48.243
> Contact:        "100311" <sip:1003 at 10.2.48.184:19374
> ;rinstance=a7557f050b39a086>
> Agent:          X-Lite release 1104o stamp 56125
> Status:         Registered(UDP)(unknown) EXP(2010-04-27 12:18:37)
> Host:           localhost.localdomain
> IP:             10.2.48.184
> Port:           19374
> Auth-User:      1003
> Auth-Realm:     fs.vn
> MWI-Account:    1003 at 10.2.48.243
>  <1001 at fs.vn>
>  user 1001 can't dial user 1003 and vice-versa. It likes because user 1001
> calls 1003 at 10.2.48.194 while user 1003 is registers 10.2.48.243.
> I read in list that can call between FS with odbc in the core. Do i miss
> config something ?
>
> 2. I have read some comments with 2 FS, we need to config DNS-SRV for load
> balance between them. But I wonder how gateway connects with fs by
> load-balance SIP trunk?
>
> Tks for help.
>
>
>
> Nguy?n M?nh H?g
>
> Team Member - Telecom Security
> Bkis Telecom Dept.
>
> Bkis Internet Security
>   Office:  HH1 Building - Yen Hoa, Cau Giay, Hanoi
>  Tel:      (84 4) 37677090 ext 101
>  Mobile: (84) 1656 722 375
>  Email:  HungNgM at bkav.com.vn
> Website: www.bkav.com.vn
>  ____________________________________________
>  Do your best, the rest will come !
> H? l? vi?c h?t m?h, nh?ng ?i?u t?t ??p s? ??n v?i b?n !
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From shroukkhan at softverk.is  Mon Apr 26 07:56:13 2010
From: shroukkhan at softverk.is (Shrouk Khan)
Date: Mon, 26 Apr 2010 21:56:13 +0700
Subject: [Freeswitch-users] jBilling Setup on freeswitch
Message-ID: <x2uaea1d7231004260756u95d9e3ecu12c6de948c856150@mail.gmail.com>

hi , has anyone been able to integrate freeswtich with jbilling ?
 swearching google did not come up with much answer , but it is claimed that
freeswitch works with jbilling . I already tried astpp , but it wasnt very
impressive.

Can anyone please point me to some resources or tips on that ? :)

-- 
Regards

Shrouk Khan (Khan)
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From alain.meliot at gmail.com  Mon Apr 26 09:28:32 2010
From: alain.meliot at gmail.com (Alain MELIOT)
Date: Mon, 26 Apr 2010 12:28:32 -0400
Subject: [Freeswitch-users] Calling Web service from Freeswitch
Message-ID: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>

Hi All

New to freeswitch i rewrite an asterisk application to freeswitch.
I have a javascript application where i collect some user data and i must
send  the data to a web service (wsdl) for processing.
The problem is that i have no idea  how to call the web service from curl or
javascript.
Any help will be welcome.
Thank in advance

Alain MELIOT
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From steveayre at gmail.com  Mon Apr 26 09:55:57 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Apr 2010 17:55:57 +0100
Subject: [Freeswitch-users] Calling Web service from Freeswitch
In-Reply-To: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
References: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
Message-ID: <q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>

To use mod_curl to do a GET/POST request:

<action application="curl"
data="http://www.myhost.com/script?getName=myGetValue" />
<action application="curl" data="http://www.myhost.com/script post
postName=myPostValue%20a%20space" />

You'd have to form a request string to match what your application expects. E.g.
<action application="curl" data="http://www.myhost.com post
username=${username}&ip=${network_addr}"/>

You can parse the XML from a script after the curl request, you'll
find the curl_response_code and curl_response variables contain the
HTTP status code (200 if ok) and XML result.

-Steve


On 26 April 2010 17:28, Alain MELIOT <alain.meliot at gmail.com> wrote:
>
> Hi All
>
> New to freeswitch i rewrite an asterisk application?to freeswitch.
> I have a javascript application where i collect some user data and i must
> send? the data to a web service (wsdl) for processing.
> The problem is that i have no idea? how to call the web service from curl or
> javascript.
> Any?help will be welcome.
> Thank in advance
>
> Alain MELIOT
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Apr 26 09:57:18 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Apr 2010 17:57:18 +0100
Subject: [Freeswitch-users] Calling Web service from Freeswitch
In-Reply-To: <q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
References: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
	<q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
Message-ID: <h2jeafa772f1004260957z5ae20964n81467738230c69b2@mail.gmail.com>

mod_curl documentation here:
http://wiki.freeswitch.org/wiki/Mod_curl

-Steve


On 26 April 2010 17:55, Steven Ayre <steveayre at gmail.com> wrote:
> To use mod_curl to do a GET/POST request:
>
> <action application="curl"
> data="http://www.myhost.com/script?getName=myGetValue" />
> <action application="curl" data="http://www.myhost.com/script post
> postName=myPostValue%20a%20space" />
>
> You'd have to form a request string to match what your application expects. E.g.
> <action application="curl" data="http://www.myhost.com post
> username=${username}&ip=${network_addr}"/>
>
> You can parse the XML from a script after the curl request, you'll
> find the curl_response_code and curl_response variables contain the
> HTTP status code (200 if ok) and XML result.
>
> -Steve
>
>
> On 26 April 2010 17:28, Alain MELIOT <alain.meliot at gmail.com> wrote:
>>
>> Hi All
>>
>> New to freeswitch i rewrite an asterisk application?to freeswitch.
>> I have a javascript application where i collect some user data and i must
>> send? the data to a web service (wsdl) for processing.
>> The problem is that i have no idea? how to call the web service from curl or
>> javascript.
>> Any?help will be welcome.
>> Thank in advance
>>
>> Alain MELIOT
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From alain.meliot at gmail.com  Mon Apr 26 10:12:59 2010
From: alain.meliot at gmail.com (Alain MELIOT)
Date: Mon, 26 Apr 2010 13:12:59 -0400
Subject: [Freeswitch-users] Calling Web service from Freeswitch
In-Reply-To: <q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
References: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
	<q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
Message-ID: <i2o8076277e1004261012se5c646ear279e872b2be10db3@mail.gmail.com>

Hi steven

Thank you for your answer
i will try

2010/4/26 Steven Ayre <steveayre at gmail.com>

> To use mod_curl to do a GET/POST request:
>
> <action application="curl"
> data="http://www.myhost.com/script?getName=myGetValue" />
> <action application="curl" data="http://www.myhost.com/script post
> postName=myPostValue%20a%20space" />
>
> You'd have to form a request string to match what your application expects.
> E.g.
> <action application="curl" data="http://www.myhost.com post
> username=${username}&ip=${network_addr}"/>
>
> You can parse the XML from a script after the curl request, you'll
> find the curl_response_code and curl_response variables contain the
> HTTP status code (200 if ok) and XML result.
>
> -Steve
>
>
> On 26 April 2010 17:28, Alain MELIOT <alain.meliot at gmail.com> wrote:
> >
> > Hi All
> >
> > New to freeswitch i rewrite an asterisk application to freeswitch.
> > I have a javascript application where i collect some user data and i must
> > send  the data to a web service (wsdl) for processing.
> > The problem is that i have no idea  how to call the web service from curl
> or
> > javascript.
> > Any help will be welcome.
> > Thank in advance
> >
> > Alain MELIOT
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mardy at voysys.com  Mon Apr 26 12:50:21 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Mon, 26 Apr 2010 15:50:21 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<z2jebbc2d551004191412g98150508lcb0fee976f3d1bf3@mail.gmail.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
Message-ID: <30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>

Here is the setup that was used to reproduce the consultative transfer problem.

There are two boxes, the first is running a proxy based PBX (sipXecs) and the
second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones, extension
200 and 202, registered with it.  The PBX has configured a mapping rule which
will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060 which
is the address of the second box running FreeSWITCH.  FreeSWITCH has been
configured to allow connections from the PBX box via an ACL configuration and
the public dialplan includes an "echo" extension:

    <extension name="echo">
      <condition field="destination_number" expression="^9996$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>

Any of the phones registered with the PBX can dial extension 9996 and be
connected to the FreeSWITCH echo application.  But when one phone attempts
to transfer another phone to extension 9996 via a consultative transfer,
FreeSWITCH does not properly complete the transfer.  You can see in the log
at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
initiated the transfer.  Without that terminating BYE, the transfer
controller thinks that the transfer failed.

The corresponding FreeSWITCH log file - http://pastebin.freeswitch.org/12806

If it will help, I can also forward a corresponding PCAP file.

Thanks

-Mardy

On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:

> I'm trying to understand this:
> 
> If FS is acting as a phone in your scenario why are you sending a refer to it and not the server?
> In most situations there is a b2bua server who routes the calls and takes all the REFER.
> Is this one of those PROXY only sip servers?
> 
> I think you would need to produce a full debug log of this, and if you are using some kind of proxy based setup we would need some way to easily reproduce it or visit your lab because we do not typically use anything of the sort.
> 
> Execute these commands and reproduce it and capture the whole log and put it on
> http://pastebin.freeswitch.org
> 
> sofia profile internal siptrace on
> console loglevel debug
> 
> 
> 
> 
> 
> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> instead of emailing again when impatient for an answer (something we frown upon here in this busy list) 
> produce a reproducible step by step process to duplicate your issue.  We are trying to help people but we don't have the time to do the leg work for everyone who asks a question when we get hundreds of emails a day.
> 
> 
> 
> 
> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
> Just following up...  Does anyone have any suggestions on how to proceed with this?  I've run out of ideas.
> 
> Thanks,
> 
> -Mardy
> 
> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
> 
>> The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.
>> 
>> To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.
>> 
>> -Mardy
>> 
>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>> 
>>> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
>>> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
>>> 
>>> 
>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
>>> 
>>> -Mardy
>>> 
>>> 
>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>> 
>>>> but what is the client sending the REFER?
>>>> 
>>>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>>>> 
>>>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>>>> your UA has an interop problem.
>>>> 
>>>> 
>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>> 
>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>> 
>>>>> uuid_simplify will issue the refer...
>>>> 
>>>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>>>> 
>>>>> 
>>>>> May I ask what application you are developing?
>>>> 
>>>> An ACD.
>>>> 
>>>>> 
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>> FSComm developer
>>>>> 
>>>>> 
>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>>>> 
>>>>> -Mardy
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> -- 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> 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
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Mon Apr 26 13:18:49 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 26 Apr 2010 16:18:49 -0400
Subject: [Freeswitch-users] Any breaking change on register with 1.0.6
In-Reply-To: <BFFF601E-68E0-4AEF-A78B-4ADC03C97F9D@jerris.com>
References: <w2m367751821004231055h8b2ea90cxe0046f3c61ab59f3@mail.gmail.com>
	<BFFF601E-68E0-4AEF-A78B-4ADC03C97F9D@jerris.com>
Message-ID: <n2i367751821004261318o5766019fz72d7b32e68e83f81@mail.gmail.com>

Thanks for the reply Mike.

Anthony gave me the heads up on irc that NAT head been simplified in 1.0.6 -
so I looked at the cnf file.

# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "71.xxx.60.xxx"
voip_control_port: "5900"
start_media_port: "16384"
end_media_port:  "32766"
nat_received_processing: "0"

Setting nat_received_processing: "1" did  the trick and the Cisco phone is
now registering correctly again.

Cheers

On Sat, Apr 24, 2010 at 3:23 PM, Michael Jerris <mike at jerris.com> wrote:

> can you provide any logs of the failure?
>
> On Apr 23, 2010, at 1:55 PM, Phillip Jones wrote:
>
> > Hi there,
> >
> > I have been using my Cisco phones registered with FreeSWITCH (1.0.4) for
> months now. Today I upgraded to 1.0.6 and they no longer registers.
> >
> > This what configured to get them to work.
> >
> > 1) Add
> >
> >  <list name="lan" default="allow">
> >         <node type="allow" cidr="71.**5.60.***/32"/>
> >
> > to acl.conf.xml
> >
> > 2) Add the appropriate directory entry.
> >
> >
> > That worked (may be it shouldn't have done) - but on 1.0.6 (today's
> trunk) is fails.
> >
> > Is something else required in 1.0.6 do you 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 kenfulmer at icstechnologysolutions.com  Mon Apr 26 13:49:24 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 26 Apr 2010 15:49:24 -0500
Subject: [Freeswitch-users] Add a diversion header?
Message-ID: <024701cae581$f48b9250$dda2b6f0$@com>

We have customers using a sipX internal PBX and a Freeswitch device as a
softswitch to route calls to our upstream provider, PaeTec. However, the
provider doesn't support the P-Asserted Identity for external call
forwarding. PaeTec wants to see a SIP Diversion header instead. 


How can we add this header in the Freeswitch device? 

 

Thanks,

 

Ken Fulmer

 

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From freeswitch at anticat.ch  Mon Apr 26 13:49:03 2010
From: freeswitch at anticat.ch (=?ISO-8859-15?Q?Andreas_Dr=F6scher?=)
Date: Mon, 26 Apr 2010 22:49:03 +0200
Subject: [Freeswitch-users] Route calls based on the Nb they are divert by
Message-ID: <4BD5FC3F.7060104@anticat.ch>

Hi everyone

I was wondering if it is possible route calls based on the numbers they were 
divert by, perhaps using "sip_redirected_by".

Let's assume I own two phone numbers of my old fixed telephone lines and one 
VoIP Account/Number. If I redirect the two fixnet numbers to the same VoIP 
number, is it possible to "demultiplex" the incoming calls based on the number 
they were redirected by?

I am not sure about the available meta data provided by SIP, however I am 
certain that my ISDN phone is sometimes showing "xxxx is calling, redirected by 
yyyyy"

Best Wishes
Andreas


From kenfulmer at icstechnologysolutions.com  Mon Apr 26 14:01:18 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 26 Apr 2010 16:01:18 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>	<017601cae25f$b2067aa0$16136fe0$@com>
	<3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>
Message-ID: <025201cae583$9df89170$d9e9b450$@com>

I try to add the license files to the system but get the following error:
ERROR [(null)]

 

I noticed that the license.zip file has 0 bytes. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 4:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org

 

If you buy more licenses... activate them on the machine,, copy the .conf
file to /etc/freeswitch then restart or HUP the freeswitch_licence_server
that is running and it will add the licenses together into one.

 

/b

 

On Apr 22, 2010, at 4:06 PM, Ken Fulmer wrote:





I apologize, but I'm not sure what you mean.

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 3:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org
<http://Freeswitch.org/> 

 

Add the additional license files in and HUP the freeswitch_licence_server
thats running.

 

/b

 

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From brian at freeswitch.org  Mon Apr 26 14:04:58 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Apr 2010 16:04:58 -0500
Subject: [Freeswitch-users] Add a diversion header?
In-Reply-To: <024701cae581$f48b9250$dda2b6f0$@com>
References: <024701cae581$f48b9250$dda2b6f0$@com>
Message-ID: <E261EDBC-972A-4922-A2B1-B98079A9680C@freeswitch.org>

set/export the variable sip_h_Diversion

/b

On Apr 26, 2010, at 3:49 PM, Ken Fulmer wrote:

> We have customers using a sipX internal PBX and a Freeswitch device as a softswitch to route calls to our upstream provider, PaeTec. However, the provider doesn?t support the P-Asserted Identity for external call forwarding. PaeTec wants to see a SIP Diversion header instead.
> 
> How can we add this header in the Freeswitch device?
>  
> Thanks,
>  
> Ken Fulmer

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From brian at freeswitch.org  Mon Apr 26 14:07:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Apr 2010 16:07:15 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <025201cae583$9df89170$d9e9b450$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>	<017601cae25f$b2067aa0$16136fe0$@com>
	<3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>
	<025201cae583$9df89170$d9e9b450$@com>
Message-ID: <D27CEB1E-A30D-435D-AA90-53EE18A364D0@freeswitch.org>

You know I think I might know what is going on... 

/b

On Apr 26, 2010, at 4:01 PM, Ken Fulmer wrote:

> I try to add the license files to the system but get the following error:     ERROR [(null)]
>  
> I noticed that the license.zip file has 0 bytes.
>  
> Ken

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From brian at freeswitch.org  Mon Apr 26 14:11:02 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Apr 2010 16:11:02 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <025201cae583$9df89170$d9e9b450$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>	<017601cae25f$b2067aa0$16136fe0$@com>
	<3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>
	<025201cae583$9df89170$d9e9b450$@com>
Message-ID: <2C818FDF-41E6-4116-B5E7-6E4284895062@freeswitch.org>

Are you trying to activate both licenses at the same time again? or are you entering just the NEW license into the validator?

/b

On Apr 26, 2010, at 4:01 PM, Ken Fulmer wrote:

> I try to add the license files to the system but get the following error:     ERROR [(null)]
>  
> I noticed that the license.zip file has 0 bytes.
>  
> Ken
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Thursday, April 22, 2010 4:14 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org
>  
> If you buy more licenses... activate them on the machine,, copy the .conf file to /etc/freeswitch then restart or HUP the freeswitch_licence_server that is running and it will add the licenses together into one.
>  
> /b
>  
> On Apr 22, 2010, at 4:06 PM, Ken Fulmer wrote:
> 
> 
> I apologize, but I?m not sure what you mean.
>  
> Thanks,
>  
> Ken
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Thursday, April 22, 2010 3:06 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org
>  
> Add the additional license files in and HUP the freeswitch_licence_server thats running.
>  
> /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

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From kenfulmer at icstechnologysolutions.com  Mon Apr 26 14:27:35 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 26 Apr 2010 16:27:35 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <025201cae583$9df89170$d9e9b450$@com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>	<012c01cae255$44971510$cdc53f30$@com>	<ACAC3ADF-A721-4770-8AB0-92D6E62BE0DE@freeswitch.org>	<017601cae25f$b2067aa0$16136fe0$@com>	<3A139AFB-72C2-4ED7-8D79-E68723140FD7@freeswitch.org>
	<025201cae583$9df89170$d9e9b450$@com>
Message-ID: <025d01cae587$4a5c7780$df156680$@com>

Please disregard. Once I rebooted the server, the license files were
combined. 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Monday, April 26, 2010 4:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org

 

I try to add the license files to the system but get the following error:
ERROR [(null)]

 

I noticed that the license.zip file has 0 bytes. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 4:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org

 

If you buy more licenses... activate them on the machine,, copy the .conf
file to /etc/freeswitch then restart or HUP the freeswitch_licence_server
that is running and it will add the licenses together into one.

 

/b

 

On Apr 22, 2010, at 4:06 PM, Ken Fulmer wrote:

 

I apologize, but I'm not sure what you mean.

 

Thanks,

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 22, 2010 3:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g.729 Licenses from Freeswitch.org
<http://Freeswitch.org/> 

 

Add the additional license files in and HUP the freeswitch_licence_server
thats running.

 

/b

 

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From rupa at rupa.com  Mon Apr 26 15:04:33 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 26 Apr 2010 17:04:33 -0500
Subject: [Freeswitch-users] Route calls based on the Nb they are divert
	by
In-Reply-To: <4BD5FC3F.7060104@anticat.ch>
References: <4BD5FC3F.7060104@anticat.ch>
Message-ID: <u2lb39273531004261504v575ca729r320d5991f5e2d510@mail.gmail.com>

It all depends on what your ITSP sends you.  Try the "info" app to see if
there is any mention of the number you are diverting from.  Pretty much
everything there can be used for routing decisions.

On Mon, Apr 26, 2010 at 3:49 PM, Andreas Dr?scher <freeswitch at anticat.ch>wrote:

> Hi everyone
>
> I was wondering if it is possible route calls based on the numbers they
> were
> divert by, perhaps using "sip_redirected_by".
>
> Let's assume I own two phone numbers of my old fixed telephone lines and
> one
> VoIP Account/Number. If I redirect the two fixnet numbers to the same VoIP
> number, is it possible to "demultiplex" the incoming calls based on the
> number
> they were redirected by?
>
> I am not sure about the available meta data provided by SIP, however I am
> certain that my ISDN phone is sometimes showing "xxxx is calling,
> redirected by
> yyyyy"
>
> Best Wishes
> Andreas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From anthony.minessale at gmail.com  Mon Apr 26 15:22:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Apr 2010 17:22:30 -0500
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
	<p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
Message-ID: <x2g191c3a031004261522z4b607aaaw19b3e75a084d5d9e@mail.gmail.com>

Based on your response, I could be wrong but, I get the impression you are
intentionally trying to stir up some kind of controversy.  Is there a
specific reason you feel the need to raise your concerns on our general
users mailing lists and not to consulting at freeswitch.org where they belong.
This has gone from asking some questions to some sort of rhetoric against
our policy that clearly already addresses your concerns.

Do you really think you are going to have a whole bunch of boxes fail to
some degree that we will not be willing to re-issue your licenses or are you
just more interested in starting a flame war on our mailing list?  The
licenses are to the specifications of the requirements of the contracts we
signed.  We did so taking the on the risk and responsibility to make g729
available to the community.

We had a discussion about floating licenses and concluded it was unwise and
not worth the risk and security implications.  If you use g729 you know how
much traffic you are going to use and if you want to remain stable you
should not want to add another moving part to the equation that could go
down and make all of your calls fail.

This is the policy we have implemented and I think you are getting the good
end of the bargain with a free telephony server with an efficient stable
g729 codec implementation at an affordable cost.  If a high-enough demand
arises for floating licenses and we have the time and resources to implement
it, we may reconsider it but only a tiny fraction of our customers have even
asked about it, let alone demanded it.



On Fri, Apr 23, 2010 at 12:30 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Send proof the machine is dead? Would you like a picture of smoke pouring
> out of a box in flames?
>
> What if you want to re-allocate the machine to do something else or just
> replace it with a more powerful machine? How are the licenses tied to the
> machine, MAC address?
>
> Any reason the "option" for a floating license model isn't available? It
> seems given the option between the two, I'd rather have the floating license
> model so I wouldn't need to prove my machine is dead before asking to have
> the licenses re-issued, unless of course the license server is dead. We swap
> production machines in/out all the time, particularly because we perform
> maintenance on one, so we have a spare, bring it up, take the other one
> down, perform maintenance, and bring it back up.
>
> With this model, floating licenses would be our only option, I really
> wouldn't want to be purchasing a bunch of extra licenses for spare machines.
>
> I can't imagine you guys really like making us feel like criminals by
> tieing the license to an actual box, what's wrong with the good 'ole "on
> your honor policy"? I realize the need to pay for valid g729 licenses, but
> prove the machine is dead? Is this requirement coming from Sipro?
>
> Cheers,
> Gabe
>
>
> On Thu, Apr 22, 2010 at 11:30 AM, Brian West <brian at freeswitch.org> wrote:
>
>> If you have to replace the machine you'll have to send me the proof the
>> machine is dead and I can reset them... I can only do this 3 times before
>> you have to repurchase the license.
>>
>> /b
>>
>> On Apr 22, 2010, at 1:24 PM, Ken Fulmer wrote:
>>
>> 1.       Are these licenses additive? In other words, if I?ve purchased
>> one for a machine and need a total of five, can I purchase four more and add
>> them to the mix?
>> 2.       If we have to replace a machine, how can we replace the licenses
>> without repurchasing them?
>>
>> Thanks,
>>
>> Ken Fulmer
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tacvbo at tacvbo.net  Mon Apr 26 17:25:26 2010
From: tacvbo at tacvbo.net (tacvbo at tacvbo.net)
Date: Tue, 27 Apr 2010 02:25:26 +0200
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <d9853e4f1003272054r4842bfd1le9dbf18e2f137353@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com> 
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com> 
	<d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com> 
	<EEAC4F0A-3B09-48F5-8026-BC3F56AD4525@freeswitch.org>
	<d9853e4f1003272054r4842bfd1le9dbf18e2f137353@mail.gmail.com>
Message-ID: <g2od9853e4f1004261725u69fab3c9s82511b4a47f28733@mail.gmail.com>

Any thoughts regard this?  The scenario is a Sangoma card without HWEC
using TDM API or DAHDI without software EC but with very noticeable
echo. Trying to use DAHDI + OSLEC or MG2 with OpenZap there is no
sound.


On Sun, Mar 28, 2010 at 05:54, tacvbo at tacvbo.net <tacvbo at tacvbo.net> wrote:
> On Sat, Mar 27, 2010 at 14:30, Brian West <brian at freeswitch.org> wrote:
>> You don't have to use Dahdi with any Sangoma card in conjunction with OpenZAP... its able to use the native interface.
>
> I know and I've tried that, I've forgot to mention: card do not have
> HWEC. The problem is if I use TDM API mode, there is echo as if I use
> DAHDI without SWEC.
>
> --
> Octavio H. Ruiz Cervera

-- 
Octavio H. Ruiz Cervera


From jason at jasonjgw.net  Mon Apr 26 18:19:13 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 27 Apr 2010 11:19:13 +1000
Subject: [Freeswitch-users] Building latest Git master branch with ZRTP
	enabled
Message-ID: <20100427011913.GA30687@jdc.jasonjgw.net>

Upon trying to build the latest code from Git with ZRTP enabled I get:

src/switch_rtp.c: In function ?read_rtcp_packet?:
src/switch_rtp.c:2090: error: cast from pointer to integer of different size
src/switch_rtp.c:2097: error: assignment makes pointer from integer without a
cast
src/switch_rtp.c:2105: error: ?ret? undeclared (first use in this function)
src/switch_rtp.c:2105: error: (Each undeclared identifier is reported only
once
src/switch_rtp.c:2105: error: for each function it appears in.)
src/switch_rtp.c:2106: error: label ?end? used but not defined
src/switch_rtp.c:2102: error: label ?do_continue? used but not defined

Debian x86_64 architecture, gcc 4.4.3.

I haven't looked at the code in detail, but it is all enclosed in #ifdef
preprocessor directives, so perhaps surrounding code has been changed without
updating the ZRTP-dependent portions.

Since I'm not familiar with the changes, the fixes weren't obvious.

It's also possible that I tried to build at just the wrong moment.




From dujinfang at gmail.com  Mon Apr 26 20:25:29 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 27 Apr 2010 11:25:29 +0800
Subject: [Freeswitch-users] Calling Web service from Freeswitch
In-Reply-To: <i2o8076277e1004261012se5c646ear279e872b2be10db3@mail.gmail.com>
References: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
	<q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
	<i2o8076277e1004261012se5c646ear279e872b2be10db3@mail.gmail.com>
Message-ID: <n2o23f91031004262025x8bead428ze2d7c0d579745d9c@mail.gmail.com>

is there a timeout param for curl ?

2010/4/27 Alain MELIOT <alain.meliot at gmail.com>:
> Hi steven
>
> Thank you for your answer
> i will try
>
> 2010/4/26 Steven Ayre <steveayre at gmail.com>
>>
>> To use mod_curl to do a GET/POST request:
>>
>> <action application="curl"
>> data="http://www.myhost.com/script?getName=myGetValue" />
>> <action application="curl" data="http://www.myhost.com/script post
>> postName=myPostValue%20a%20space" />
>>
>> You'd have to form a request string to match what your application
>> expects. E.g.
>> <action application="curl" data="http://www.myhost.com post
>> username=${username}&ip=${network_addr}"/>
>>
>> You can parse the XML from a script after the curl request, you'll
>> find the curl_response_code and curl_response variables contain the
>> HTTP status code (200 if ok) and XML result.
>>
>> -Steve
>>
>>
>> On 26 April 2010 17:28, Alain MELIOT <alain.meliot at gmail.com> wrote:
>> >
>> > Hi All
>> >
>> > New to freeswitch i rewrite an asterisk application?to freeswitch.
>> > I have a javascript application where i collect some user data and i
>> > must
>> > send? the data to a web service (wsdl) for processing.
>> > The problem is that i have no idea? how to call the web service from
>> > curl or
>> > javascript.
>> > Any?help will be welcome.
>> > Thank in advance
>> >
>> > Alain MELIOT
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn


From cucku.cucku at yahoo.com.vn  Mon Apr 26 20:40:20 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Tue, 27 Apr 2010 11:40:20 +0800 (SGT)
Subject: [Freeswitch-users]  need help on IVR
In-Reply-To: <1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
Message-ID: <166310.44652.qm@web76201.mail.sg1.yahoo.com>

Hi Mike

let me make it clear :)

ip phone1 (ext 1000) registers to SIP server
ip phone2 (ext 1002) registers to SIP server
FS (ext 1003) register to SIP server

the media proxy and sip server are in the same server

the signaling flow :
 ext1000 --->  INVITE ---> SIP server ---> INVITE---> ext 1002

the media flow:
ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002

so the signaling for FS when i call 1003
step1: ext1000--->INVITE--->Sip server---->INVITE---->FS
step2: ext1000 press 1002, FS will transfer call to SIP server
signaling flow for step 2:
1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold
2. FS --->INVITE---> SIP server----> INVITE--->ext1002
when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 
3. ext1000 connected to ext1002

media flow:
step1:ext1000----RTP---media proxy----RTP---FS
step2:
1. FS -----no RTP---media proxy----no RTp----ext1000
2. FS----RTP----media proxy---ext1002
3 ext1000----media proxy----ext1002


Thank you





________________________________
T?: Michael Jerris <mike at jerris.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 11:23:25, Th? Hai, 26 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] need help on IVR

I meant this literally.  People always use the term "sip server" and I never understand what they are trying to say.  What exactly is a sip server?  I think you have your terms all completely confused in your explanation below or I am just being really dense.  This cisco server is a sip server and a media proxy?  Can you try to be very clear on what exactly these things are and what you are trying to do?

Mike


On Apr 25, 2010, at 10:44 PM, false wrote:

Hi Michael Jerris
>
>
>my sip server is cisco sip server
>could you guide me some clue or show the same config for the case
>
>
>Thank you
>Ha`
>
>
________________________________
T?: Michael Jerris <mike at jerris.com>
>??n: freeswitch-users at lists.freeswitch.org
>G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
>Ch? ??: Re: [Freeswitch-users] need help on IVR
>
>what is a sip server?  Yes, you can do pretty much anything like this.
>
>
>On Apr 19, 2010, at 12:08 AM, false wrote:
>
>Hi all
>>
>>
>>my network topology:
>>
>>
>>endpoint 1(100)-----sip server ---IVR(Freeswitch)
>>|
>>|
>>endpoint2(101)
>>
>>
>>endpoint1 + endpoint2 are registered to sip server
>>Freeswitch is regsitered to sip server with 103
>>
>>
>>my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
>>and RTP from endpoint 1 --> media proxy---> FS
>>then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
>>and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
>>
>>
>>can FS do it


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From mike at jerris.com  Mon Apr 26 21:02:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Apr 2010 00:02:52 -0400
Subject: [Freeswitch-users] need help on IVR
In-Reply-To: <166310.44652.qm@web76201.mail.sg1.yahoo.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>
Message-ID: <2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>

Well.. I still don't get what a "sip server" is.  Lets just call it a figgledingerbort, that term seems just as useful and defined.  You could use the deflect app to do what you seem to be suggesting.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect

This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri.

Mike



On Apr 26, 2010, at 11:40 PM, false wrote:

> Hi Mike
> 
> let me make it clear :)
> 
> ip phone1 (ext 1000) registers to SIP server
> ip phone2 (ext 1002) registers to SIP server
> FS (ext 1003) register to SIP server
> 
> the media proxy and sip server are in the same server
> 
> the signaling flow :
>  ext1000 --->  INVITE ---> SIP server ---> INVITE---> ext 1002
> 
> the media flow:
> ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002
> 
> so the signaling for FS when i call 1003
> step1: 	ext1000--->INVITE--->Sip server---->INVITE---->FS
> step2: 	ext1000 press 1002, FS will transfer call to SIP server
> signaling flow for step 2:
> 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold
> 2. FS --->INVITE---> SIP server----> INVITE--->ext1002
> when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 
> 3. ext1000 connected to ext1002
> 
> media flow:
> step1:	ext1000----RTP---media proxy----RTP---FS
> step2:
> 1. FS -----no RTP---media proxy----no RTp----ext1000
> 2. FS----RTP----media proxy---ext1002
> 3 ext1000----media proxy----ext1002
> 
> 
> Thank you
> 
> 
> 
> T?: Michael Jerris <mike at jerris.com>
> ??n: freeswitch-users at lists.freeswitch.org
> G?i ng?y: 11:23:25, Th? Hai, 26 th?ng 4 2010 
> Ch? ??: Re: [Freeswitch-users] need help on IVR
> 
> I meant this literally.  People always use the term "sip server" and I never understand what they are trying to say.  What exactly is a sip server?  I think you have your terms all completely confused in your explanation below or I am just being really dense.  This cisco server is a sip server and a media proxy?  Can you try to be very clear on what exactly these things are and what you are trying to do?
> 
> Mike
> 
> On Apr 25, 2010, at 10:44 PM, false wrote:
> 
>> Hi Michael Jerris
>> 
>> my sip server is cisco sip server
>> could you guide me some clue or show the same config for the case
>> 
>> Thank you
>> Ha`
>> T?: Michael Jerris <mike at jerris.com>
>> ??n: freeswitch-users at lists.freeswitch.org
>> G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
>> Ch? ??: Re: [Freeswitch-users] need help on IVR
>> 
>> what is a sip server?  Yes, you can do pretty much anything like this.
>> 
>> On Apr 19, 2010, at 12:08 AM, false wrote:
>> 
>>> Hi all
>>> 
>>> my network topology:
>>> 
>>> endpoint 1(100)-----sip server ---IVR(Freeswitch)
>>> 					|
>>> 					|	
>>> 				endpoint2(101)
>>> 
>>> endpoint1 + endpoint2 are registered to sip server
>>> Freeswitch is regsitered to sip server with 103
>>> 
>>> my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
>>> and RTP from endpoint 1 --> media proxy---> FS
>>> then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
>>> and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
>>> 
>>> can FS do it

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From michal.bielicki at halo2.pl  Mon Apr 26 23:55:47 2010
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Tue, 27 Apr 2010 08:55:47 +0200
Subject: [Freeswitch-users] need help on IVR
In-Reply-To: <2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
Message-ID: <59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>

Are there fries availabkle with a figgledingerbort ?

Am 27.04.2010 um 06:02 schrieb Michael Jerris:

> Well.. I still don't get what a "sip server" is.  Lets just call it a figgledingerbort, that term seems just as useful and defined.  You could use the deflect app to do what you seem to be suggesting.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> 
> This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri.
> 
> Mike
> 
> 
> 
> On Apr 26, 2010, at 11:40 PM, false wrote:
> 
>> Hi Mike
>> 
>> let me make it clear :)
>> 
>> ip phone1 (ext 1000) registers to SIP server
>> ip phone2 (ext 1002) registers to SIP server
>> FS (ext 1003) register to SIP server
>> 
>> the media proxy and sip server are in the same server
>> 
>> the signaling flow :
>>  ext1000 --->  INVITE ---> SIP server ---> INVITE---> ext 1002
>> 
>> the media flow:
>> ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002
>> 
>> so the signaling for FS when i call 1003
>> step1: 	ext1000--->INVITE--->Sip server---->INVITE---->FS
>> step2: 	ext1000 press 1002, FS will transfer call to SIP server
>> signaling flow for step 2:
>> 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold
>> 2. FS --->INVITE---> SIP server----> INVITE--->ext1002
>> when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 
>> 3. ext1000 connected to ext1002
>> 
>> media flow:
>> step1:	ext1000----RTP---media proxy----RTP---FS
>> step2:
>> 1. FS -----no RTP---media proxy----no RTp----ext1000
>> 2. FS----RTP----media proxy---ext1002
>> 3 ext1000----media proxy----ext1002
>> 
>> 
>> Thank you
>> 
>> 
>> 
>> T?: Michael Jerris <mike at jerris.com>
>> ??n: freeswitch-users at lists.freeswitch.org
>> G?i ng?y: 11:23:25, Th? Hai, 26 th?ng 4 2010 
>> Ch? ??: Re: [Freeswitch-users] need help on IVR
>> 
>> I meant this literally.  People always use the term "sip server" and I never understand what they are trying to say.  What exactly is a sip server?  I think you have your terms all completely confused in your explanation below or I am just being really dense.  This cisco server is a sip server and a media proxy?  Can you try to be very clear on what exactly these things are and what you are trying to do?
>> 
>> Mike
>> 
>> On Apr 25, 2010, at 10:44 PM, false wrote:
>> 
>>> Hi Michael Jerris
>>> 
>>> my sip server is cisco sip server
>>> could you guide me some clue or show the same config for the case
>>> 
>>> Thank you
>>> Ha`
>>> T?: Michael Jerris <mike at jerris.com>
>>> ??n: freeswitch-users at lists.freeswitch.org
>>> G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
>>> Ch? ??: Re: [Freeswitch-users] need help on IVR
>>> 
>>> what is a sip server?  Yes, you can do pretty much anything like this.
>>> 
>>> On Apr 19, 2010, at 12:08 AM, false wrote:
>>> 
>>>> Hi all
>>>> 
>>>> my network topology:
>>>> 
>>>> endpoint 1(100)-----sip server ---IVR(Freeswitch)
>>>> 					|
>>>> 					|	
>>>> 				endpoint2(101)
>>>> 
>>>> endpoint1 + endpoint2 are registered to sip server
>>>> Freeswitch is regsitered to sip server with 103
>>>> 
>>>> my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
>>>> and RTP from endpoint 1 --> media proxy---> FS
>>>> then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
>>>> and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101
>>>> 
>>>> can FS do it
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa
t. +48228753290 | f. +48228753291 
michal.bielicki at halokwadrat.pl | w. www.halokwadrat.pl
 
Knowledge & Low Prices. Guaranteed!

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From janvb at live.com  Tue Apr 27 02:25:39 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 11:25:39 +0200
Subject: [Freeswitch-users] need help on IVR
In-Reply-To: <2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>,
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>,
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>,
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>,
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>,
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
Message-ID: <SNT135-w41A8A0D7B6881C10F97A9D2030@phx.gbl>


I want a red figgledingerbort :)
 


From: mike at jerris.com
Date: Tue, 27 Apr 2010 00:02:52 -0400
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] need help on IVR

Well.. I still don't get what a "sip server" is.  Lets just call it a figgledingerbort, that term seems just as useful and defined.  You could use the deflect app to do what you seem to be suggesting.


http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect


This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri.


Mike









On Apr 26, 2010, at 11:40 PM, false wrote:



Hi Mike


let me make it clear :)


ip phone1 (ext 1000) registers to SIP server
ip phone2 (ext 1002) registers to SIP server
FS (ext 1003) register to SIP server


the media proxy and sip server are in the same server


the signaling flow :
 ext1000 --->  INVITE ---> SIP server ---> INVITE---> ext 1002


the media flow:
ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002


so the signaling for FS when i call 1003
step1:  ext1000--->INVITE--->Sip server---->INVITE---->FS
step2:  ext1000 press 1002, FS will transfer call to SIP server
signaling flow for step 2:
1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold
2. FS --->INVITE---> SIP server----> INVITE--->ext1002
when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 
3. ext1000 connected to ext1002


media flow:
step1: ext1000----RTP---media proxy----RTP---FS
step2:
1. FS -----no RTP---media proxy----no RTp----ext1000
2. FS----RTP----media proxy---ext1002
3 ext1000----media proxy----ext1002




Thank you








T?: Michael Jerris <mike at jerris.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 11:23:25, Th? Hai, 26 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] need help on IVR

I meant this literally.  People always use the term "sip server" and I never understand what they are trying to say.  What exactly is a sip server?  I think you have your terms all completely confused in your explanation below or I am just being really dense.  This cisco server is a sip server and a media proxy?  Can you try to be very clear on what exactly these things are and what you are trying to do?


Mike



On Apr 25, 2010, at 10:44 PM, false wrote:



Hi Michael Jerris


my sip server is cisco sip server
could you guide me some clue or show the same config for the case


Thank you
Ha`



T?: Michael Jerris <mike at jerris.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 1:08:26, Ch? nh?t, 25 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] need help on IVR

what is a sip server?  Yes, you can do pretty much anything like this.



On Apr 19, 2010, at 12:08 AM, false wrote:



Hi all


my network topology:


endpoint 1(100)-----sip server ---IVR(Freeswitch)
|
| 
endpoint2(101)


endpoint1 + endpoint2 are registered to sip server
Freeswitch is regsitered to sip server with 103


my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR 
and RTP from endpoint 1 --> media proxy---> FS
then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS
and RTP from endpoint1--> media proxy -->endpoint1,  RTP will not go through the FS after FS transfer call to 101


can FS do it
 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From janvb at live.com  Tue Apr 27 02:38:25 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 11:38:25 +0200
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>,
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>,
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>,
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>,
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>,
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>,
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>
Message-ID: <SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>


hi folks,

 

If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.

 

Jan

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From rupa at rupa.com  Tue Apr 27 02:41:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 27 Apr 2010 04:41:18 -0500
Subject: [Freeswitch-users] Calling Web service from Freeswitch
In-Reply-To: <n2o23f91031004262025x8bead428ze2d7c0d579745d9c@mail.gmail.com>
References: <i2l8076277e1004260928pa7f11c9dk3c5d94ccd879eb36@mail.gmail.com>
	<q2weafa772f1004260955v6217aee3p5a50e8a68c1b023b@mail.gmail.com>
	<i2o8076277e1004261012se5c646ear279e872b2be10db3@mail.gmail.com>
	<n2o23f91031004262025x8bead428ze2d7c0d579745d9c@mail.gmail.com>
Message-ID: <w2pb39273531004270241r1c043224i1086b2216e1d45f1@mail.gmail.com>

No (t yet).  Open a jira for enhancement... (or send a patch)

On Mon, Apr 26, 2010 at 10:25 PM, Seven Du <dujinfang at gmail.com> wrote:

> is there a timeout param for curl ?
>
> 2010/4/27 Alain MELIOT <alain.meliot at gmail.com>:
> > Hi steven
> >
> > Thank you for your answer
> > i will try
> >
> > 2010/4/26 Steven Ayre <steveayre at gmail.com>
> >>
> >> To use mod_curl to do a GET/POST request:
> >>
> >> <action application="curl"
> >> data="http://www.myhost.com/script?getName=myGetValue" />
> >> <action application="curl" data="http://www.myhost.com/script post
> >> postName=myPostValue%20a%20space" />
> >>
> >> You'd have to form a request string to match what your application
> >> expects. E.g.
> >> <action application="curl" data="http://www.myhost.com post
> >> username=${username}&ip=${network_addr}"/>
> >>
> >> You can parse the XML from a script after the curl request, you'll
> >> find the curl_response_code and curl_response variables contain the
> >> HTTP status code (200 if ok) and XML result.
> >>
> >> -Steve
> >>
> >>
> >> On 26 April 2010 17:28, Alain MELIOT <alain.meliot at gmail.com> wrote:
> >> >
> >> > Hi All
> >> >
> >> > New to freeswitch i rewrite an asterisk application to freeswitch.
> >> > I have a javascript application where i collect some user data and i
> >> > must
> >> > send  the data to a web service (wsdl) for processing.
> >> > The problem is that i have no idea  how to call the web service from
> >> > curl or
> >> > javascript.
> >> > Any help will be welcome.
> >> > Thank in advance
> >> >
> >> > Alain MELIOT
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From jjmartres at gmail.com  Tue Apr 27 05:06:47 2010
From: jjmartres at gmail.com (=?UTF-8?Q?Martr=C3=A8s_Jean=2DJacques?=)
Date: Tue, 27 Apr 2010 14:06:47 +0200
Subject: [Freeswitch-users] Freeswitch and dynamic application
Message-ID: <u2q95237e3b1004270506rc0bd580cte61730445d0b6754@mail.gmail.com>

Hi guys,

I need to build dynamic application using freeswitch.
Based on your experiences what is the best solution ?

Using a LUA script or use the event_socket ?


Thanks for your help.

Regards.

-- 
MARTRES Jean-Jacques
email : jjmartres at networkconcordance.com
website : http://www.networkconcordance.com
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From pjintheusa at gmail.com  Tue Apr 27 05:13:59 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 27 Apr 2010 08:13:59 -0400
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>
	<SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>
Message-ID: <w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>

Hi,

I have done this using scripting. managed/c# in my case. Basically all
inbound calls are passed to a script - get-session-data, This script calls a
ws and populates session variables with the result. I then pass the call
back to the dialplan for further processing,

HTH

Phil


If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me
> know how you did it.
>
> Jan
>
>
> ------------------------------
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aep.lists at it46.se  Tue Apr 27 06:07:36 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Tue, 27 Apr 2010 15:07:36 +0200
Subject: [Freeswitch-users] session.steamFile misses DTMF event for
 first few seconds
In-Reply-To: <3F7821C562CD4EC6A7C5C342BB7077A7@ws4>
References: <3F7821C562CD4EC6A7C5C342BB7077A7@ws4>
Message-ID: <b33e001e8bfb3b4d583205a643e7ecd2.squirrel@correo.nodo50.org>

How are you  receiving those DTMFs, inbound or outbound? Are those calls
coming from a mobile network (GSM). We have experienced lots of problem
with DTMF detection in noisy lines.

/aep
-- 
Stopping junk mailers is good for the environment

> I recently upgraded from FS 12790M to svn 17188.  When I did, I noticed
> that session.streamFile behaved differently and I started having
> problems with my IVR app.
>
> With the upgraded FS, I have a problem with streamFile no firing on the
> DTMF and calling the callback function for the first few seconds of the
> wav file playback.  It behaves as though it does not hear the DTMFs.  If
> I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> catches the DTMF and all is well.  If I key as soon as I hear the wav
> file start, streamFile just keeps playing the wav and does not call the
> callback function.
>
> When I revert back to the previous version of FS, streamFile always
> fires the callback right away no matter how quickly I press the first
> DTMF as the wav file starts to stream out.
>
> The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> 16 bit, mono 8000 Hz
>
> The snippet of js code I am using is as follows.
>
> if(session.ready()) {
>     session.answer();
>     session.sleep(750);
>     while(session.ready()) {
>         session.sleep(500);
>         session.flushDigits(); // clear out input buffers
>
>
> if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> ts_cb,""))===false) {
>             pin=session.getDigits(pinmax,pinterm,pinwait);
>         } else {
>             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
>         }
>         // more code here..
>     }
>
> Do I need to change the way I use streamFile in the later release?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From janvb at live.com  Tue Apr 27 07:14:36 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 16:14:36 +0200
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>,
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>,
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>,
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>,
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>,
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>,
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>,
	<SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>,
	<w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>
Message-ID: <SNT135-w680E2DE6E2A9A0709B386D2030@phx.gbl>


Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java. 

 

As a matter of interest how do you communicate with C# ?

 

---

 

One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use.

 

Jan
 


Date: Tue, 27 Apr 2010 08:13:59 -0400
From: pjintheusa at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL

Hi,

I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing,

HTH

Phil





If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.
 
Jan
 



Hotmail: Trusted email with powerful SPAM protection. Sign up 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


 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
https://signup.live.com/signup.aspx?id=60969
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From mrene_lists at avgs.ca  Tue Apr 27 07:17:38 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 27 Apr 2010 10:17:38 -0400
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <SNT135-w680E2DE6E2A9A0709B386D2030@phx.gbl>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>,
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>,
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>,
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>,
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>,
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>,
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>,
	<SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>,
	<w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>
	<SNT135-w680E2DE6E2A9A0709B386D2030@phx.gbl>
Message-ID: <614DC31D-12EA-404A-840B-70212AC4DBDC@avgs.ca>

You can use mod_managed and run managed code on top of an active call. 

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-27, at 10:14 AM, Jan Berger wrote:

> Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java. 
>  
> As a matter of interest how do you communicate with C# ?
>  
> ---
>  
> One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use.
>  
> Jan
>  
> Date: Tue, 27 Apr 2010 08:13:59 -0400
> From: pjintheusa at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
> 
> Hi,
> 
> I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing,
> 
> HTH
> 
> Phil
> 
> 
> If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.
>  
> Jan
>  
> 
> Hotmail: Trusted email with powerful SPAM protection. Sign up 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
> 
> 
> 
> Hotmail: Free, trusted and rich email service. Get it 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 freeswitch at cartissolutions.com  Tue Apr 27 02:21:45 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 27 Apr 2010 04:21:45 -0500
Subject: [Freeswitch-users] Freeswitch and dynamic application
In-Reply-To: <u2q95237e3b1004270506rc0bd580cte61730445d0b6754@mail.gmail.com>
References: <u2q95237e3b1004270506rc0bd580cte61730445d0b6754@mail.gmail.com>
Message-ID: <4BD6ACA9.7000607@cartissolutions.com>

On 4/27/2010 7:06 AM, Martr?s Jean-Jacques wrote:
> Hi guys,
>
> I need to build dynamic application using freeswitch.
> Based on your experiences what is the best solution ?
>
> Using a LUA script or use the event_socket ?

There isn't any one particular "best" way other than the way that 
accomplishes what you need in the most efficient manner.  FreeSWITCH is 
sort of like a swiss army knife in this regard - you can use LUA, 
Javascript, C, C++, managed code, etc. etc. etc.

Sure, we can give you our opinions on what we think is "best" based upon 
your actual use-case scenario...  But again, it's only general guidance 
really.

-- 
Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com



From toaltas at gmail.com  Tue Apr 27 04:21:40 2010
From: toaltas at gmail.com (joseba)
Date: Tue, 27 Apr 2010 13:21:40 +0200
Subject: [Freeswitch-users] Spanish sound  voices on freeswitch 1.0.6
Message-ID: <4BD6C8C4.7060306@gmail.com>

I see that on information about freeswitch 1.06, but I dont find this.

Only EN and RU voices.

Is this an error?

How can I find spanish voices?

Thx
JuanjoA


From anita.hall at simmortel.com  Tue Apr 27 07:43:51 2010
From: anita.hall at simmortel.com (Anita Hall)
Date: Tue, 27 Apr 2010 20:13:51 +0530
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
Message-ID: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>

Hi

Please check out this product

http://www.sangoma.com/products/hardware_products/data_networking/a301.html

Does it work on Freeswitch or Asterisk ?
Do Telcos provide an E3 connection ?

One of our customers had an inquiry for terminating 6000 calls
simultaneously. I want to do some homework before taking it further with
him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not
look feasible ?

Thanks for any input you may provide.

regards,

Anita Hall,
Simmortel Voice.
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From theis at lyth.de  Tue Apr 27 07:57:59 2010
From: theis at lyth.de (Armin Theis)
Date: Tue, 27 Apr 2010 16:57:59 +0200
Subject: [Freeswitch-users] sofia.c:896 nua_i_media_error: unknown event 22:
	988 Incomplete offer/answer
Message-ID: <4BD6FB77.7000008@lyth.de>

During a FreeSwitch call a sofia media error occured (sofia.c:896 
nua_i_media_error: unknown event 22: 988 Incomplete offer/answer)

I am using FreeSwitch Stable Release 1.0.6 and want to use it for 
production. Does anyone know why this error comes?

2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:2292 Already using PCMA
2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf 
send/recv payload to 101
2010-04-27 10:02:38.421875 [DEBUG] sofia.c:4604 Processing updated SDP
2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:2579 Audio params are 
unchanged for sofia/internal/1010 at 192.168.1.234.
2010-04-27 10:03:39.328125 [DEBUG] sofia.c:4153 Channel 
sofia/internal/1010 at 192.168.1.234 entering state [calling][0]
2010-04-27 10:03:39.484375 [DEBUG] sofia.c:896 nua_i_media_error: 
unknown event 22: 988 Incomplete offer/answer
2010-04-27 10:03:39.484375 [DEBUG] sofia.c:4153 Channel 
sofia/internal/1010 at 192.168.1.234 entering state [terminating][200]
2010-04-27 10:03:39.484375 [NOTICE] sofia.c:4789 Hangup 
sofia/internal/1010 at 192.168.1.234 [CS_EXECUTE] [NORMAL_CLEARING]
2010-04-27 10:03:39.484375 [DEBUG] switch_channel.c:2102 Send signal 
sofia/internal/1010 at 192.168.1.234 [KILL]
2010-04-27 10:03:39.484375 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1010 at 192.168.1.234 [BREAK]
2010-04-27 10:03:39.500000 [ERR] switch_rtp.c:2395 switch_rtp.c if 
(switch_rtp_ready(rtp_session))
...



From kenfulmer at icstechnologysolutions.com  Tue Apr 27 08:02:35 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 10:02:35 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
Message-ID: <00a601cae61a$ab89bf70$029d3e50$@com>

This is a trivial request but here goes:

 

We are using the User Agent parameter to replace the name "Freeswitch" with
our company name in the SIP header. How can we do the same in the SDP? 

 

Thanks,

 

Ken Fulmer

 

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From kenfulmer at icstechnologysolutions.com  Tue Apr 27 08:02:51 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 10:02:51 -0500
Subject: [Freeswitch-users] Add a diversion header?
In-Reply-To: <E261EDBC-972A-4922-A2B1-B98079A9680C@freeswitch.org>
References: <024701cae581$f48b9250$dda2b6f0$@com>
	<E261EDBC-972A-4922-A2B1-B98079A9680C@freeswitch.org>
Message-ID: <00ab01cae61a$b5a38090$20ea81b0$@com>

Thanks, that works well! 

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, April 26, 2010 4:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Add a diversion header?

 

set/export the variable sip_h_Diversion

 

/b

 

On Apr 26, 2010, at 3:49 PM, Ken Fulmer wrote:





We have customers using a sipX internal PBX and a Freeswitch device as a
softswitch to route calls to our upstream provider, PaeTec. However, the
provider doesn't support the P-Asserted Identity for external call
forwarding. PaeTec wants to see a SIP Diversion header instead.


How can we add this header in the Freeswitch device?

 

Thanks,

 

Ken Fulmer

 

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From janvb at live.com  Tue Apr 27 08:04:14 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 17:04:14 +0200
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <614DC31D-12EA-404A-840B-70212AC4DBDC@avgs.ca>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>, ,
	<7D986D04-EC3A-44D8-B4E5-451429F45890@jerris.com>, ,
	<911182.21146.qm@web76210.mail.sg1.yahoo.com>, ,
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>, ,
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>, ,
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>, ,
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>, ,
	<SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>, ,
	<w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>,
	<SNT135-w680E2DE6E2A9A0709B386D2030@phx.gbl>,
	<614DC31D-12EA-404A-840B-70212AC4DBDC@avgs.ca>
Message-ID: <SNT135-w42E5750DDE7A363FE1CD77D2030@phx.gbl>


Thanks, I will look into this one.

 


From: mrene_lists at avgs.ca
Date: Tue, 27 Apr 2010 10:17:38 -0400
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL

You can use mod_managed and run managed code on top of an active call. 




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-04-27, at 10:14 AM, Jan Berger wrote:

Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java. 
 
As a matter of interest how do you communicate with C# ?
 
---
 
One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use.
 
Jan
 


Date: Tue, 27 Apr 2010 08:13:59 -0400
From: pjintheusa at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL

Hi,

I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing,

HTH

Phil





If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.
 
Jan
 



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From brian at freeswitch.org  Tue Apr 27 08:05:15 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Apr 2010 10:05:15 -0500
Subject: [Freeswitch-users] sofia.c:896 nua_i_media_error: unknown event
	22: 988 Incomplete offer/answer
In-Reply-To: <4BD6FB77.7000008@lyth.de>
References: <4BD6FB77.7000008@lyth.de>
Message-ID: <54ACE51B-AA0B-4590-8FAE-49809F86D67B@freeswitch.org>

Well without the siptrace to go along with it we can only guess.

sofia loglevel all 9
sofia profile xxx siptrace on
(replace xxx with the profile)

do it again and pastebin the results on pastebin.freeswitch.org

/b

On Apr 27, 2010, at 9:57 AM, Armin Theis wrote:

> During a FreeSwitch call a sofia media error occured (sofia.c:896 
> nua_i_media_error: unknown event 22: 988 Incomplete offer/answer)
> 
> I am using FreeSwitch Stable Release 1.0.6 and want to use it for 
> production. Does anyone know why this error comes?



From brian at freeswitch.org  Tue Apr 27 08:05:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Apr 2010 10:05:26 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <00a601cae61a$ab89bf70$029d3e50$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
Message-ID: <F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>

Why would you wanna do that?

/b

On Apr 27, 2010, at 10:02 AM, Ken Fulmer wrote:

> This is a trivial request but here goes:
>  
> We are using the User Agent parameter to replace the name ?Freeswitch? with our company name in the SIP header. How can we do the same in the SDP?
>  
> Thanks,
>  
> Ken Fulmer
>  

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From frank at impactfax.com  Tue Apr 27 08:15:54 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 27 Apr 2010 11:15:54 -0400
Subject: [Freeswitch-users] session.steamFile misses DTMF event for
	first few seconds
In-Reply-To: <b33e001e8bfb3b4d583205a643e7ecd2.squirrel@correo.nodo50.org>
Message-ID: <788C79CD1E584493B92EB90A5EA474DB@ws4>

The calls are coming from land based lines.  Traditional POTS (not cable
company).  Caller is calling into FS and providing those DTMF.  I can
reproduce on my POTS line (3000' from CO) and there is no discernable
noise on the line. 

The call comes into the media gateway and then goes sip to FS.

I can reproduce the problem just by starting FS version from the latest
trunk.  And then I can eliminate the problem by restarting the FS
version 12790.  In both test cases, the media gateway remains constant.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Alberto Escudero
Sent: Tuesday, April 27, 2010 9:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
first few seconds

How are you  receiving those DTMFs, inbound or outbound? Are those calls
coming from a mobile network (GSM). We have experienced lots of problem
with DTMF detection in noisy lines.

/aep
-- 
Stopping junk mailers is good for the environment

> I recently upgraded from FS 12790M to svn 17188.  When I did, I
noticed
> that session.streamFile behaved differently and I started having
> problems with my IVR app.
>
> With the upgraded FS, I have a problem with streamFile no firing on
the
> DTMF and calling the callback function for the first few seconds of
the
> wav file playback.  It behaves as though it does not hear the DTMFs.
If
> I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> catches the DTMF and all is well.  If I key as soon as I hear the wav
> file start, streamFile just keeps playing the wav and does not call
the
> callback function.
>
> When I revert back to the previous version of FS, streamFile always
> fires the callback right away no matter how quickly I press the first
> DTMF as the wav file starts to stream out.
>
> The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> 16 bit, mono 8000 Hz
>
> The snippet of js code I am using is as follows.
>
> if(session.ready()) {
>     session.answer();
>     session.sleep(750);
>     while(session.ready()) {
>         session.sleep(500);
>         session.flushDigits(); // clear out input buffers
>
>
>
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> ts_cb,""))===false) {
>             pin=session.getDigits(pinmax,pinterm,pinwait);
>         } else {
>             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
>         }
>         // more code here..
>     }
>
> Do I need to change the way I use streamFile in the later release?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://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 david.ponzone at gmail.com  Tue Apr 27 08:16:28 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 27 Apr 2010 17:16:28 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
Message-ID: <BCEB4421-46EB-4C9B-A30E-90C3B3FCDC8F@gmail.com>

Anita,

this card is a data card.
It does not provide ISDN protocol.

For 6000 calls, you should better find a telco that provides SIP  
trunking over Fast/GigaEthernet.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/04/2010 ? 16:43, Anita Hall a ?crit :

> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Freeswitch or Asterisk ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls  
> simultaneously. I want to do some homework before taking it further  
> with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines,  
> which does not look feasible ?
>
> Thanks for any input you may provide.
>
> regards,
>
> Anita Hall,
> Simmortel Voice.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Tue Apr 27 08:19:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 27 Apr 2010 17:19:16 +0200
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <00a601cae61a$ab89bf70$029d3e50$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
Message-ID: <2CA14D9E-8F83-4BAE-8137-AE0164492EE0@gmail.com>

As far as I know, you can't.
You would need to patch the source and recompile.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 27/04/2010 ? 17:02, Ken Fulmer a ?crit :

> This is a trivial request but here goes:
>
> We are using the User Agent parameter to replace the name  
> ?Freeswitch? with our company name in the SIP header. How can we  
> do the same in the SDP?
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 27 08:30:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 10:30:17 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <2CA14D9E-8F83-4BAE-8137-AE0164492EE0@gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<2CA14D9E-8F83-4BAE-8137-AE0164492EE0@gmail.com>
Message-ID: <y2j191c3a031004270830w86daa04didae917f5cbdbc036@mail.gmail.com>

Messing with SDP like that is not wise since it makes it harder for media
devices to recognize each other.  Do you have a particular reason to hide in
the FS closet?


2010/4/27 David Ponzone <david.ponzone at gmail.com>

> As far as I know, you can't.
> You would need to patch the source and recompile.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 27/04/2010 ? 17:02, Ken Fulmer a ?crit :
>
> This is a trivial request but here goes:
>
> We are using the User Agent parameter to replace the name ?Freeswitch? with
> our company name in the SIP header. How can we do the same in the SDP?
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Apr 27 08:40:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 10:40:15 -0500
Subject: [Freeswitch-users] session.steamFile misses DTMF event for
	first few seconds
In-Reply-To: <788C79CD1E584493B92EB90A5EA474DB@ws4>
References: <b33e001e8bfb3b4d583205a643e7ecd2.squirrel@correo.nodo50.org>
	<788C79CD1E584493B92EB90A5EA474DB@ws4>
Message-ID: <u2r191c3a031004270840o6948472blfa90e9ca0a238914@mail.gmail.com>

12XXX is so many years old, i wish users who want free help would at least
stay up to date with the code.

FYI, our repo is on git now and the svn mirror is not updating at the
moment.

produce a complete minimal script that reproduces your problem and can be
run on git HEAD (see download instructions to learn how to build with git)
Use existing sound files from the FS install so we can just run it in our
lab to reproduce the issue.

Open an issue on http://jira.freeswitch.org and attach the script.



On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank at impactfax.com>wrote:

> The calls are coming from land based lines.  Traditional POTS (not cable
> company).  Caller is calling into FS and providing those DTMF.  I can
> reproduce on my POTS line (3000' from CO) and there is no discernable
> noise on the line.
>
> The call comes into the media gateway and then goes sip to FS.
>
> I can reproduce the problem just by starting FS version from the latest
> trunk.  And then I can eliminate the problem by restarting the FS
> version 12790.  In both test cases, the media gateway remains constant.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Alberto Escudero
> Sent: Tuesday, April 27, 2010 9:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
> first few seconds
>
> How are you  receiving those DTMFs, inbound or outbound? Are those calls
> coming from a mobile network (GSM). We have experienced lots of problem
> with DTMF detection in noisy lines.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > I recently upgraded from FS 12790M to svn 17188.  When I did, I
> noticed
> > that session.streamFile behaved differently and I started having
> > problems with my IVR app.
> >
> > With the upgraded FS, I have a problem with streamFile no firing on
> the
> > DTMF and calling the callback function for the first few seconds of
> the
> > wav file playback.  It behaves as though it does not hear the DTMFs.
> If
> > I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> > catches the DTMF and all is well.  If I key as soon as I hear the wav
> > file start, streamFile just keeps playing the wav and does not call
> the
> > callback function.
> >
> > When I revert back to the previous version of FS, streamFile always
> > fires the callback right away no matter how quickly I press the first
> > DTMF as the wav file starts to stream out.
> >
> > The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> > 16 bit, mono 8000 Hz
> >
> > The snippet of js code I am using is as follows.
> >
> > if(session.ready()) {
> >     session.answer();
> >     session.sleep(750);
> >     while(session.ready()) {
> >         session.sleep(500);
> >         session.flushDigits(); // clear out input buffers
> >
> >
> >
> if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> > ts_cb,""))===false) {
> >             pin=session.getDigits(pinmax,pinterm,pinwait);
> >         } else {
> >             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
> >         }
> >         // more code here..
> >     }
> >
> > Do I need to change the way I use streamFile in the later release?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Apr 27 08:42:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 10:42:30 -0500
Subject: [Freeswitch-users] sofia.c:896 nua_i_media_error: unknown event
	22: 988 Incomplete offer/answer
In-Reply-To: <54ACE51B-AA0B-4590-8FAE-49809F86D67B@freeswitch.org>
References: <4BD6FB77.7000008@lyth.de>
	<54ACE51B-AA0B-4590-8FAE-49809F86D67B@freeswitch.org>
Message-ID: <x2o191c3a031004270842lc274baa5t56a7be74f12f388c@mail.gmail.com>

2010-04-27 10:03:39.484375 [DEBUG] sofia.c:896 nua_i_media_error:
unknown event 22: 988 Incomplete offer/answer

That means the other side of the call sent an invalid SDP.

Do as Brian says and get a sip trace so you can see it.



On Tue, Apr 27, 2010 at 10:05 AM, Brian West <brian at freeswitch.org> wrote:

> Well without the siptrace to go along with it we can only guess.
>
> sofia loglevel all 9
> sofia profile xxx siptrace on
> (replace xxx with the profile)
>
> do it again and pastebin the results on pastebin.freeswitch.org
>
> /b
>
> On Apr 27, 2010, at 9:57 AM, Armin Theis wrote:
>
> > During a FreeSwitch call a sofia media error occured (sofia.c:896
> > nua_i_media_error: unknown event 22: 988 Incomplete offer/answer)
> >
> > I am using FreeSwitch Stable Release 1.0.6 and want to use it for
> > production. Does anyone know why this error comes?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Apr 27 08:47:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 10:47:11 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
Message-ID: <i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>

There is no sign of a REFER packet or anything indicating an attended
transfer in this trace.
Did you send the wrong one possibly?


On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:

> Here is the setup that was used to reproduce the consultative transfer
> problem.
>
> There are two boxes, the first is running a proxy based PBX (sipXecs) and
> the
> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones,
> extension
> 200 and 202, registered with it.  The PBX has configured a mapping rule
> which
> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060which
> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
> configured to allow connections from the PBX box via an ACL configuration
> and
> the public dialplan includes an "echo" extension:
>
>     <extension name="echo">
>       <condition field="destination_number" expression="^9996$">
>         <action application="answer"/>
>         <action application="echo"/>
>       </condition>
>     </extension>
>
> Any of the phones registered with the PBX can dial extension 9996 and be
> connected to the FreeSWITCH echo application.  But when one phone attempts
> to transfer another phone to extension 9996 via a consultative transfer,
> FreeSWITCH does not properly complete the transfer.  You can see in the log
> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
> initiated the transfer.  Without that terminating BYE, the transfer
> controller thinks that the transfer failed.
>
> The corresponding FreeSWITCH log file -
> http://pastebin.freeswitch.org/12806
>
> If it will help, I can also forward a corresponding PCAP file.
>
> Thanks
>
> -Mardy
>
> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>
> I'm trying to understand this:
>
> If FS is acting as a phone in your scenario why are you sending a refer to
> it and not the server?
> In most situations there is a b2bua server who routes the calls and takes
> all the REFER.
> Is this one of those PROXY only sip servers?
>
> I think you would need to produce a full debug log of this, and if you are
> using some kind of proxy based setup we would need some way to easily
> reproduce it or visit your lab because we do not typically use anything of
> the sort.
>
> Execute these commands and reproduce it and capture the whole log and put
> it on
> http://pastebin.freeswitch.org
>
> sofia profile internal siptrace on
> console loglevel debug
>
>
>
>
>
> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> instead of emailing again when impatient for an answer (something we frown
>> upon here in this busy list)
>> produce a reproducible step by step process to duplicate your issue.  We
>> are trying to help people but we don't have the time to do the leg work for
>> everyone who asks a question when we get hundreds of emails a day.
>>
>>
>>
>>
>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>
>>> Just following up...  Does anyone have any suggestions on how to proceed
>>> with this?  I've run out of ideas.
>>>
>>> Thanks,
>>>
>>> -Mardy
>>>
>>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>>
>>> The phones that I am using are not registered with FS.  They are
>>> registered with another proxy based PBX.  I am simply using FS as B2BUA
>>> which is also registered with the PBX.  And yes, I can successfully transfer
>>> a call to another phone with this setup.
>>>
>>> To simplify things I tried the same scenario using FSComm in place of my
>>> own FS application and tried to transfer a call to FSComm with the same
>>> results.  And just in case there might be a problem specific to FSComm, I
>>> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
>>> echo application with the same results.  By the way, I have no problems with
>>> blind transfers, only attended transfers.
>>>
>>> -Mardy
>>>
>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>>
>>> did you try just setting up 2 phones on plain fresh FS install, and
>>> calling them normally and transferring them around?
>>> That description is still pretty vague? What is an Event Socket
>>> application, which has nothing to do with sip and sip transfers, that's a FS
>>> protocol.
>>>
>>>
>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>
>>>> I have two phones (Polycom) and an event_socket application, all of
>>>> which are using a SIP proxy for call routing.  The first phone calls the
>>>> second phone.  The second phone then attempts to transfer the call to the
>>>> FS/event_socket application by first placing the call on hold and then
>>>> calling the FS application, followed by a consultative transfer.  The REFER
>>>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>>>  The transferred call leg appears to be answered by FS and the application
>>>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>>>> that I see is that the original call leg, created when the user called the
>>>> FS application to announce the transfer, does not get canceled by FS and
>>>> subsequently does not send the BYE back to the Polycom.  Is there something
>>>> that I need to do at the event_socket application to complete the transfer?
>>>>  I've tried killing the UUID associated with the first call leg as well as
>>>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>>
>>>> -Mardy
>>>>
>>>>
>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>>
>>>> but what is the client sending the REFER?
>>>>
>>>> FS gets refer+replaces all the time, if it's the one where the dest is
>>>> on another box (aka the nightmare xfer that you should see references to in
>>>> the debug log if so) then it will not complete until that far end call is
>>>> answered.
>>>>
>>>> FS handles this scenerio for us hundreds of times a day using a wide
>>>> range of sip devices so perhaps
>>>> your UA has an interop problem.
>>>>
>>>>
>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>
>>>>>
>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>>
>>>>> uuid_simplify will issue the refer...
>>>>>
>>>>>
>>>>> I looked at uuid_simplify and if I understand it correctly it is for
>>>>> use when one wants to act as the transfer controller.  In my case, FS is the
>>>>> transfer destination.  Another phone has already generated the refer and FS
>>>>> has been sent an invite with replaces.
>>>>>
>>>>>
>>>>> May I ask what application you are developing?
>>>>>
>>>>>
>>>>> An ACD.
>>>>>
>>>>>
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>> FSComm developer
>>>>>
>>>>>
>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>
>>>>>> I'm having a problem with attended transfers where the destination of
>>>>>> the transfer is a FreeSWITCH based application such as FSComm.  (It should
>>>>>> be noted that in my setup the phone performing the transfer and the caller
>>>>>> which is being transferred are parties of another SIP server.)  What I see,
>>>>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>>>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>>>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>>>> application side, FS still thinks that both the initial call leg and
>>>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>>>> kill all legs of the call.  Is there a specific action that the application
>>>>>> must take in order for the transfer to complete?
>>>>>>
>>>>>> -Mardy
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ranjtech at gmail.com  Tue Apr 27 08:54:45 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 27 Apr 2010 11:54:45 -0400
Subject: [Freeswitch-users] understanding effective_caller_id_number and
	other variables before bridge
Message-ID: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>

Hello All,

just want to clarify my understanding re: variables etc that can be set
before a call bridge/transfer etc. It appears that we need to set these
variables immediately before the bridge without any other operation
occurring from when they are setup and when the bridge happens. Is that
true?

I have a very simple dialplan like so:

<include>
  <extension name="DIDtest">
    <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/>
    <action application="export" data="effective_caller_id_number=$1"/>
    <condition field="destination_number" expression="^\+?+1?1?(.*)$">
           <action application="bridge"
data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
     </condition>
  </extension>
</include>

But when the call is sent to gateway blade2, the callerID on the call is
whatever the original callerID was with all the 1s and '+'s and whatever
prefixes the call came in with.

Questions:
a) can the variable effective_caller_id_number not be set dynamically from a
temp variable such as $1. I have also tried forcing it to a number
like effective_caller_id_number=20239388383 and even then it doesn't work

 b) I have also tried

<action application="bridge"
data="{effective_caller_id_number=20239388383}{sip_invite_domain=${sip_from_host}}sofia/gateway/blade9/$1"/>

and that doesn't work either.

Any ideas?

Thanks
RR
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From mardy at voysys.com  Tue Apr 27 09:03:08 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Tue, 27 Apr 2010 12:03:08 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<461667B8-6BD2-428C-9C75-EDC38F7E9704@voysys.com>
	<g2n191c3a031004191519i5182a21fi86421df96bcec2e3@mail.gmail.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
	<i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
Message-ID: <D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>

The REFER exchange occurs on the other side of the proxy between the two Polycom phones.  Since
FreeSWITCH is the destination of the transfer, it will only see the INVITE w/Replaces.  I've sent you the
PCAP file which captures the complete SIP exchange.  That should help to clarify.

-Mardy

On Apr 27, 2010, at 11:47 AM, Anthony Minessale wrote:

> There is no sign of a REFER packet or anything indicating an attended transfer in this trace.
> Did you send the wrong one possibly?
> 
> 
> On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:
> Here is the setup that was used to reproduce the consultative transfer problem.
> 
> There are two boxes, the first is running a proxy based PBX (sipXecs) and the
> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones, extension
> 200 and 202, registered with it.  The PBX has configured a mapping rule which
> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060 which
> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
> configured to allow connections from the PBX box via an ACL configuration and
> the public dialplan includes an "echo" extension:
> 
>     <extension name="echo">
>       <condition field="destination_number" expression="^9996$">
>         <action application="answer"/>
>         <action application="echo"/>
>       </condition>
>     </extension>
> 
> Any of the phones registered with the PBX can dial extension 9996 and be
> connected to the FreeSWITCH echo application.  But when one phone attempts
> to transfer another phone to extension 9996 via a consultative transfer,
> FreeSWITCH does not properly complete the transfer.  You can see in the log
> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
> initiated the transfer.  Without that terminating BYE, the transfer
> controller thinks that the transfer failed.
> 
> The corresponding FreeSWITCH log file - http://pastebin.freeswitch.org/12806
> 
> If it will help, I can also forward a corresponding PCAP file.
> 
> Thanks
> 
> -Mardy
> 
> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
> 
>> I'm trying to understand this:
>> 
>> If FS is acting as a phone in your scenario why are you sending a refer to it and not the server?
>> In most situations there is a b2bua server who routes the calls and takes all the REFER.
>> Is this one of those PROXY only sip servers?
>> 
>> I think you would need to produce a full debug log of this, and if you are using some kind of proxy based setup we would need some way to easily reproduce it or visit your lab because we do not typically use anything of the sort.
>> 
>> Execute these commands and reproduce it and capture the whole log and put it on
>> http://pastebin.freeswitch.org
>> 
>> sofia profile internal siptrace on
>> console loglevel debug
>> 
>> 
>> 
>> 
>> 
>> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> instead of emailing again when impatient for an answer (something we frown upon here in this busy list) 
>> produce a reproducible step by step process to duplicate your issue.  We are trying to help people but we don't have the time to do the leg work for everyone who asks a question when we get hundreds of emails a day.
>> 
>> 
>> 
>> 
>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>> Just following up...  Does anyone have any suggestions on how to proceed with this?  I've run out of ideas.
>> 
>> Thanks,
>> 
>> -Mardy
>> 
>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>> 
>>> The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.
>>> 
>>> To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.
>>> 
>>> -Mardy
>>> 
>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>> 
>>>> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
>>>> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
>>>> 
>>>> 
>>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>> 
>>>> -Mardy
>>>> 
>>>> 
>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>> 
>>>>> but what is the client sending the REFER?
>>>>> 
>>>>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>>>>> 
>>>>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>>>>> your UA has an interop problem.
>>>>> 
>>>>> 
>>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>> 
>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>> 
>>>>>> uuid_simplify will issue the refer...
>>>>> 
>>>>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>>>>> 
>>>>>> 
>>>>>> May I ask what application you are developing?
>>>>> 
>>>>> An ACD.
>>>>> 
>>>>>> 
>>>>>> Regards,
>>>>>> Jo?o Mesquita
>>>>>> FSComm developer
>>>>>> 
>>>>>> 
>>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>>>>> 
>>>>>> -Mardy
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> Anthony Minessale II
>>>>> 
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> 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
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> -- 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> 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
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kenfulmer at icstechnologysolutions.com  Tue Apr 27 09:06:36 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 11:06:36 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
Message-ID: <00fb01cae623$9f364be0$dda2e3a0$@com>

It's not meant to be offensive or anything. We'd just like to ensure the
competition doesn't know what we're doing for as long as possible. That's
all. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, April 27, 2010 10:05 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

 

Why would you wanna do that?

 

/b

 

On Apr 27, 2010, at 10:02 AM, Ken Fulmer wrote:





This is a trivial request but here goes:

 

We are using the User Agent parameter to replace the name "Freeswitch" with
our company name in the SIP header. How can we do the same in the SDP?

 

Thanks,

 

Ken Fulmer

 

 

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From yehavi.bourvine at gmail.com  Tue Apr 27 09:10:12 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 27 Apr 2010 19:10:12 +0300
Subject: [Freeswitch-users] What to do after 1.0.6 has been released?
Message-ID: <q2qd361ca7b1004270910j228bcc23m121e0a1e60443cc7@mail.gmail.com>

Hello,

  I would like to know what other people are doing in this situation: We've
upgraded to 1.0.6 which is very solid. At the last 2-3 months before 1.0.6
has been released I did a weekely upgrade to the latest trunk.

  What to do now? The stability is very important, but I do not want to open
too large gap from 1.0.6 to the deveopment version. What do you suggest?

                                 Thanks! __Yehavi:
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From david.ponzone at gmail.com  Tue Apr 27 09:16:44 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 27 Apr 2010 18:16:44 +0200
Subject: [Freeswitch-users] understanding effective_caller_id_number and
	other variables before bridge
In-Reply-To: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>
References: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>
Message-ID: <8969B9C0-29EB-4B4E-8EC7-5EF1BFAFC951@gmail.com>

You should first correct your dialplan to:

<include>
   <extension name="DIDtest">
     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"  
break="never">
            <action application="set" data="effective_caller_id_number= 
$1"/>
     </condition>
     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
            <action application="bridge" data="{sip_invite_domain=$ 
{sip_from_host}}sofia/gateway/blade2/$1"/>
      </condition>
   </extension>
</include>


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/04/2010 ? 17:54, RR a ?crit :

> Hello All,
>
> just want to clarify my understanding re: variables etc that can be  
> set before a call bridge/transfer etc. It appears that we need to  
> set these variables immediately before the bridge without any other  
> operation occurring from when they are setup and when the bridge  
> happens. Is that true?
>
> I have a very simple dialplan like so:
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/>
>     <action application="export" data="effective_caller_id_number= 
> $1"/>
>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>            <action application="bridge" data="{sip_invite_domain=$ 
> {sip_from_host}}sofia/gateway/blade2/$1"/>
>      </condition>
>   </extension>
> </include>
>
> But when the call is sent to gateway blade2, the callerID on the  
> call is whatever the original callerID was with all the 1s and '+'s  
> and whatever prefixes the call came in with.
>
> Questions:
> a) can the variable effective_caller_id_number not be set  
> dynamically from a temp variable such as $1. I have also tried  
> forcing it to a number like effective_caller_id_number=20239388383  
> and even then it doesn't work
>
>  b) I have also tried
>
> <action application="bridge"  
> data="{effective_caller_id_number=20239388383}{sip_invite_domain=$ 
> {sip_from_host}}sofia/gateway/blade9/$1"/>
>
> and that doesn't work either.
>
> Any ideas?
>
> Thanks
> RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From janvb at live.com  Tue Apr 27 09:36:04 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 18:36:04 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
Message-ID: <SNT135-w2002D7E129BEFFEFF297B1D2030@phx.gbl>


hi Anita,

 

I think the answer differ from telco to telco, but it's in their interest to provide solutions to volum customers. I will recommend you ask for quotes and compare raw SIP, E1''s, E3's and STM-1's.

 

A major question is how do you plan interfacing to the telco? Will you be using Q.931, Q.SIG, SS7 or something else? 

 

FreeSWITCH "as is" can't support this card, but it can be made to do so. I assume Sangoma delivers a L1/L2 driver ? In which case I easely can tweak both L2 and L3 to fit. I have to make a few assumtions here because I have no knowledge about how smart Sangoma have been on the PCI interface/driver side of this card.

 

Jan



Date: Tue, 27 Apr 2010 20:13:51 +0530
From: anita.hall at simmortel.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] E3 Card on Freeswitch ?

Hi



Please check out this product

http://www.sangoma.com/products/hardware_products/data_networking/a301.html

Does it work on Freeswitch or Asterisk ?
Do Telcos provide an E3 connection ?

One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ?

Thanks for any input you may provide.

regards,

Anita Hall,
Simmortel Voice.

 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From ranjtech at gmail.com  Tue Apr 27 09:41:40 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 27 Apr 2010 12:41:40 -0400
Subject: [Freeswitch-users] understanding effective_caller_id_number and
	other variables before bridge
In-Reply-To: <8969B9C0-29EB-4B4E-8EC7-5EF1BFAFC951@gmail.com>
References: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>
	<8969B9C0-29EB-4B4E-8EC7-5EF1BFAFC951@gmail.com>
Message-ID: <v2xdcde51f21004270941g343c0670kcdaf09ac8c369bb3@mail.gmail.com>

Thanks David.

Unfortunately, that didn't fix the issue. I still see the original 'ani'
going out with all the prefixes attached to the actual number :(

The only time it works is if I set the effective_caller_id_number JUST
before the bridge. Otherwise it seems to get lost.

On Tue, Apr 27, 2010 at 12:16 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> You should first correct your dialplan to:
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$" break="never">
>            <action application="set" data="effective_caller_id_number=$1"/>
>     </condition>
>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>            <action application="bridge"
> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>      </condition>
>   </extension>
> </include>
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 27/04/2010 ? 17:54, RR a ?crit :
>
> Hello All,
>
> just want to clarify my understanding re: variables etc that can be set
> before a call bridge/transfer etc. It appears that we need to set these
> variables immediately before the bridge without any other operation
> occurring from when they are setup and when the bridge happens. Is that
> true?
>
> I have a very simple dialplan like so:
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/>
>     <action application="export" data="effective_caller_id_number=$1"/>
>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>            <action application="bridge"
> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>      </condition>
>   </extension>
> </include>
>
> But when the call is sent to gateway blade2, the callerID on the call is
> whatever the original callerID was with all the 1s and '+'s and whatever
> prefixes the call came in with.
>
> Questions:
> a) can the variable effective_caller_id_number not be set dynamically from
> a temp variable such as $1. I have also tried forcing it to a number
> like effective_caller_id_number=20239388383 and even then it doesn't work
>
>  b) I have also tried
>
> <action application="bridge"
> data="{effective_caller_id_number=20239388383}{sip_invite_domain=${sip_from_host}}sofia/gateway/blade9/$1"/>
>
> and that doesn't work either.
>
> Any ideas?
>
> Thanks
> RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Apr 27 09:53:39 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 27 Apr 2010 09:53:39 -0700
Subject: [Freeswitch-users] understanding effective_caller_id_number and
	other variables before bridge
In-Reply-To: <v2xdcde51f21004270941g343c0670kcdaf09ac8c369bb3@mail.gmail.com>
References: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>
	<8969B9C0-29EB-4B4E-8EC7-5EF1BFAFC951@gmail.com>
	<v2xdcde51f21004270941g343c0670kcdaf09ac8c369bb3@mail.gmail.com>
Message-ID: <EC0AB824-E6E6-4B32-8C47-E9875C163150@tlainvestments.com>

It looks like your regex is wrong.  Try the following:

^(\+?|\+1?|1?)(\d+).*$

and substitute $2 where you have $1 in the effective_caller_id_number action.

i.e.

<include>
  <extension name="DIDtest">
    <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$" break="never">
           <action application="set" data="effective_caller_id_number=$2"/>
    </condition>
    <condition field="destination_number" expression="^\+?+1?1?(.*)$">
           <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
     </condition>
  </extension>
</include>

-Troy


On Apr 27, 2010, at 9:41 AM, RR wrote:

> Thanks David.
> 
> Unfortunately, that didn't fix the issue. I still see the original 'ani' going out with all the prefixes attached to the actual number :(
> 
> The only time it works is if I set the effective_caller_id_number JUST before the bridge. Otherwise it seems to get lost. 
> 
> On Tue, Apr 27, 2010 at 12:16 PM, David Ponzone <david.ponzone at gmail.com> wrote:
> You should first correct your dialplan to:
> 
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$" break="never">
>            <action application="set" data="effective_caller_id_number=$1"/>
>     </condition>
>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>            <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>      </condition>
>   </extension>
> </include>
> 
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 27/04/2010 ? 17:54, RR a ?crit :
> 
>> Hello All,
>> 
>> just want to clarify my understanding re: variables etc that can be set before a call bridge/transfer etc. It appears that we need to set these variables immediately before the bridge without any other operation occurring from when they are setup and when the bridge happens. Is that true?
>> 
>> I have a very simple dialplan like so:
>> 
>> <include>
>>   <extension name="DIDtest">
>>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/>
>>     <action application="export" data="effective_caller_id_number=$1"/>
>>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>>            <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>>      </condition>
>>   </extension>
>> </include>
>> 
>> But when the call is sent to gateway blade2, the callerID on the call is whatever the original callerID was with all the 1s and '+'s and whatever prefixes the call came in with. 
>> 
>> Questions:
>> a) can the variable effective_caller_id_number not be set dynamically from a temp variable such as $1. I have also tried forcing it to a number like effective_caller_id_number=20239388383 and even then it doesn't work
>> 
>>  b) I have also tried 
>> 
>> <action application="bridge" data="{effective_caller_id_number=20239388383}{sip_invite_domain=${sip_from_host}}sofia/gateway/blade9/$1"/>
>> 
>> and that doesn't work either.
>> 
>> Any ideas?
>> 
>> Thanks
>> RR
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Tue Apr 27 10:00:16 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 27 Apr 2010 13:00:16 -0400
Subject: [Freeswitch-users] What to do after 1.0.6 has been released?
In-Reply-To: <q2qd361ca7b1004270910j228bcc23m121e0a1e60443cc7@mail.gmail.com>
References: <q2qd361ca7b1004270910j228bcc23m121e0a1e60443cc7@mail.gmail.com>
Message-ID: <r2o6b65470d1004271000if1f6a745i1c2d70b915522912@mail.gmail.com>

I usually tend to keep tabs on the changes committed to the tree since the
version I'm running looking in particular for changes that effect core or
modules I use. Upgrading production systems isn't something to take lightly
so I try to test the version in a lab first before pushing to production.

Knock on wood I haven't had any major issues since any issues found in the
source tree tend to be fixed quickly when reported with enough detail.

-- W
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From steveu at coppice.org  Tue Apr 27 10:28:46 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 28 Apr 2010 01:28:46 +0800
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
Message-ID: <4BD71ECE.60003@coppice.org>

On 04/27/2010 10:43 PM, Anita Hall wrote:
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Freeswitch or Asterisk ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls 
> simultaneously. I want to do some homework before taking it further 
> with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, 
> which does not look feasible ?
>
> Thanks for any input you may provide.
>
> regards,
>
That card only operates as an E3 or T3 data card. It is not channelised to work as a PSTN voice card. So, no, it doesn't work with Asterisk or Freeswitch - unless you are looking for IP connecion for VoIP.

Sangoma used to talk about producing a channelised revision of the card, but it looks like the potential sales have never looked promising enough to make it happen.

Digium used to have an E3/T3 card on their web site, but I assume they canned it. You don't hear anything about it now.

I never found an E3/T3 card which is properly channelised to act as a PSTN interface for a package like Freeswitch or Asterisk.

A lot of people will try to tell you that you really wouldn't want to handle so many calls in one box, but that is a rather odd thing to say these days. Servers are much faster than when Asterisk started, and the goal posts have moved. People handle this many VoIP calls on a single box these days. If suitable E3/T3 hardware were available, a modern server could be constructed to handle this number of PSTN calls.

Steve




From kris at kriskinc.com  Tue Apr 27 10:37:23 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 27 Apr 2010 13:37:23 -0400
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <00fb01cae623$9f364be0$dda2e3a0$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
Message-ID: <r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real
name and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> It?s not meant to be offensive or anything. We?d just like to ensure the
> competition doesn?t know what we?re doing for as long as possible. That?s
> all.
>
>
>
> Ken

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


From msc at freeswitch.org  Tue Apr 27 11:21:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Apr 2010 11:21:18 -0700
Subject: [Freeswitch-users] Spanish sound voices on freeswitch 1.0.6
In-Reply-To: <4BD6C8C4.7060306@gmail.com>
References: <4BD6C8C4.7060306@gmail.com>
Message-ID: <x2p87f2f3b91004271121m4235180ex1e1cdc1a5843609c@mail.gmail.com>

We've had a few volunteers talk about the spanish files but nothing is
finalized. If someone has a set of sound files for the community to review
then by all means contact me and I will put the package up where people can
download it and test them.
-MC

On Tue, Apr 27, 2010 at 4:21 AM, joseba <toaltas at gmail.com> wrote:

> I see that on information about freeswitch 1.06, but I dont find this.
>
> Only EN and RU voices.
>
> Is this an error?
>
> How can I find spanish voices?
>
> Thx
> JuanjoA
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ranjtech at gmail.com  Tue Apr 27 11:28:09 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 27 Apr 2010 14:28:09 -0400
Subject: [Freeswitch-users] understanding effective_caller_id_number and
	other variables before bridge
In-Reply-To: <EC0AB824-E6E6-4B32-8C47-E9875C163150@tlainvestments.com>
References: <q2ydcde51f21004270854p5078bb1cncb90c39292bccce5@mail.gmail.com>
	<8969B9C0-29EB-4B4E-8EC7-5EF1BFAFC951@gmail.com>
	<v2xdcde51f21004270941g343c0670kcdaf09ac8c369bb3@mail.gmail.com>
	<EC0AB824-E6E6-4B32-8C47-E9875C163150@tlainvestments.com>
Message-ID: <j2idcde51f21004271128xfd68b522k51e1bb893863dcab@mail.gmail.com>

Thanks Troy,

that seems to have worked.

Great!

\RR

On Tue, Apr 27, 2010 at 12:53 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> It looks like your regex is wrong.  Try the following:
>
> ^(\+?|\+1?|1?)(\d+).*$
>
> and substitute $2 where you have $1 in the effective_caller_id_number
> action.
>
> i.e.
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
> break="never">
>            <action application="set" data="effective_caller_id_number=$2"/>
>
>     </condition>
>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>            <action application="bridge"
> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>      </condition>
>   </extension>
> </include>
>
> -Troy
>
>
> On Apr 27, 2010, at 9:41 AM, RR wrote:
>
> Thanks David.
>
> Unfortunately, that didn't fix the issue. I still see the original 'ani'
> going out with all the prefixes attached to the actual number :(
>
> The only time it works is if I set the effective_caller_id_number JUST
> before the bridge. Otherwise it seems to get lost.
>
> On Tue, Apr 27, 2010 at 12:16 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> You should first correct your dialplan to:
>>
>> <include>
>>   <extension name="DIDtest">
>>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$" break="never">
>>            <action application="set"
>> data="effective_caller_id_number=$1"/>
>>     </condition>
>>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>>            <action application="bridge"
>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>>      </condition>
>>   </extension>
>> </include>
>>
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 27/04/2010 ? 17:54, RR a ?crit :
>>
>> Hello All,
>>
>> just want to clarify my understanding re: variables etc that can be set
>> before a call bridge/transfer etc. It appears that we need to set these
>> variables immediately before the bridge without any other operation
>> occurring from when they are setup and when the bridge happens. Is that
>> true?
>>
>> I have a very simple dialplan like so:
>>
>> <include>
>>   <extension name="DIDtest">
>>     <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/>
>>     <action application="export" data="effective_caller_id_number=$1"/>
>>     <condition field="destination_number" expression="^\+?+1?1?(.*)$">
>>            <action application="bridge"
>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/>
>>      </condition>
>>   </extension>
>> </include>
>>
>> But when the call is sent to gateway blade2, the callerID on the call is
>> whatever the original callerID was with all the 1s and '+'s and whatever
>> prefixes the call came in with.
>>
>> Questions:
>> a) can the variable effective_caller_id_number not be set dynamically from
>> a temp variable such as $1. I have also tried forcing it to a number
>> like effective_caller_id_number=20239388383 and even then it doesn't work
>>
>>  b) I have also tried
>>
>> <action application="bridge"
>> data="{effective_caller_id_number=20239388383}{sip_invite_domain=${sip_from_host}}sofia/gateway/blade9/$1"/>
>>
>> and that doesn't work either.
>>
>> Any ideas?
>>
>> Thanks
>> RR
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Apr 27 11:34:20 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 27 Apr 2010 14:34:20 -0400
Subject: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
In-Reply-To: <SNT135-w42E5750DDE7A363FE1CD77D2030@phx.gbl>
References: <879420.3757.qm@web76215.mail.sg1.yahoo.com>
	<1050429E-17A7-401B-B19A-DD9BDCA19D8B@jerris.com>
	<166310.44652.qm@web76201.mail.sg1.yahoo.com>
	<2EEACAF9-76CA-44E3-BD0F-49C7015BB5CC@jerris.com>
	<59AD9EC6-630E-491E-A6AF-58BFECBAFC06@halo2.pl>
	<SNT135-w707DDD6206A4B20389AEAD2030@phx.gbl>
	<w2u367751821004270513kfa5ef502o6ad931eb808281c8@mail.gmail.com>
	<SNT135-w680E2DE6E2A9A0709B386D2030@phx.gbl>
	<614DC31D-12EA-404A-840B-70212AC4DBDC@avgs.ca>
	<SNT135-w42E5750DDE7A363FE1CD77D2030@phx.gbl>
Message-ID: <k2p367751821004271134u296e2b4bh6fefa07f0e224b@mail.gmail.com>

You might also consider the event socket (just search esl in the wiki)

I am toying with this as it shields us from any future changes that break
mod_managed.

I really like mod_managed though and for me it works great so far.

On Tue, Apr 27, 2010 at 11:04 AM, Jan Berger <janvb at live.com> wrote:

>  Thanks, I will look into this one.
>
> ------------------------------
> From: mrene_lists at avgs.ca
> Date: Tue, 27 Apr 2010 10:17:38 -0400
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
>
> You can use mod_managed and run managed code on top of an active call.
>
>  Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  On 2010-04-27, at 10:14 AM, Jan Berger wrote:
>
>  Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or
> Java.
>
> As a matter of interest how do you communicate with C# ?
>
> ---
>
> One of the alternatives I am toying with is to use gSOAP to expose
> FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the
> issues I see with using FS in real projects. Have not looked into this, and
> need to see if there is any standards I can use.
>
> Jan
>
> ------------------------------
> Date: Tue, 27 Apr 2010 08:13:59 -0400
> From: pjintheusa at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
>
> Hi,
>
> I have done this using scripting. managed/c# in my case. Basically all
> inbound calls are passed to a script - get-session-data, This script calls a
> ws and populates session variables with the result. I then pass the call
> back to the dialplan for further processing,
>
> HTH
>
> Phil
>
>
>  If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let
> me know how you did it.
>
> Jan
>
>
> ------------------------------
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Free, trusted and rich email service. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Free, trusted and rich email service. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Apr 27 11:39:56 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 27 Apr 2010 14:39:56 -0400
Subject: [Freeswitch-users] Freeswitch and dynamic application
In-Reply-To: <4BD6ACA9.7000607@cartissolutions.com>
References: <u2q95237e3b1004270506rc0bd580cte61730445d0b6754@mail.gmail.com>
	<4BD6ACA9.7000607@cartissolutions.com>
Message-ID: <x2i367751821004271139r3e4ae29au66f3a9eff42c5002@mail.gmail.com>

My opinion

Lua - depending on your app - more load your server and more to maintain on
multiple servers
ESL - you can move your scripting logic off the freeswitch server and
control multiple servers from one app - potentially a more scalable
solution.

So it just depends on how big and how dynamic your app is.

On Tue, Apr 27, 2010 at 5:21 AM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

> On 4/27/2010 7:06 AM, Martr?s Jean-Jacques wrote:
> > Hi guys,
> >
> > I need to build dynamic application using freeswitch.
> > Based on your experiences what is the best solution ?
> >
> > Using a LUA script or use the event_socket ?
>
> There isn't any one particular "best" way other than the way that
> accomplishes what you need in the most efficient manner.  FreeSWITCH is
> sort of like a swiss army knife in this regard - you can use LUA,
> Javascript, C, C++, managed code, etc. etc. etc.
>
> Sure, we can give you our opinions on what we think is "best" based upon
> your actual use-case scenario...  But again, it's only general guidance
> really.
>
> --
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.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 mranga at gmail.com  Tue Apr 27 11:46:49 2010
From: mranga at gmail.com (M. Ranganathan)
Date: Tue, 27 Apr 2010 14:46:49 -0400
Subject: [Freeswitch-users] Header url encoding in Free SWITCH Event
Message-ID: <n2q5c7eebdd1004271146q7ab60ed3i8a501dd7f2a6cdf8@mail.gmail.com>

Hello,

When SIP Headers are passed in FreeSWITCH Events, is the entire header
expected to be URL encoded (including things like Display Name) ?

Thanks

Ranga

-- 
M. Ranganathan


From kenfulmer at icstechnologysolutions.com  Tue Apr 27 11:48:16 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 13:48:16 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
Message-ID: <014e01cae63a$32fcf250$98f6d6f0$@com>

Very true. We're just trying to stay one step ahead of the local
competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real
name and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure the
> competition doesn't know what we're doing for as long as possible. That's
> all.
>
>
>
> Ken

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.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 mustafa.pk at gmail.com  Tue Apr 27 12:02:41 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Wed, 28 Apr 2010 00:02:41 +0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <014e01cae63a$32fcf250$98f6d6f0$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
Message-ID: <y2o8213d6071004271202vb5768842g2db8df7c9faf47c4@mail.gmail.com>

I have seen many people doing this, but competitors are the people
first to know what you are hiding behind  :) trust me

On Tue, Apr 27, 2010 at 11:48 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> Very true. We're just trying to stay one step ahead of the local
> competition.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
> Kielhofner
> Sent: Tuesday, April 27, 2010 12:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> While I understand the request:
>
> 1) ?They are going to find out anyway.
> 2) ?You are posting to FreeSWITCH-users using (presumably) your real
> name and corporate e-mail address.
>
> If you were looking to hide FreeSWITCH it's a little too late.
>
> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
>> It's not meant to be offensive or anything. We'd just like to ensure the
>> competition doesn't know what we're doing for as long as possible. That's
>> all.
>>
>>
>>
>> Ken
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.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
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From david.ponzone at gmail.com  Tue Apr 27 12:13:28 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 27 Apr 2010 21:13:28 +0200
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <y2o8213d6071004271202vb5768842g2db8df7c9faf47c4@mail.gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<y2o8213d6071004271202vb5768842g2db8df7c9faf47c4@mail.gmail.com>
Message-ID: <ADC5A2A7-BEFA-4476-95DA-4799C440170C@gmail.com>

Well, if you dont show anything, probably.
But if you replace all strings containing FreeSWITCH by Asterisk or  
Sonus, you can make them crazy for months :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 27/04/2010 ? 21:02, Ghulam Mustafa a ?crit :

> I have seen many people doing this, but competitors are the people
> first to know what you are hiding behind  :) trust me
>
> On Tue, Apr 27, 2010 at 11:48 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
>> Very true. We're just trying to stay one step ahead of the local
>> competition.
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
>> Kristian
>> Kielhofner
>> Sent: Tuesday, April 27, 2010 12:37 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>>
>> While I understand the request:
>>
>> 1)  They are going to find out anyway.
>> 2)  You are posting to FreeSWITCH-users using (presumably) your real
>> name and corporate e-mail address.
>>
>> If you were looking to hide FreeSWITCH it's a little too late.
>>
>> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
>> <kenfulmer at icstechnologysolutions.com> wrote:
>>> It's not meant to be offensive or anything. We'd just like to  
>>> ensure the
>>> competition doesn't know what we're doing for as long as possible.  
>>> That's
>>> all.
>>>
>>>
>>>
>>> Ken
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.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
>>
>
>
>
> -- 
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.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 brian at freeswitch.org  Tue Apr 27 12:18:46 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Apr 2010 14:18:46 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <ADC5A2A7-BEFA-4476-95DA-4799C440170C@gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<y2o8213d6071004271202vb5768842g2db8df7c9faf47c4@mail.gmail.com>
	<ADC5A2A7-BEFA-4476-95DA-4799C440170C@gmail.com>
Message-ID: <136855E2-4AB5-448C-A435-611B58B4087F@freeswitch.org>

Its kinda pointless because the X-FS headers all over the place GIVE IT AWAY!

/b

On Apr 27, 2010, at 2:13 PM, David Ponzone wrote:

> Well, if you dont show anything, probably.
> But if you replace all strings containing FreeSWITCH by Asterisk or Sonus, you can make them crazy for months :)
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 

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From anthony.minessale at gmail.com  Tue Apr 27 12:43:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 14:43:45 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
	<i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
	<D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>
Message-ID: <l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>

The problem here is that you were testing it against a 1 legged call (echo
application is not bridged to anything)

This code is usually only encountered when replacing a leg of a call.

try git rev [8660b6f] or better

I added a patch so the new channel executes the same app that the old was
originally executing and hangs up on the original.

Note, if you want the transferred call to pick up on the other call to the
application in-progress you would have to loop the original call.




On Tue, Apr 27, 2010 at 11:03 AM, Mardy Marshall <mardy at voysys.com> wrote:

> The REFER exchange occurs on the other side of the proxy between the two
> Polycom phones.  Since
> FreeSWITCH is the destination of the transfer, it will only see the INVITE
> w/Replaces.  I've sent you the
> PCAP file which captures the complete SIP exchange.  That should help to
> clarify.
>
> -Mardy
>
> On Apr 27, 2010, at 11:47 AM, Anthony Minessale wrote:
>
> There is no sign of a REFER packet or anything indicating an attended
> transfer in this trace.
> Did you send the wrong one possibly?
>
>
> On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> Here is the setup that was used to reproduce the consultative transfer
>> problem.
>>
>> There are two boxes, the first is running a proxy based PBX (sipXecs) and
>> the
>> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones,
>> extension
>> 200 and 202, registered with it.  The PBX has configured a mapping rule
>> which
>> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060which
>> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
>> configured to allow connections from the PBX box via an ACL configuration
>> and
>> the public dialplan includes an "echo" extension:
>>
>>     <extension name="echo">
>>       <condition field="destination_number" expression="^9996$">
>>         <action application="answer"/>
>>         <action application="echo"/>
>>       </condition>
>>     </extension>
>>
>> Any of the phones registered with the PBX can dial extension 9996 and be
>> connected to the FreeSWITCH echo application.  But when one phone attempts
>> to transfer another phone to extension 9996 via a consultative transfer,
>> FreeSWITCH does not properly complete the transfer.  You can see in the
>> log
>> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
>> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
>> initiated the transfer.  Without that terminating BYE, the transfer
>> controller thinks that the transfer failed.
>>
>> The corresponding FreeSWITCH log file -
>> http://pastebin.freeswitch.org/12806
>>
>> If it will help, I can also forward a corresponding PCAP file.
>>
>> Thanks
>>
>> -Mardy
>>
>> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>>
>> I'm trying to understand this:
>>
>> If FS is acting as a phone in your scenario why are you sending a refer to
>> it and not the server?
>> In most situations there is a b2bua server who routes the calls and takes
>> all the REFER.
>> Is this one of those PROXY only sip servers?
>>
>> I think you would need to produce a full debug log of this, and if you are
>> using some kind of proxy based setup we would need some way to easily
>> reproduce it or visit your lab because we do not typically use anything of
>> the sort.
>>
>> Execute these commands and reproduce it and capture the whole log and put
>> it on
>> http://pastebin.freeswitch.org
>>
>> sofia profile internal siptrace on
>> console loglevel debug
>>
>>
>>
>>
>>
>> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> instead of emailing again when impatient for an answer (something we
>>> frown upon here in this busy list)
>>> produce a reproducible step by step process to duplicate your issue.  We
>>> are trying to help people but we don't have the time to do the leg work for
>>> everyone who asks a question when we get hundreds of emails a day.
>>>
>>>
>>>
>>>
>>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>
>>>> Just following up...  Does anyone have any suggestions on how to proceed
>>>> with this?  I've run out of ideas.
>>>>
>>>> Thanks,
>>>>
>>>> -Mardy
>>>>
>>>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>>>
>>>> The phones that I am using are not registered with FS.  They are
>>>> registered with another proxy based PBX.  I am simply using FS as B2BUA
>>>> which is also registered with the PBX.  And yes, I can successfully transfer
>>>> a call to another phone with this setup.
>>>>
>>>> To simplify things I tried the same scenario using FSComm in place of my
>>>> own FS application and tried to transfer a call to FSComm with the same
>>>> results.  And just in case there might be a problem specific to FSComm, I
>>>> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
>>>> echo application with the same results.  By the way, I have no problems with
>>>> blind transfers, only attended transfers.
>>>>
>>>> -Mardy
>>>>
>>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>>>
>>>> did you try just setting up 2 phones on plain fresh FS install, and
>>>> calling them normally and transferring them around?
>>>> That description is still pretty vague? What is an Event Socket
>>>> application, which has nothing to do with sip and sip transfers, that's a FS
>>>> protocol.
>>>>
>>>>
>>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>
>>>>> I have two phones (Polycom) and an event_socket application, all of
>>>>> which are using a SIP proxy for call routing.  The first phone calls the
>>>>> second phone.  The second phone then attempts to transfer the call to the
>>>>> FS/event_socket application by first placing the call on hold and then
>>>>> calling the FS application, followed by a consultative transfer.  The REFER
>>>>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>>>>  The transferred call leg appears to be answered by FS and the application
>>>>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>>>>> that I see is that the original call leg, created when the user called the
>>>>> FS application to announce the transfer, does not get canceled by FS and
>>>>> subsequently does not send the BYE back to the Polycom.  Is there something
>>>>> that I need to do at the event_socket application to complete the transfer?
>>>>>  I've tried killing the UUID associated with the first call leg as well as
>>>>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>>>
>>>>> -Mardy
>>>>>
>>>>>
>>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>>>
>>>>> but what is the client sending the REFER?
>>>>>
>>>>> FS gets refer+replaces all the time, if it's the one where the dest is
>>>>> on another box (aka the nightmare xfer that you should see references to in
>>>>> the debug log if so) then it will not complete until that far end call is
>>>>> answered.
>>>>>
>>>>> FS handles this scenerio for us hundreds of times a day using a wide
>>>>> range of sip devices so perhaps
>>>>> your UA has an interop problem.
>>>>>
>>>>>
>>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>
>>>>>>
>>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>>>
>>>>>> uuid_simplify will issue the refer...
>>>>>>
>>>>>>
>>>>>> I looked at uuid_simplify and if I understand it correctly it is for
>>>>>> use when one wants to act as the transfer controller.  In my case, FS is the
>>>>>> transfer destination.  Another phone has already generated the refer and FS
>>>>>> has been sent an invite with replaces.
>>>>>>
>>>>>>
>>>>>> May I ask what application you are developing?
>>>>>>
>>>>>>
>>>>>> An ACD.
>>>>>>
>>>>>>
>>>>>> Regards,
>>>>>> Jo?o Mesquita
>>>>>> FSComm developer
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>>
>>>>>>> I'm having a problem with attended transfers where the destination of
>>>>>>> the transfer is a FreeSWITCH based application such as FSComm.  (It should
>>>>>>> be noted that in my setup the phone performing the transfer and the caller
>>>>>>> which is being transferred are parties of another SIP server.)  What I see,
>>>>>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>>>>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>>>>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>>>>> application side, FS still thinks that both the initial call leg and
>>>>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>>>>> kill all legs of the call.  Is there a specific action that the application
>>>>>>> must take in order for the transfer to complete?
>>>>>>>
>>>>>>> -Mardy
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> 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>
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mardy at voysys.com  Tue Apr 27 12:56:30 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Tue, 27 Apr 2010 15:56:30 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
	<i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
	<D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>
	<l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>
Message-ID: <1E3EE87F-7D6B-4BD8-A19A-B5FDF3623539@voysys.com>

I'll rebuild and give it a try.

Thanks!

-Mardy

On Apr 27, 2010, at 3:43 PM, Anthony Minessale wrote:

> The problem here is that you were testing it against a 1 legged call (echo application is not bridged to anything)
> 
> This code is usually only encountered when replacing a leg of a call.
> 
> try git rev [8660b6f] or better
> 
> I added a patch so the new channel executes the same app that the old was originally executing and hangs up on the original.
> 
> Note, if you want the transferred call to pick up on the other call to the application in-progress you would have to loop the original call.
> 
> 
> 
> 
> On Tue, Apr 27, 2010 at 11:03 AM, Mardy Marshall <mardy at voysys.com> wrote:
> The REFER exchange occurs on the other side of the proxy between the two Polycom phones.  Since
> FreeSWITCH is the destination of the transfer, it will only see the INVITE w/Replaces.  I've sent you the
> PCAP file which captures the complete SIP exchange.  That should help to clarify.
> 
> -Mardy
> 
> On Apr 27, 2010, at 11:47 AM, Anthony Minessale wrote:
> 
>> There is no sign of a REFER packet or anything indicating an attended transfer in this trace.
>> Did you send the wrong one possibly?
>> 
>> 
>> On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:
>> Here is the setup that was used to reproduce the consultative transfer problem.
>> 
>> There are two boxes, the first is running a proxy based PBX (sipXecs) and the
>> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones, extension
>> 200 and 202, registered with it.  The PBX has configured a mapping rule which
>> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060 which
>> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
>> configured to allow connections from the PBX box via an ACL configuration and
>> the public dialplan includes an "echo" extension:
>> 
>>     <extension name="echo">
>>       <condition field="destination_number" expression="^9996$">
>>         <action application="answer"/>
>>         <action application="echo"/>
>>       </condition>
>>     </extension>
>> 
>> Any of the phones registered with the PBX can dial extension 9996 and be
>> connected to the FreeSWITCH echo application.  But when one phone attempts
>> to transfer another phone to extension 9996 via a consultative transfer,
>> FreeSWITCH does not properly complete the transfer.  You can see in the log
>> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
>> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
>> initiated the transfer.  Without that terminating BYE, the transfer
>> controller thinks that the transfer failed.
>> 
>> The corresponding FreeSWITCH log file - http://pastebin.freeswitch.org/12806
>> 
>> If it will help, I can also forward a corresponding PCAP file.
>> 
>> Thanks
>> 
>> -Mardy
>> 
>> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>> 
>>> I'm trying to understand this:
>>> 
>>> If FS is acting as a phone in your scenario why are you sending a refer to it and not the server?
>>> In most situations there is a b2bua server who routes the calls and takes all the REFER.
>>> Is this one of those PROXY only sip servers?
>>> 
>>> I think you would need to produce a full debug log of this, and if you are using some kind of proxy based setup we would need some way to easily reproduce it or visit your lab because we do not typically use anything of the sort.
>>> 
>>> Execute these commands and reproduce it and capture the whole log and put it on
>>> http://pastebin.freeswitch.org
>>> 
>>> sofia profile internal siptrace on
>>> console loglevel debug
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> instead of emailing again when impatient for an answer (something we frown upon here in this busy list) 
>>> produce a reproducible step by step process to duplicate your issue.  We are trying to help people but we don't have the time to do the leg work for everyone who asks a question when we get hundreds of emails a day.
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>> Just following up...  Does anyone have any suggestions on how to proceed with this?  I've run out of ideas.
>>> 
>>> Thanks,
>>> 
>>> -Mardy
>>> 
>>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>> 
>>>> The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.
>>>> 
>>>> To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.
>>>> 
>>>> -Mardy
>>>> 
>>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>>> 
>>>>> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
>>>>> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
>>>>> 
>>>>> 
>>>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>>> 
>>>>> -Mardy
>>>>> 
>>>>> 
>>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>>> 
>>>>>> but what is the client sending the REFER?
>>>>>> 
>>>>>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>>>>>> 
>>>>>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>>>>>> your UA has an interop problem.
>>>>>> 
>>>>>> 
>>>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>>> 
>>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>>> 
>>>>>>> uuid_simplify will issue the refer...
>>>>>> 
>>>>>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>>>>>> 
>>>>>>> 
>>>>>>> May I ask what application you are developing?
>>>>>> 
>>>>>> An ACD.
>>>>>> 
>>>>>>> 
>>>>>>> Regards,
>>>>>>> Jo?o Mesquita
>>>>>>> FSComm developer
>>>>>>> 
>>>>>>> 
>>>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>>>>>> 
>>>>>>> -Mardy
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> -- 
>>>>>> Anthony Minessale II
>>>>>> 
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>> 
>>>>>> 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
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> Anthony Minessale II
>>>>> 
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> 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
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From math.parent at gmail.com  Tue Apr 27 13:15:51 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 27 Apr 2010 22:15:51 +0200
Subject: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
In-Reply-To: <0405D805B8EA40CD9288FF4AF9974963@bp1.ad.bp.com>
References: <4B281EC072E8405A9B2ACF2925901D65@bp1.ad.bp.com>
	<0405D805B8EA40CD9288FF4AF9974963@bp1.ad.bp.com>
Message-ID: <n2q960738411004271315u16d5e5dcj27e2a89a67312bb8@mail.gmail.com>

On Fri, Apr 23, 2010 at 10:23 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Hi,
>
> I'm still struggling to get the VS2008 Express Build working, but there
> seems to be another problem with mod_skinny
I can't help much on VS 2k8 part but,

> Mathieu,
>
> mod_skinny.c has a #include looking for skinny_api.h, the file does not seem
> to be there ?

It is here. see
http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_skinny


Mathieu


From oseslija at gmail.com  Tue Apr 27 13:18:50 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 27 Apr 2010 22:18:50 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
Message-ID: <k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>

E3 afaik is 34 Mbit/s. I think when channelized, you can get max of 16 E1s.
16 x 30 = 480 calls max, so 6000 calls can't fit.
You will need a channelized STM-1 I think.

O.


On Tue, Apr 27, 2010 at 4:43 PM, Anita Hall <anita.hall at simmortel.com>wrote:

> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Freeswitch or Asterisk ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls
> simultaneously. I want to do some homework before taking it further with
> him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not
> look feasible ?
>
> Thanks for any input you may provide.
>
> regards,
>
> Anita Hall,
> Simmortel Voice.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at anticat.ch  Tue Apr 27 13:20:14 2010
From: freeswitch at anticat.ch (=?ISO-8859-1?Q?Andreas_Dr=F6scher?=)
Date: Tue, 27 Apr 2010 22:20:14 +0200
Subject: [Freeswitch-users] Route calls based on the Nb they are divert
 by
In-Reply-To: <u2lb39273531004261504v575ca729r320d5991f5e2d510@mail.gmail.com>
References: <4BD5FC3F.7060104@anticat.ch>
	<u2lb39273531004261504v575ca729r320d5991f5e2d510@mail.gmail.com>
Message-ID: <4BD746FE.1010304@anticat.ch>

There is a whole lot of information. If I knew this command earlier 
other stuff would have been easer also.

T/hank you/ for the /hint/!.

However the number I was hoping for, does not show up. ;-(

Am 27.04.2010 00:04, schrieb Rupa Schomaker:

> It all depends on what your ITSP sends you.  Try the "info" app to see 
> if there is any mention of the number you are diverting from.  Pretty 
> much everything there can be used for routing decisions.
>
> On Mon, Apr 26, 2010 at 3:49 PM, Andreas Dr?scher 
> <freeswitch at anticat.ch <mailto:freeswitch at anticat.ch>> wrote:
>
>     Hi everyone
>
>     I was wondering if it is possible route calls based on the numbers
>     they were
>     divert by, perhaps using "sip_redirected_by".
>
>     Let's assume I own two phone numbers of my old fixed telephone
>     lines and one
>     VoIP Account/Number. If I redirect the two fixnet numbers to the
>     same VoIP
>     number, is it possible to "demultiplex" the incoming calls based
>     on the number
>     they were redirected by?
>
>     I am not sure about the available meta data provided by SIP,
>     however I am
>     certain that my ISDN phone is sometimes showing "xxxx is calling,
>     redirected by
>     yyyyy"
>
>     Best Wishes
>     Andreas
>



From mgg at giagnocavo.net  Tue Apr 27 13:38:32 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 27 Apr 2010 16:38:32 -0400
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <014e01cae63a$32fcf250$98f6d6f0$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>

Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are still competition? I mean, you got people sitting around going "hmm how can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Fulmer
Sent: Tuesday, April 27, 2010 12:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Very true. We're just trying to stay one step ahead of the local competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real name and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure 
> the competition doesn't know what we're doing for as long as possible. 
> That's all.
>
>
>
> Ken

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.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 stevendt at primrosebank.net  Tue Apr 27 13:45:20 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 27 Apr 2010 21:45:20 +0100
Subject: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
Message-ID: <B2760BB7724A40D992588D1529E815D1@bp1.ad.bp.com>

Mathieu,

yes, it has made it into the SVN distribution now. I don't know why it was 
missing before, but it's there now

regards
Dave


----- Original Message ----- 
From: "Dave Stevenson" <stevendt at primrosebank.net>
To: "Mathieu Parent" <math.parent at gmail.com>
Sent: Tuesday, April 27, 2010 9:36 PM
Subject: Re: [Freeswitch-users] mod_skinny - Building with VS 2008 Express


> Hi Mathieu
>
> thanks for the reply - yes, I can see it in the Git link that you 
> included, but I'm currently using SVN. I don't know how far SVN is behind 
> Git at the moment, but the skinny_api files were not downloaded when I 
> last did an SVN update - I'll try again now and see if they have made it 
> to the SVN distribution yet. I will be able to download them from the link 
> in the meantime though
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Mathieu Parent" <math.parent at gmail.com>
> To: "Dave Stevenson" <stevendt at primrosebank.net>
> Cc: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, April 27, 2010 9:15 PM
> Subject: Re: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
>
>
>> On Fri, Apr 23, 2010 at 10:23 PM, Dave Stevenson
>> <stevendt at primrosebank.net> wrote:
>>> Hi,
>>>
>>> I'm still struggling to get the VS2008 Express Build working, but there
>>> seems to be another problem with mod_skinny
>> I can't help much on VS 2k8 part but,
>>
>>> Mathieu,
>>>
>>> mod_skinny.c has a #include looking for skinny_api.h, the file does not 
>>> seem
>>> to be there ?
>>
>> It is here. see
>> http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_skinny
>>
>>
>> Mathieu
>>
> 



From kenfulmer at icstechnologysolutions.com  Tue Apr 27 13:54:38 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 15:54:38 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
References: <00a601cae61a$ab89bf70$029d3e50$@com>	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>	<00fb01cae623$9f364be0$dda2e3a0$@com>	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
Message-ID: <017901cae64b$da496820$8edc3860$@com>

We have competitors using vendor solutions that are expensive as hell. So,
we are trying to counter with a less expensive but powerful solution. What
exactly is wrong with that? And what is wrong with trying to extend the
barrier to entry for our competitors as long as possible? Not sure why
there's such a mystery here...




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Giagnocavo
Sent: Tuesday, April 27, 2010 3:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
still competition? I mean, you got people sitting around going "hmm how can
bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Tuesday, April 27, 2010 12:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Very true. We're just trying to stay one step ahead of the local
competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real name
and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure 
> the competition doesn't know what we're doing for as long as possible. 
> That's all.
>
>
>
> Ken

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



From anthony.minessale at gmail.com  Tue Apr 27 13:54:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 15:54:35 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
Message-ID: <h2y191c3a031004271354j5262b804za5e82c8f3f9923b2@mail.gmail.com>

Come to ClueCon!

Hmm a bit offtopic but this thread is a dead horse anyway may as well spin
it into a ClueCon add.
Maybe you can sponsor the conference before the competition figures out to
sponsor as well.


On Tue, Apr 27, 2010 at 3:38 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are still competition? I mean, you got people sitting around going "hmm how
> can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Fulmer
> Sent: Tuesday, April 27, 2010 12:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Very true. We're just trying to stay one step ahead of the local
> competition.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian Kielhofner
> Sent: Tuesday, April 27, 2010 12:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> While I understand the request:
>
> 1)  They are going to find out anyway.
> 2)  You are posting to FreeSWITCH-users using (presumably) your real name
> and corporate e-mail address.
>
> If you were looking to hide FreeSWITCH it's a little too late.
>
> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
> > It's not meant to be offensive or anything. We'd just like to ensure
> > the competition doesn't know what we're doing for as long as possible.
> > That's all.
> >
> >
> >
> > Ken
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From janvb at live.com  Tue Apr 27 14:01:05 2010
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Apr 2010 23:01:05 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>,
	<k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>
Message-ID: <SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>


Actually she could need an STM-4 

 



Date: Tue, 27 Apr 2010 22:18:50 +0200
From: oseslija at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] E3 Card on Freeswitch ?

E3 afaik is 34 Mbit/s. I think when channelized, you can get max of 16 E1s. 
16 x 30 = 480 calls max, so 6000 calls can't fit.
You will need a channelized STM-1 I think.

O.




On Tue, Apr 27, 2010 at 4:43 PM, Anita Hall <anita.hall at simmortel.com> wrote:

Hi


Please check out this product

http://www.sangoma.com/products/hardware_products/data_networking/a301.html

Does it work on Freeswitch or Asterisk ?
Do Telcos provide an E3 connection ?

One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ?

Thanks for any input you may provide.

regards,

Anita Hall,
Simmortel Voice.


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


 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
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From anthony.minessale at gmail.com  Tue Apr 27 14:08:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 16:08:26 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <017901cae64b$da496820$8edc3860$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
Message-ID: <p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>

Ok so I guess that means you don't want to sponsor ClueCon......

What's wrong with it is that is your problem not ours.
Our problem is to write a full featured soft switch for free for everyone to
use not just you.

Wanting to make FreeSWITCH the world's best kept secret is not on our
roadmap so we don't rush out to teach everyone how to hide it but if you
look hard enough you'll probably figure it out.




On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

> We have competitors using vendor solutions that are expensive as hell. So,
> we are trying to counter with a less expensive but powerful solution. What
> exactly is wrong with that? And what is wrong with trying to extend the
> barrier to entry for our competitors as long as possible? Not sure why
> there's such a mystery here...
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> Giagnocavo
> Sent: Tuesday, April 27, 2010 3:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are
> still competition? I mean, you got people sitting around going "hmm how can
> bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> Fulmer
> Sent: Tuesday, April 27, 2010 12:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Very true. We're just trying to stay one step ahead of the local
> competition.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian
> Kielhofner
> Sent: Tuesday, April 27, 2010 12:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> While I understand the request:
>
> 1)  They are going to find out anyway.
> 2)  You are posting to FreeSWITCH-users using (presumably) your real name
> and corporate e-mail address.
>
> If you were looking to hide FreeSWITCH it's a little too late.
>
> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
> > It's not meant to be offensive or anything. We'd just like to ensure
> > the competition doesn't know what we're doing for as long as possible.
> > That's all.
> >
> >
> >
> > Ken
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Apr 27 14:09:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Apr 2010 16:09:50 -0500
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
	<k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>
	<SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>
Message-ID: <s2r191c3a031004271409y45cdc458jcb7bf7a4c6d6c72c@mail.gmail.com>

Not on one box though! We can only work so many miracles. ;)



On Tue, Apr 27, 2010 at 4:01 PM, Jan Berger <janvb at live.com> wrote:

>  Actually she could need an STM-4
>
> ------------------------------
> Date: Tue, 27 Apr 2010 22:18:50 +0200
> From: oseslija at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] E3 Card on Freeswitch ?
>
>
> E3 afaik is 34 Mbit/s. I think when channelized, you can get max of 16 E1s.
>
> 16 x 30 = 480 calls max, so 6000 calls can't fit.
> You will need a channelized STM-1 I think.
>
> O.
>
>
>
> On Tue, Apr 27, 2010 at 4:43 PM, Anita Hall <anita.hall at simmortel.com>wrote:
>
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Freeswitch or Asterisk ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls
> simultaneously. I want to do some homework before taking it further with
> him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not
> look feasible ?
>
> Thanks for any input you may provide.
>
> regards,
>
> Anita Hall,
> Simmortel Voice.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Free, trusted and rich email service. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Tue Apr 27 14:14:53 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 27 Apr 2010 23:14:53 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
	<k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>
	<SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>
Message-ID: <h2y4468a6771004271414q7b09d6deh49f253e9992ab7d6@mail.gmail.com>

Right.

On Tue, Apr 27, 2010 at 11:01 PM, Jan Berger <janvb at live.com> wrote:

>  Actually she could need an STM-4
>
> ------------------------------
> Date: Tue, 27 Apr 2010 22:18:50 +0200
> From: oseslija at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] E3 Card on Freeswitch ?
>
>
> E3 afaik is 34 Mbit/s. I think when channelized, you can get max of 16 E1s.
>
> 16 x 30 = 480 calls max, so 6000 calls can't fit.
> You will need a channelized STM-1 I think.
>
> O.
>
>
>
> On Tue, Apr 27, 2010 at 4:43 PM, Anita Hall <anita.hall at simmortel.com>wrote:
>
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Freeswitch or Asterisk ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls
> simultaneously. I want to do some homework before taking it further with
> him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not
> look feasible ?
>
> Thanks for any input you may provide.
>
> regards,
>
> Anita Hall,
> Simmortel Voice.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Free, trusted and rich email service. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 27 14:20:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Apr 2010 14:20:55 -0700
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <017901cae64b$da496820$8edc3860$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
Message-ID: <o2k87f2f3b91004271420j2d1c3c1fq49f6ccded50b50ed@mail.gmail.com>

On Tue, Apr 27, 2010 at 1:54 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

> We have competitors using vendor solutions that are expensive as hell. So,
> we are trying to counter with a less expensive but powerful solution. What
> exactly is wrong with that? And what is wrong with trying to extend the
> barrier to entry for our competitors as long as possible? Not sure why
> there's such a mystery here...
>

No mystery, just a question of whether masking your UA in the SDP will have
a net positive effect. The guys who've chimed in on this thread are
battle-hardened veterans of the VoIP business. If they felt that messing
with the SDP could reasonably give you a competitive edge then they'd tell
you how to do it. You are not at all the first person to ask about this.
People tried this years ago in Asterisk, and AFAICT the ROI was near zero or
worse - things stopped working and people had to waste time/money/energy
chasing down SIP interop issues.

That all being said, no one here will try to stop you from trying. Just be
prepared to go it alone because most of the experts here really aren't
interested in helping people do things that could have such a potentially
negative impact on their systems.

-MC


>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> Giagnocavo
> Sent: Tuesday, April 27, 2010 3:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are
> still competition? I mean, you got people sitting around going "hmm how can
> bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> Fulmer
> Sent: Tuesday, April 27, 2010 12:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Very true. We're just trying to stay one step ahead of the local
> competition.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian
> Kielhofner
> Sent: Tuesday, April 27, 2010 12:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> While I understand the request:
>
> 1)  They are going to find out anyway.
> 2)  You are posting to FreeSWITCH-users using (presumably) your real name
> and corporate e-mail address.
>
> If you were looking to hide FreeSWITCH it's a little too late.
>
> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
> > It's not meant to be offensive or anything. We'd just like to ensure
> > the competition doesn't know what we're doing for as long as possible.
> > That's all.
> >
> >
> >
> > Ken
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gkuri at ieee.org  Mon Apr 26 16:41:37 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 26 Apr 2010 16:41:37 -0700
Subject: [Freeswitch-users] g.729 Licenses from Freeswitch.org
In-Reply-To: <x2g191c3a031004261522z4b607aaaw19b3e75a084d5d9e@mail.gmail.com>
References: <00eb01cae248$fe1f7e80$fa5e7b80$@com>
	<5FBBBF6C-99C5-4959-8B3E-55C2B470A4DB@freeswitch.org>
	<p2o8b1c9cda1004222230t43c05083zc62b04d374885d2f@mail.gmail.com>
	<x2g191c3a031004261522z4b607aaaw19b3e75a084d5d9e@mail.gmail.com>
Message-ID: <n2w8b1c9cda1004261641h711ff617p9a30d4adebfbadbe@mail.gmail.com>

Responses inline ...

On Mon, Apr 26, 2010 at 3:22 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Based on your response, I could be wrong but, I get the impression you are
> intentionally trying to stir up some kind of controversy.  Is there a
> specific reason you feel the need to raise your concerns on our general
> users mailing lists and not to consulting at freeswitch.org where they
> belong.  This has gone from asking some questions to some sort of rhetoric
> against our policy that clearly already addresses your concerns.
>

I'm not trying to stir up any controversy, I simply responded to Brian's
request on the mailing list, to "prove the machine is dead" when someone
asked about having a license reissued. We might have a need to purchase g729
licenses down the road, and I certainly think it's fair enough for us or
anyone else purchasing licenses, to know what you would expect of those
requesting a license reissued for any valid reason (not just for a dead
box). Requesting proof the box is dead, obviously seems a little difficult
to prove, which is why I posed those questions. Brian's and Mike's responses
of a signed paper stating the reason, seems fine to me.

There's nothing documented on how the licensing actually works other than
how to get it running - is it tied to the MAC address or some other hardware
ID, that would be good to know? If we were to purchase licenses, we'd want
to test them in our development environment first. I think something should
be added to the wiki so people don't accidentally activate the licenses on
their development server first, in order to test out mod_com_g729, and then
think they're just going to move the licenses over to their production
server and expect it to work. Why should I email consulting at freeswitch.org,
and exclude other people in the community that might be purchasing g729
licenses, I think they're completely valid questions anyone in the community
should know the answer to, no?

Do you really think you are going to have a whole bunch of boxes fail to
> some degree that we will not be willing to re-issue your licenses or are you
> just more interested in starting a flame war on our mailing list?  The
> licenses are to the specifications of the requirements of the contracts we
> signed.  We did so taking the on the risk and responsibility to make g729
> available to the community.
>

No, I don't really think all of our boxes are going to fail at once, it was
about reallocation of resources or upgrading a box, which occurs more often.
I think it's fair for us to know what to expect should we ask to have a
license reissued.

We had a discussion about floating licenses and concluded it was unwise and
> not worth the risk and security implications.  If you use g729 you know how
> much traffic you are going to use and if you want to remain stable you
> should not want to add another moving part to the equation that could go
> down and make all of your calls fail


While I understand your reasoning of adding something else to the equation
that could fail and cause dropped calls, I also feel that's a business
decision that should be made by the company purchasing the licenses,
assuming they're given both options (floating and server assigned). Given my
choice, I'd probably take that "chance" and would rather have floating
licenses but will take whatever you guys offer (not Howler), as I'm
committed to supporting the project as much as possible.


> This is the policy we have implemented and I think you are getting the good
> end of the bargain with a free telephony server with an efficient stable
> g729 codec implementation at an affordable cost.  If a high-enough demand
> arises for floating licenses and we have the time and resources to implement
> it, we may reconsider it but only a tiny fraction of our customers have even
> asked about it, let alone demanded it.


Yes, FreeSWITCH is awesome!

Cheers,
Gabe
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From jesse at cronomagic.com  Tue Apr 27 11:04:01 2010
From: jesse at cronomagic.com (Jesse Cloutier)
Date: Tue, 27 Apr 2010 14:04:01 -0400
Subject: [Freeswitch-users] say phrase into recordFile using lua script
Message-ID: <4BD72711.5060605@cronomagic.com>

Hello, I am having trouble with a lua script that i am trying to put 
together.

The script basically makes a recording using the "session:recordFile()" 
function, and is terminated by a callback function that returns "break" 
on dtmf "#"

What I would like to do is have the system say the date and time ( 
session:say() ) into the recording at the end of the recording. It 
should be triggered by
the call back.

When I try it out it works great, I hear the system say the date / time, 
except that it is never recorded into the recording.

I have been searching the docs for something that might clear up why its 
not recording but I can't find anything. I thought maybe doing a 
uuid_broadcast would fix the problem
but there is no builtin function for that in the lua api and I was not 
able to make it work using the "session:execute" function either.

Any help would be really appreaciated!

This is the relavent code, for testing purposes I am having the date 
spoken using an alternative digit, you can also see my attempt to do a 
uuid_broadcast and sched_broadcast commented out:

######################
--call back function
function onInputRecord(s, type, obj) -- function to end recording

   if (type == "dtmf") then
       if ( obj['digit'] == '1') then
       curDate = os.date();
         session:say(curDate, "en", "CURRENT_DATE_TIME", "pronounced");

         --uuidBroadcast = " say::en CURRENT_DATE_TIME pronounced " .. 
os.date() .. " both";
         --session:execute("uuid_broadcast ".. UUID, uuidBroadcast);
         --session:execute("sched_broadcast", "data=\"+1 
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-hello.wav both\"");

         return true;
       elseif ( obj['digit'] == '#' ) then
         return "break";
       end
   end
end

session:setInputCallback("onInputRecord", "");
RECORD_PATH = /parth/to/file
session:recordFile(RECORD_PATH, 30000, 10, 10); -- pressing # ends the 
recording
####################

Thanks alot!

JC



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From paul at cupis.co.uk  Tue Apr 27 13:38:10 2010
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 27 Apr 2010 21:38:10 +0100
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>
Message-ID: <hr7hvi$2jl$1@dough.gmane.org>

On 27/04/10 15:43, Anita Hall wrote:
> One of our customers had an inquiry for terminating 6000 calls
> simultaneously. I want to do some homework before taking it further with
> him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not
> look feasible ?

I would recommend you look at a provider who can give you SIP 
interconnect, or looking at a hardware media gateways which can support 
the TDM side of this equation and convert to SIP for your FreeSwitch 
server(s).




From gkuri at ieee.org  Tue Apr 27 14:29:45 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 27 Apr 2010 14:29:45 -0700
Subject: [Freeswitch-users] Intercom w/ Mute
Message-ID: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>

Is there support for dialing an extension and automatically answering and
muting the dialed extension? I found this page on the wiki, but it doesn't
mention anything about automaticaly muting the call?

     http://wiki.freeswitch.org/wiki/Intercom

Cheers,
Gabe
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From kenfulmer at icstechnologysolutions.com  Tue Apr 27 14:34:28 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 16:34:28 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>	<00fb01cae623$9f364be0$dda2e3a0$@com>	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>	<014e01cae63a$32fcf250$98f6d6f0$@com>	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
Message-ID: <019201cae651$6b38bb60$41aa3220$@com>

Sorry if I came across harshly toward the group. My response was meant for
the following remark:

 

"Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
are still competition? I mean, you got people sitting around going "hmm how
can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?"

 

I was merely trying to ask a question to see if it's possible. I didn't mean
to cause a problem on this user list. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, April 27, 2010 4:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

 

Ok so I guess that means you don't want to sponsor ClueCon......

 

What's wrong with it is that is your problem not ours.

Our problem is to write a full featured soft switch for free for everyone to
use not just you.

 

Wanting to make FreeSWITCH the world's best kept secret is not on our
roadmap so we don't rush out to teach everyone how to hide it but if you
look hard enough you'll probably figure it out.

 

 

 

 

On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

We have competitors using vendor solutions that are expensive as hell. So,
we are trying to counter with a less expensive but powerful solution. What
exactly is wrong with that? And what is wrong with trying to extend the
barrier to entry for our competitors as long as possible? Not sure why
there's such a mystery here...





-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Giagnocavo
Sent: Tuesday, April 27, 2010 3:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
still competition? I mean, you got people sitting around going "hmm how can
bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Tuesday, April 27, 2010 12:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Very true. We're just trying to stay one step ahead of the local
competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real name
and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure
> the competition doesn't know what we're doing for as long as possible.
> That's all.
>
>
>
> Ken

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

_______________________________________________
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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


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From msc at freeswitch.org  Tue Apr 27 14:40:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Apr 2010 14:40:33 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call April 28
Message-ID: <l2n87f2f3b91004271440x4ed4d068wb5d7d2e9abf57fc8@mail.gmail.com>

Hello all!

I've posted the agenda for tomorrow's call:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28

Please get in there and add your items. It's been a bit hectic for me of
late so I appreciate any and all contributors to the agenda. If you have a
topic to discuss or a question to ask then definitely put it on the agenda
so that we can be ready for it.

Don't forget, we've pushed the meeting back one hour, so it starts at 1PM
Eastern, 10AM Pacific (1700 UTC)

See you all tomorrow!
-Michael
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From mgg at giagnocavo.net  Tue Apr 27 15:06:11 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 27 Apr 2010 18:06:11 -0400
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <019201cae651$6b38bb60$41aa3220$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
	<019201cae651$6b38bb60$41aa3220$@com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>

And I'm merely poking fun at the concept of competitors serious enough to pose a threat, yet dumb enough not to know how to use Google.

Honestly, if your competition is sitting there saying "oh well, I guess we gotta dish out half a mil for NexTone" and the only thing that changes their mind is "oh wow, we saw the SDP of Ken's company and it says freeswitch - yey, we can save tons of money!" - well that's some funny competitors and I can't see them posing any threat regardless if you gave them all your secrets.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Fulmer
Sent: Tuesday, April 27, 2010 3:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP


Sorry if I came across harshly toward the group. My response was meant for the following remark:


"Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are still competition? I mean, you got people sitting around going "hmm how can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?"



I was merely trying to ask a question to see if it's possible. I didn't mean to cause a problem on this user list.



Ken



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Tuesday, April 27, 2010 4:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Ok so I guess that means you don't want to sponsor ClueCon......

What's wrong with it is that is your problem not ours.
Our problem is to write a full featured soft switch for free for everyone to use not just you.

Wanting to make FreeSWITCH the world's best kept secret is not on our roadmap so we don't rush out to teach everyone how to hide it but if you look hard enough you'll probably figure it out.




On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com<mailto:kenfulmer at icstechnologysolutions.com>> wrote:
We have competitors using vendor solutions that are expensive as hell. So,
we are trying to counter with a less expensive but powerful solution. What
exactly is wrong with that? And what is wrong with trying to extend the
barrier to entry for our competitors as long as possible? Not sure why
there's such a mystery here...




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael
Giagnocavo
Sent: Tuesday, April 27, 2010 3:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
still competition? I mean, you got people sitting around going "hmm how can
bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Ken
Fulmer
Sent: Tuesday, April 27, 2010 12:48 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Very true. We're just trying to stay one step ahead of the local
competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real name
and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com<mailto:kenfulmer at icstechnologysolutions.com>> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure
> the competition doesn't know what we're doing for as long as possible.
> That's all.
>
>
>
> Ken

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

_______________________________________________
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<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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From janvb at live.com  Tue Apr 27 15:07:37 2010
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Apr 2010 00:07:37 +0200
Subject: [Freeswitch-users] E3 Card on Freeswitch ?
In-Reply-To: <s2r191c3a031004271409y45cdc458jcb7bf7a4c6d6c72c@mail.gmail.com>
References: <h2wda3229331004270743p8aa2af5dieadf9e53e5f5194b@mail.gmail.com>,
	<k2q4468a6771004271318i5421b25dtd3bacbbbbc52c524@mail.gmail.com>,
	<SNT135-w43BDDDA13986C0378B893AD2030@phx.gbl>,
	<s2r191c3a031004271409y45cdc458jcb7bf7a4c6d6c72c@mail.gmail.com>
Message-ID: <SNT135-w44ACB4F0677A85EBEAA8C0D2030@phx.gbl>


Nah - We get one of the larger Sun/Solaris boxes and sufficient supply of beer and pitza and miracles will happen :)
 


Date: Tue, 27 Apr 2010 16:09:50 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] E3 Card on Freeswitch ?

Not on one box though! We can only work so many miracles. ;)





On Tue, Apr 27, 2010 at 4:01 PM, Jan Berger <janvb at live.com> wrote:


Actually she could need an STM-4 
 


Date: Tue, 27 Apr 2010 22:18:50 +0200
From: oseslija at gmail.com

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] E3 Card on Freeswitch ?




E3 afaik is 34 Mbit/s. I think when channelized, you can get max of 16 E1s. 
16 x 30 = 480 calls max, so 6000 calls can't fit.
You will need a channelized STM-1 I think.

O.




On Tue, Apr 27, 2010 at 4:43 PM, Anita Hall <anita.hall at simmortel.com> wrote:

Hi


Please check out this product

http://www.sangoma.com/products/hardware_products/data_networking/a301.html

Does it work on Freeswitch or Asterisk ?
Do Telcos provide an E3 connection ?

One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ?

Thanks for any input you may provide.

regards,

Anita Hall,
Simmortel Voice.


_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Hotmail: Free, trusted and rich email service. Get it now.
_______________________________________________
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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



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
_________________________________________________________________
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From kris at kriskinc.com  Tue Apr 27 15:17:39 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 27 Apr 2010 18:17:39 -0400
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <019201cae651$6b38bb60$41aa3220$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
	<019201cae651$6b38bb60$41aa3220$@com>
Message-ID: <o2w4d15ff861004271517tbaff39a7pbd0a1c199df2f9af@mail.gmail.com>

Ken,

  You'll find in many technical communities (this is certainly one of
them) people are usually more interested in the "why" behind a
question than the question itself.

  This is certainly one of those cases.  You'll noticed no one (myself
included) responded with anything constructive.  While I can see how
that is frustrating I'll try to explain where this comes from.

  It would certainly seem more efficient to just answer individual
questions as quickly and succinctly as possible:

Q: "How do I change FreeSWITCH in the SDP?"
A: "Edit line xxx in mod_sofia.c" (or whatever)."

  The problem is this direct question/answer model often causes
behavior undesired by the person asking the question.  In this case,
it's quite likely you'd come back to the list in a couple of weeks
with something like:

Q: "When calling device XXX from manufacturer YYY I'm experiencing the
following strange audio problems..."

  Unless the list regulars happen to remember who you are and your
previous question about modifying the SDP it might not be immediately
obvious that your problem is caused by your previous source code
modification.  A SIP trace showing the modified SDP may give some
hints but it's unlikely that would be obvious as the source of your
audio problems.  This would cause you unnecessary strife and probably
bewilder more than a few list participants.

  The "veterans" Michael referred to have seen this play out time and
time again.  They (I'm not assigning myself any titles) know enough to
ALWAYS ask more questions.  They've seen platforms (FreeSWITCH
included) modify their behavior based on the remote endpoint (whether
identified by SIP User Agent or SDP origin).

  We're lucky they are around to share all of their experience and
beat us up when we get these crazy ideas in our heads ;).

On Tue, Apr 27, 2010 at 5:34 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> Sorry if I came across harshly toward the group. My response was meant for
> the following remark:
>
>
>
> ?Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are still competition? I mean, you got people sitting around going "hmm how
> can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"??
>
>
>
> I was merely trying to ask a question to see if it?s possible. I didn?t mean
> to cause a problem on this user list.
>
>
>
> Ken
>

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


From kenfulmer at icstechnologysolutions.com  Tue Apr 27 15:35:33 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 27 Apr 2010 17:35:33 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>
References: <00a601cae61a$ab89bf70$029d3e50$@com>	<F47BC738-DBE4-4012-A2ED-0E42B6229A3A@freeswitch.org>	<00fb01cae623$9f364be0$dda2e3a0$@com>	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>	<014e01cae63a$32fcf250$98f6d6f0$@com>	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>	<017901cae64b$da496820$8edc3860$@com>	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>	<019201cae651$6b38bb60$41aa3220$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>
Message-ID: <01b701cae659$f3a2c8d0$dae85a70$@com>

I can see your point. They probably won't be churning out new Freeswitch
systems anytime soon, regardless of what we're doing / not doing. 

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Giagnocavo
Sent: Tuesday, April 27, 2010 5:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

 

And I'm merely poking fun at the concept of competitors serious enough to
pose a threat, yet dumb enough not to know how to use Google.

 

Honestly, if your competition is sitting there saying "oh well, I guess we
gotta dish out half a mil for NexTone" and the only thing that changes their
mind is "oh wow, we saw the SDP of Ken's company and it says freeswitch -
yey, we can save tons of money!" - well that's some funny competitors and I
can't see them posing any threat regardless if you gave them all your
secrets.

 

-Michael

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Tuesday, April 27, 2010 3:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

 

Sorry if I came across harshly toward the group. My response was meant for
the following remark:

 

"Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
are still competition? I mean, you got people sitting around going "hmm how
can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?"

 

I was merely trying to ask a question to see if it's possible. I didn't mean
to cause a problem on this user list. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, April 27, 2010 4:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

 

Ok so I guess that means you don't want to sponsor ClueCon......

 

What's wrong with it is that is your problem not ours.

Our problem is to write a full featured soft switch for free for everyone to
use not just you.

 

Wanting to make FreeSWITCH the world's best kept secret is not on our
roadmap so we don't rush out to teach everyone how to hide it but if you
look hard enough you'll probably figure it out.

 

 

 

 

On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

We have competitors using vendor solutions that are expensive as hell. So,
we are trying to counter with a less expensive but powerful solution. What
exactly is wrong with that? And what is wrong with trying to extend the
barrier to entry for our competitors as long as possible? Not sure why
there's such a mystery here...





-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Giagnocavo
Sent: Tuesday, April 27, 2010 3:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
still competition? I mean, you got people sitting around going "hmm how can
bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Tuesday, April 27, 2010 12:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

Very true. We're just trying to stay one step ahead of the local
competition.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, April 27, 2010 12:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP

While I understand the request:

1)  They are going to find out anyway.
2)  You are posting to FreeSWITCH-users using (presumably) your real name
and corporate e-mail address.

If you were looking to hide FreeSWITCH it's a little too late.

On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> It's not meant to be offensive or anything. We'd just like to ensure
> the competition doesn't know what we're doing for as long as possible.
> That's all.
>
>
>
> Ken

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


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




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From codecomplete at free.fr  Tue Apr 27 16:56:44 2010
From: codecomplete at free.fr (GillesToo)
Date: Tue, 27 Apr 2010 16:56:44 -0700 (PDT)
Subject: [Freeswitch-users] Good book on (VoIP) telephony?
Message-ID: <1272412604620-4971798.post@n2.nabble.com>


Hello

I was wondering: For those of you experienced Freeswitch users, is there a
good book on telephony that you would recommend for software people who have
never worked in the telephony business and would provide a good basis to
make sense of VoIP in general, and Freeswitch in particular?

FWIW, here's what Amazon returns with "telephony":

www.amazon.com/s/qid=1272412320/ref=sr_pg_2?keywords=telephony&rh=n:!1000,i:stripbooks,k:telephony

Thank  you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Good-book-on-VoIP-telephony-tp4971798p4971798.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From edpimentl at gmail.com  Tue Apr 27 17:48:36 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 27 Apr 2010 20:48:36 -0400
Subject: [Freeswitch-users] Good book on (VoIP) telephony?
In-Reply-To: <1272412604620-4971798.post@n2.nabble.com>
References: <1272412604620-4971798.post@n2.nabble.com>
Message-ID: <x2u9dc4a1671004271748lafb75ea8s1c8cb69708a9e1ed@mail.gmail.com>

If you are a new at VoIP look into these publications:
http://www.amazon.com/SIP-Understanding-Initiation-Protocol-Telecommunications/dp/1607839954/ref=sr_1_1?ie=UTF8&s=books&qid=1272415143&sr=1-1

http://www.amazon.com/SIP-Beyond-VoIP-Communications-Revolution/dp/0974813001/ref=sr_1_10?ie=UTF8&s=books&qid=1272415143&sr=1-10

http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741/ref=sr_1_12/189-7099384-6630443?ie=UTF8&s=books&qid=1272415088&sr=1-12

http://www.amazon.com/Understanding-Servlets-Artech-House-Telecommunications/dp/159693428X/ref=sr_1_9?ie=UTF8&s=books&qid=1272415143&sr=1-9

http://www.amazon.com/Switching-VoIP-Theodore-Wallingford/dp/0596008686/ref=sr_1_1?ie=UTF8&s=books&qid=1272415530&sr=1-1

Or look into these Google eBooks links
http://books.google.com/books?id=9_wRFy5OGw4C&dq=asterisk+book&printsec=frontcover&source=bn&hl=en&ei=mYXXS6DeG4bc8ASul_SSBw&sa=X&oi=book_result&ct=result&resnum=4&ved=0CCsQ6AEwAw#v=onepage&q=asterisk%20book&f=false

-E
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From vfclists at googlemail.com  Tue Apr 27 18:21:30 2010
From: vfclists at googlemail.com (Frank Church)
Date: Wed, 28 Apr 2010 02:21:30 +0100
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <01b701cae659$f3a2c8d0$dae85a70$@com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
	<019201cae651$6b38bb60$41aa3220$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>
	<01b701cae659$f3a2c8d0$dae85a70$@com>
Message-ID: <h2t7abab2411004271821h45eac761s2bd738f91ad22833@mail.gmail.com>

There is a point to hiding the UA. Recently a system I developed based
on you know what, was triggering a crash in a provider's system after
a few calls were made. It would all come to halt. Their UA used their
company's name and it was only later I realized that it was 'YKW.

In such a case no one could claim that it was because they were using
'YKW' that their system was so unstable. They did upgrade their system
to the latest version of 'YKW' and the problem went away. I also made
some changes in my systems NATing. To date I don't know whether the
change I made stopped triggering the crashing, or whether it the
upgrade they made that caused it to stop.

But I don't think that is what Ken is trying to avoid.

The problem is if there are some known interopability issues, hiding
the UA name will not help diagnosis.

On 27 April 2010 23:35, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:
> I can see your point. They probably won?t be churning out new Freeswitch
> systems anytime soon, regardless of what we?re doing / not doing.
>
>
>
> Ken
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Giagnocavo
> Sent: Tuesday, April 27, 2010 5:06 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
>
>
> And I?m merely poking fun at the concept of competitors serious enough to
> pose a threat, yet dumb enough not to know how to use Google.
>
>
>
> Honestly, if your competition is sitting there saying ?oh well, I guess we
> gotta dish out half a mil for NexTone? and the only thing that changes their
> mind is ?oh wow, we saw the SDP of Ken?s company and it says freeswitch ?
> yey, we can save tons of money!? ? well that?s some funny competitors and I
> can?t see them posing any threat regardless if you gave them all your
> secrets.
>
>
>
> -Michael
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> Fulmer
> Sent: Tuesday, April 27, 2010 3:34 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
>
>
> Sorry if I came across harshly toward the group. My response was meant for
> the following remark:
>
>
>
> ?Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are still competition? I mean, you got people sitting around going "hmm how
> can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"??
>
>
>
> I was merely trying to ask a question to see if it?s possible. I didn?t mean
> to cause a problem on this user list.
>
>
>
> Ken
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Tuesday, April 27, 2010 4:08 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
>
>
> Ok so I guess that means you don't want to sponsor ClueCon......
>
>
>
> What's wrong with it is that is your problem not ours.
>
> Our problem is to write a full featured soft switch for free for everyone to
> use not just you.
>
>
>
> Wanting to make FreeSWITCH the world's best kept secret is not on our
> roadmap so we don't rush out to teach everyone how to hide it but if you
> look hard enough you'll probably figure it out.
>
>
>
>
>
>
>
>
>
> On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
>
> We have competitors using vendor solutions that are expensive as hell. So,
> we are trying to counter with a less expensive but powerful solution. What
> exactly is wrong with that? And what is wrong with trying to extend the
> barrier to entry for our competitors as long as possible? Not sure why
> there's such a mystery here...
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Giagnocavo
> Sent: Tuesday, April 27, 2010 3:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
> still competition? I mean, you got people sitting around going "hmm how can
> bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> Fulmer
> Sent: Tuesday, April 27, 2010 12:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> Very true. We're just trying to stay one step ahead of the local
> competition.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
> Kielhofner
> Sent: Tuesday, April 27, 2010 12:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>
> While I understand the request:
>
> 1) ?They are going to find out anyway.
> 2) ?You are posting to FreeSWITCH-users using (presumably) your real name
> and corporate e-mail address.
>
> If you were looking to hide FreeSWITCH it's a little too late.
>
> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
> <kenfulmer at icstechnologysolutions.com> wrote:
>> It's not meant to be offensive or anything. We'd just like to ensure
>> the competition doesn't know what we're doing for as long as possible.
>> That's all.
>>
>>
>>
>> Ken
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From btsteve at yahoo.com  Tue Apr 27 19:29:48 2010
From: btsteve at yahoo.com (Travis Stevens)
Date: Tue, 27 Apr 2010 19:29:48 -0700 (PDT)
Subject: [Freeswitch-users] need registration help
Message-ID: <95245.27320.qm@web30201.mail.mud.yahoo.com>

 I am trying to register a sip account from one freeswitch (PBX) to an 
ITSP running freeswitch. The freeswitch at the ITSP tries to route the 
call but the contact info has prepended gw+ to the sip account 
registration. Is there a way to stop the gw+ from being prepended?


      


From brian at freeswitch.org  Tue Apr 27 19:36:12 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Apr 2010 21:36:12 -0500
Subject: [Freeswitch-users] need registration help
In-Reply-To: <95245.27320.qm@web30201.mail.mud.yahoo.com>
References: <95245.27320.qm@web30201.mail.mud.yahoo.com>
Message-ID: <F08BCB3E-B7FA-4A34-9262-BD0AE7A07839@freeswitch.org>

It shouldn't matter... anyway the gateway param is 'extension-in-contact'.  Someone should write a wiki page this.

/b

On Apr 27, 2010, at 9:29 PM, Travis Stevens wrote:

> I am trying to register a sip account from one freeswitch (PBX) to an 
> ITSP running freeswitch. The freeswitch at the ITSP tries to route the 
> call but the contact info has prepended gw+ to the sip account 
> registration. Is there a way to stop the gw+ from being prepended?
> 



From dule.maillist at gmail.com  Tue Apr 27 19:47:06 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Tue, 27 Apr 2010 22:47:06 -0400
Subject: [Freeswitch-users] SIP Headers (e.g., Retry-After)
Message-ID: <l2u914fc92a1004271947y4e3e37dgee6a4b9aefbbf81e@mail.gmail.com>

Is there a way to get SIP headers like the Retry-After field in FreeSWITCH?
Essentially we want to parse the Retry-After field and react based on the
value for when we get 503s because of congestion or what not.

Thanks,
Dan
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From boris at tagnet.ru  Tue Apr 27 20:17:42 2010
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 28 Apr 2010 09:17:42 +0600
Subject: [Freeswitch-users] mod_say_ru says time +1 hour of current time
Message-ID: <4BD7A8D6.2040105@tagnet.ru>

Hello!

     After looking to MODAPP-421 Michael Jerris wrote to me: "set 
timezone channel var" and closed the issue. So I changed my extension to:
<extension name="SayTime">
<condition field="destination_number" expression="^9000$">
<action application="answer"/>
<action application="set" data="timezone=Asia/Yekaterinburg"/>
<action application="set" data="ct=${strftime_tz(Asia/Yekaterinburg %s)}"/>
<action application="log" data="INFO TZ: ${timezone}"/>
<action application="log" data="INFO CT: ${ct}"/>
<action application="say" data="ru current_date_time pronounced ${ct}"/>
<action application="sleep" data="1000"/>
<action application="hangup"/>
</condition>
</extension>

But pronounced time is still +1 hour (but ${ct} shows right time). 
What's wrong with my extension? Have I misunderstood Michael and need to 
set channel var in another place?

-- 
Respect,
Boris




From telteclistas at gmail.com  Tue Apr 27 20:27:56 2010
From: telteclistas at gmail.com (leonardo alves)
Date: Tue, 27 Apr 2010 23:27:56 -0400
Subject: [Freeswitch-users] Error with media ptime
Message-ID: <s2naa768f581004272027m15a62473y9defe9764f48ac18@mail.gmail.com>

Hello,

I am new to freeswitch and I have just installed the last version of
freeswitch. I am doing some tests dialing with sip and when the call is
answer I play an file.
If I do 10 calls, in 5 or 6 of them I get this error in the console and the
audio gets cut like with it was a bandwithd problem.

2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use
ptime 20 but what they meant to say was 30
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who
knows what will happen.

Does anyone knows if there is a way to fix this issue ?

Thanks
Leonardo
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From telteclistas at gmail.com  Tue Apr 27 21:19:16 2010
From: telteclistas at gmail.com (leonardo alves)
Date: Wed, 28 Apr 2010 00:19:16 -0400
Subject: [Freeswitch-users] Error with media ptime
In-Reply-To: <s2naa768f581004272027m15a62473y9defe9764f48ac18@mail.gmail.com>
References: <s2naa768f581004272027m15a62473y9defe9764f48ac18@mail.gmail.com>
Message-ID: <p2haa768f581004272119qe1b44579yb07c2d0d63e895ed@mail.gmail.com>

Sorry I found this message:
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg10032.html
and the
<param name="inbound-codec-negotiation" value="scrooge"/>
fixed my problem.

Now I have another question.
What exactly this "scrooge" does ? Is this going to affect the other
providers that was working ?
Thanks
Leonardo



On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <telteclistas at gmail.com>wrote:

> Hello,
>
> I am new to freeswitch and I have just installed the last version of
> freeswitch. I am doing some tests dialing with sip and when the call is
> answer I play an file.
> If I do 10 calls, in 5 or 6 of them I get this error in the console and the
> audio gets cut like with it was a bandwithd problem.
>
> 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use
> ptime 20 but what they meant to say was 30
> This issue has so far been identified to happen on the following broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so broken
> who knows what will happen.
>
> Does anyone knows if there is a way to fix this issue ?
>
> Thanks
> Leonardo
>
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From msc at freeswitch.org  Tue Apr 27 22:22:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Apr 2010 22:22:21 -0700
Subject: [Freeswitch-users] Good book on (VoIP) telephony?
In-Reply-To: <x2u9dc4a1671004271748lafb75ea8s1c8cb69708a9e1ed@mail.gmail.com>
References: <1272412604620-4971798.post@n2.nabble.com>
	<x2u9dc4a1671004271748lafb75ea8s1c8cb69708a9e1ed@mail.gmail.com>
Message-ID: <y2m87f2f3b91004272222tecf30eep37daa7ff0bdb8cbf@mail.gmail.com>

I normally don't like "for dummies" books but honestly I have to recommend
this one:
http://www.amazon.com/VoIP-Deployment-Dummies-Stephen-Olejniczak/dp/047038543X/ref=pd_sim_b_2

In spite of the annoying typos ("Secession Initiation Protocol", "Secession
Description Protocol", "Session Boarder Controller") I found it to be very
handy for someone learning about SIP and where it fits into the LAN, WAN,
etc. plus some handy troubleshooting tips and a nice section on using
Wireshark to debug VoIP calls.

-MC

On Tue, Apr 27, 2010 at 5:48 PM, EdPimentl <edpimentl at gmail.com> wrote:

> If you are a new at VoIP look into these publications:
>
> http://www.amazon.com/SIP-Understanding-Initiation-Protocol-Telecommunications/dp/1607839954/ref=sr_1_1?ie=UTF8&s=books&qid=1272415143&sr=1-1
>
>
> http://www.amazon.com/SIP-Beyond-VoIP-Communications-Revolution/dp/0974813001/ref=sr_1_10?ie=UTF8&s=books&qid=1272415143&sr=1-10
>
>
> http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741/ref=sr_1_12/189-7099384-6630443?ie=UTF8&s=books&qid=1272415088&sr=1-12
>
>
> http://www.amazon.com/Understanding-Servlets-Artech-House-Telecommunications/dp/159693428X/ref=sr_1_9?ie=UTF8&s=books&qid=1272415143&sr=1-9
>
>
> http://www.amazon.com/Switching-VoIP-Theodore-Wallingford/dp/0596008686/ref=sr_1_1?ie=UTF8&s=books&qid=1272415530&sr=1-1
>
> Or look into these Google eBooks links
>
> http://books.google.com/books?id=9_wRFy5OGw4C&dq=asterisk+book&printsec=frontcover&source=bn&hl=en&ei=mYXXS6DeG4bc8ASul_SSBw&sa=X&oi=book_result&ct=result&resnum=4&ved=0CCsQ6AEwAw#v=onepage&q=asterisk%20book&f=false
>
> -E
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From btsteve at yahoo.com  Tue Apr 27 22:25:15 2010
From: btsteve at yahoo.com (Travis Stevens)
Date: Tue, 27 Apr 2010 22:25:15 -0700 (PDT)
Subject: [Freeswitch-users] need registration help
In-Reply-To: <F08BCB3E-B7FA-4A34-9262-BD0AE7A07839@freeswitch.org>
References: <95245.27320.qm@web30201.mail.mud.yahoo.com>
	<F08BCB3E-B7FA-4A34-9262-BD0AE7A07839@freeswitch.org>
Message-ID: <229890.16054.qm@web30204.mail.mud.yahoo.com>

Thanks Brian. 
Worked like a charm. 

Added to the wiki. Hope i did it correctly.




----- Original Message ----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, April 27, 2010 10:36:12 PM
Subject: Re: [Freeswitch-users] need registration help

It shouldn't matter... anyway the gateway param is 'extension-in-contact'.  Someone should write a wiki page this.

/b

On Apr 27, 2010, at 9:29 PM, Travis Stevens wrote:

> I am trying to register a sip account from one freeswitch (PBX) to an 
> ITSP running freeswitch. The freeswitch at the ITSP tries to route the 
> call but the contact info has prepended gw+ to the sip account 
> registration. Is there a way to stop the gw+ from being prepended?
> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Wed Apr 28 00:57:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Apr 2010 03:57:00 -0400
Subject: [Freeswitch-users] session.steamFile misses DTMF event for
	first few seconds
In-Reply-To: <u2r191c3a031004270840o6948472blfa90e9ca0a238914@mail.gmail.com>
References: <b33e001e8bfb3b4d583205a643e7ecd2.squirrel@correo.nodo50.org>
	<788C79CD1E584493B92EB90A5EA474DB@ws4>
	<u2r191c3a031004270840o6948472blfa90e9ca0a238914@mail.gmail.com>
Message-ID: <D4AEB763-AD2D-411E-AA3A-55D62E99DE2F@jerris.com>

The svn repo is updated again now, but it will tend to lag (about a day) behind git going forward.

Mike

On Apr 27, 2010, at 11:40 AM, Anthony Minessale wrote:

> 12XXX is so many years old, i wish users who want free help would at least stay up to date with the code.
> 
> FYI, our repo is on git now and the svn mirror is not updating at the moment.
> 
> produce a complete minimal script that reproduces your problem and can be run on git HEAD (see download instructions to learn how to build with git)  Use existing sound files from the FS install so we can just run it in our lab to reproduce the issue.
> 
> Open an issue on http://jira.freeswitch.org and attach the script.
> 
> 
> 
> On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank at impactfax.com> wrote:
> The calls are coming from land based lines.  Traditional POTS (not cable
> company).  Caller is calling into FS and providing those DTMF.  I can
> reproduce on my POTS line (3000' from CO) and there is no discernable
> noise on the line.
> 
> The call comes into the media gateway and then goes sip to FS.
> 
> I can reproduce the problem just by starting FS version from the latest
> trunk.  And then I can eliminate the problem by restarting the FS
> version 12790.  In both test cases, the media gateway remains constant.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Alberto Escudero
> Sent: Tuesday, April 27, 2010 9:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
> first few seconds
> 
> How are you  receiving those DTMFs, inbound or outbound? Are those calls
> coming from a mobile network (GSM). We have experienced lots of problem
> with DTMF detection in noisy lines.
> 
> /aep
> --
> Stopping junk mailers is good for the environment
> 
> > I recently upgraded from FS 12790M to svn 17188.  When I did, I
> noticed
> > that session.streamFile behaved differently and I started having
> > problems with my IVR app.
> >
> > With the upgraded FS, I have a problem with streamFile no firing on
> the
> > DTMF and calling the callback function for the first few seconds of
> the
> > wav file playback.  It behaves as though it does not hear the DTMFs.
> If
> > I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> > catches the DTMF and all is well.  If I key as soon as I hear the wav
> > file start, streamFile just keeps playing the wav and does not call
> the
> > callback function.
> >
> > When I revert back to the previous version of FS, streamFile always
> > fires the callback right away no matter how quickly I press the first
> > DTMF as the wav file starts to stream out.
> >
> > The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> > 16 bit, mono 8000 Hz
> >
> > The snippet of js code I am using is as follows.
> >
> > if(session.ready()) {
> >     session.answer();
> >     session.sleep(750);
> >     while(session.ready()) {
> >         session.sleep(500);
> >         session.flushDigits(); // clear out input buffers
> >
> >
> >
> if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> > ts_cb,""))===false) {
> >             pin=session.getDigits(pinmax,pinterm,pinwait);
> >         } else {
> >             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
> >         }
> >         // more code here..
> >     }
> >
> > Do I need to change the way I use streamFile in the later release?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Apr 28 01:09:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Apr 2010 04:09:50 -0400
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <h2t7abab2411004271821h45eac761s2bd738f91ad22833@mail.gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<00fb01cae623$9f364be0$dda2e3a0$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
	<019201cae651$6b38bb60$41aa3220$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>
	<01b701cae659$f3a2c8d0$dae85a70$@com>
	<h2t7abab2411004271821h45eac761s2bd738f91ad22833@mail.gmail.com>
Message-ID: <F35A1668-D0CA-434E-8ABA-8BAAA1F1B5F3@jerris.com>

It's worse than that, we have a bunch of code that enables special behavior when freeswitch talks to itself or to other known broken devices.  You break that capability by changing this information.

Mike

On Apr 27, 2010, at 9:21 PM, Frank Church wrote:

> There is a point to hiding the UA. Recently a system I developed based
> on you know what, was triggering a crash in a provider's system after
> a few calls were made. It would all come to halt. Their UA used their
> company's name and it was only later I realized that it was 'YKW.
> 
> In such a case no one could claim that it was because they were using
> 'YKW' that their system was so unstable. They did upgrade their system
> to the latest version of 'YKW' and the problem went away. I also made
> some changes in my systems NATing. To date I don't know whether the
> change I made stopped triggering the crashing, or whether it the
> upgrade they made that caused it to stop.
> 
> But I don't think that is what Ken is trying to avoid.
> 
> The problem is if there are some known interopability issues, hiding
> the UA name will not help diagnosis.
> 
> On 27 April 2010 23:35, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:
>> I can see your point. They probably won?t be churning out new Freeswitch
>> systems anytime soon, regardless of what we?re doing / not doing.
>> 
>> 
>> 
>> Ken
>> 
>> 
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
>> Giagnocavo
>> Sent: Tuesday, April 27, 2010 5:06 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> 
>> 
>> And I?m merely poking fun at the concept of competitors serious enough to
>> pose a threat, yet dumb enough not to know how to use Google.
>> 
>> 
>> 
>> Honestly, if your competition is sitting there saying ?oh well, I guess we
>> gotta dish out half a mil for NexTone? and the only thing that changes their
>> mind is ?oh wow, we saw the SDP of Ken?s company and it says freeswitch ?
>> yey, we can save tons of money!? ? well that?s some funny competitors and I
>> can?t see them posing any threat regardless if you gave them all your
>> secrets.
>> 
>> 
>> 
>> -Michael
>> 
>> 
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
>> Fulmer
>> Sent: Tuesday, April 27, 2010 3:34 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> 
>> 
>> Sorry if I came across harshly toward the group. My response was meant for
>> the following remark:
>> 
>> 
>> 
>> ?Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
>> are still competition? I mean, you got people sitting around going "hmm how
>> can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"??
>> 
>> 
>> 
>> I was merely trying to ask a question to see if it?s possible. I didn?t mean
>> to cause a problem on this user list.
>> 
>> 
>> 
>> Ken
>> 
>> 
>> 
>> 
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
>> Minessale
>> Sent: Tuesday, April 27, 2010 4:08 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> 
>> 
>> Ok so I guess that means you don't want to sponsor ClueCon......
>> 
>> 
>> 
>> What's wrong with it is that is your problem not ours.
>> 
>> Our problem is to write a full featured soft switch for free for everyone to
>> use not just you.
>> 
>> 
>> 
>> Wanting to make FreeSWITCH the world's best kept secret is not on our
>> roadmap so we don't rush out to teach everyone how to hide it but if you
>> look hard enough you'll probably figure it out.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer
>> <kenfulmer at icstechnologysolutions.com> wrote:
>> 
>> We have competitors using vendor solutions that are expensive as hell. So,
>> we are trying to counter with a less expensive but powerful solution. What
>> exactly is wrong with that? And what is wrong with trying to extend the
>> barrier to entry for our competitors as long as possible? Not sure why
>> there's such a mystery here...
>> 
>> 
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
>> Giagnocavo
>> Sent: Tuesday, April 27, 2010 3:39 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> Wait, so your "competition" can't figure out to use FreeSWITCH, yet they are
>> still competition? I mean, you got people sitting around going "hmm how can
>> bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
>> 
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
>> Fulmer
>> Sent: Tuesday, April 27, 2010 12:48 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> Very true. We're just trying to stay one step ahead of the local
>> competition.
>> 
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
>> Kielhofner
>> Sent: Tuesday, April 27, 2010 12:37 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
>> 
>> While I understand the request:
>> 
>> 1)  They are going to find out anyway.
>> 2)  You are posting to FreeSWITCH-users using (presumably) your real name
>> and corporate e-mail address.
>> 
>> If you were looking to hide FreeSWITCH it's a little too late.
>> 
>> On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
>> <kenfulmer at icstechnologysolutions.com> wrote:
>>> It's not meant to be offensive or anything. We'd just like to ensure
>>> the competition doesn't know what we're doing for as long as possible.
>>> That's all.
>>> 
>>> 
>>> 
>>> Ken
>> 
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.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
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Frank Church
> 
> =======================
> http://devblog.brahmancreations.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 patrick at speechpro.com  Wed Apr 28 01:49:41 2010
From: patrick at speechpro.com (patrick)
Date: Wed, 28 Apr 2010 12:49:41 +0400
Subject: [Freeswitch-users] Stupid question ;-)
Message-ID: <4BD7F6A5.8090803@speechpro.com>

I have some var:
<action application="set"        data="input_fail_counter=0"/>

And in some moment I want to increase it by 1. I try this way:
<action application="set"        
data="input_fail_counter=${input_fail_counter}+1"/>
And have result input_fail_counter=0+1  i.e. just string merge...

How to handle it right way in dialplan?

-- 
? ?????????,

?????????? ?????? ??????????
??????? ?? ????????????
??? ?????? ??????? ???????????
???.: (812) 325-8848, ???. 6736
????: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru



From cstomi.levlist at gmail.com  Wed Apr 28 02:08:19 2010
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Wed, 28 Apr 2010 11:08:19 +0200
Subject: [Freeswitch-users] Stupid question ;-)
In-Reply-To: <4BD7F6A5.8090803@speechpro.com>
References: <4BD7F6A5.8090803@speechpro.com>
Message-ID: <4BD7FB03.4090705@gmail.com>

I think you should try expr api from mod_expr
Try this: <action application="set" 
data="input_fail_counter=${expr(${input_fail_counter}+1)}"/>


patrick wrote:
> I have some var:
> <action application="set"        data="input_fail_counter=0"/>
>
> And in some moment I want to increase it by 1. I try this way:
> <action application="set"        
> data="input_fail_counter=${input_fail_counter}+1"/>
> And have result input_fail_counter=0+1  i.e. just string merge...
>
> How to handle it right way in dialplan?
>
>   





From patrick at speechpro.com  Wed Apr 28 02:30:21 2010
From: patrick at speechpro.com (patrick)
Date: Wed, 28 Apr 2010 13:30:21 +0400
Subject: [Freeswitch-users] Stupid question ;-)
In-Reply-To: <4BD7FB03.4090705@gmail.com>
References: <4BD7F6A5.8090803@speechpro.com> <4BD7FB03.4090705@gmail.com>
Message-ID: <4BD8002D.8080401@speechpro.com>

Thank you!!! It works!!!


Tamas Cseke ?????:
> I think you should try expr api from mod_expr
> Try this: <action application="set" 
> data="input_fail_counter=${expr(${input_fail_counter}+1)}"/>
>
>
> patrick wrote:
>   
>> I have some var:
>> <action application="set"        data="input_fail_counter=0"/>
>>
>> And in some moment I want to increase it by 1. I try this way:
>> <action application="set"        
>> data="input_fail_counter=${input_fail_counter}+1"/>
>> And have result input_fail_counter=0+1  i.e. just string merge...
>>
>> How to handle it right way in dialplan?
>>
>>   
>>     
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   

-- 
? ?????????,

?????????? ?????? ??????????
??????? ?? ????????????
??? ?????? ??????? ???????????
???.: (812) 325-8848, ???. 6736
????: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru



From frank at impactfax.com  Wed Apr 28 03:08:41 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Wed, 28 Apr 2010 06:08:41 -0400
Subject: [Freeswitch-users] session.steamFile misses DTMF event forfirst
	few seconds
In-Reply-To: <u2r191c3a031004270840o6948472blfa90e9ca0a238914@mail.gmail.com>
Message-ID: <5CE98178D9124659BC930EDDBF0ED450@ws4>

Yes. I understand your position of trying to help people using old code.
That is why I tried to download the latest code to try to stay current.
 
But that is also when I found this problem with the newer code.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Tuesday, April 27, 2010 11:40 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event
forfirst few seconds
 
12XXX is so many years old, i wish users who want free help would at
least stay up to date with the code.

FYI, our repo is on git now and the svn mirror is not updating at the
moment.

produce a complete minimal script that reproduces your problem and can
be run on git HEAD (see download instructions to learn how to build with
git)  Use existing sound files from the FS install so we can just run it
in our lab to reproduce the issue.

Open an issue on http://jira.freeswitch.org and attach the script.



On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank at impactfax.com>
wrote:
The calls are coming from land based lines.  Traditional POTS (not cable
company).  Caller is calling into FS and providing those DTMF.  I can
reproduce on my POTS line (3000' from CO) and there is no discernable
noise on the line.

The call comes into the media gateway and then goes sip to FS.

I can reproduce the problem just by starting FS version from the latest
trunk.  And then I can eliminate the problem by restarting the FS
version 12790.  In both test cases, the media gateway remains constant.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Alberto Escudero
Sent: Tuesday, April 27, 2010 9:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
first few seconds

How are you  receiving those DTMFs, inbound or outbound? Are those calls
coming from a mobile network (GSM). We have experienced lots of problem
with DTMF detection in noisy lines.

/aep
--
Stopping junk mailers is good for the environment

> I recently upgraded from FS 12790M to svn 17188.  When I did, I
noticed
> that session.streamFile behaved differently and I started having
> problems with my IVR app.
>
> With the upgraded FS, I have a problem with streamFile no firing on
the
> DTMF and calling the callback function for the first few seconds of
the
> wav file playback.  It behaves as though it does not hear the DTMFs.
If
> I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> catches the DTMF and all is well.  If I key as soon as I hear the wav
> file start, streamFile just keeps playing the wav and does not call
the
> callback function.
>
> When I revert back to the previous version of FS, streamFile always
> fires the callback right away no matter how quickly I press the first
> DTMF as the wav file starts to stream out.
>
> The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> 16 bit, mono 8000 Hz
>
> The snippet of js code I am using is as follows.
>
> if(session.ready()) {
>     session.answer();
>     session.sleep(750);
>     while(session.ready()) {
>         session.sleep(500);
>         session.flushDigits(); // clear out input buffers
>
>
>
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> ts_cb,""))===false) {
>             pin=session.getDigits(pinmax,pinterm,pinwait);
>         } else {
>             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
>         }
>         // more code here..
>     }
>
> Do I need to change the way I use streamFile in the later release?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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


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



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900
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From vfclists at googlemail.com  Wed Apr 28 05:15:28 2010
From: vfclists at googlemail.com (Frank Church)
Date: Wed, 28 Apr 2010 13:15:28 +0100
Subject: [Freeswitch-users] Difference between effective-caller-id and
	outbound-caller-id
Message-ID: <o2m7abab2411004280515wad8b1732t185e7b93ac2e37cd@mail.gmail.com>

What are the differences between the two and when is one more
appropriate than the other?

What are the use cases?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From anthony.minessale at gmail.com  Wed Apr 28 06:59:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 08:59:57 -0500
Subject: [Freeswitch-users] Replace "Freeswitch" in SDP
In-Reply-To: <h2t7abab2411004271821h45eac761s2bd738f91ad22833@mail.gmail.com>
References: <00a601cae61a$ab89bf70$029d3e50$@com>
	<r2y4d15ff861004271037j95ea2061yc09baf8d08301e28@mail.gmail.com>
	<014e01cae63a$32fcf250$98f6d6f0$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B096F@mse17be1.mse17.exchange.ms>
	<017901cae64b$da496820$8edc3860$@com>
	<p2l191c3a031004271408j81f95e7ckc4fa2a54bc4ecec4@mail.gmail.com>
	<019201cae651$6b38bb60$41aa3220$@com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B09B3@mse17be1.mse17.exchange.ms>
	<01b701cae659$f3a2c8d0$dae85a70$@com>
	<h2t7abab2411004271821h45eac761s2bd738f91ad22833@mail.gmail.com>
Message-ID: <t2p191c3a031004280659u636e1958g4e3fb8e1be12abe0@mail.gmail.com>

He already changed the UA name he wants to change the media name in the SDP
too.

We have huge problems with Sonus.  We managed to fix it by detecting it was
a Sonus in the SDP.
Some providers changed the SDP and all the bugs came back.  That is
something to avoid by leaving the SDP alone.

On Tue, Apr 27, 2010 at 8:21 PM, Frank Church <vfclists at googlemail.com>wrote:

> There is a point to hiding the UA. Recently a system I developed based
> on you know what, was triggering a crash in a provider's system after
> a few calls were made. It would all come to halt. Their UA used their
> company's name and it was only later I realized that it was 'YKW.
>
> In such a case no one could claim that it was because they were using
> 'YKW' that their system was so unstable. They did upgrade their system
> to the latest version of 'YKW' and the problem went away. I also made
> some changes in my systems NATing. To date I don't know whether the
> change I made stopped triggering the crashing, or whether it the
> upgrade they made that caused it to stop.
>
> But I don't think that is what Ken is trying to avoid.
>
> The problem is if there are some known interopability issues, hiding
> the UA name will not help diagnosis.
>
> On 27 April 2010 23:35, Ken Fulmer <kenfulmer at icstechnologysolutions.com>
> wrote:
> > I can see your point. They probably won?t be churning out new Freeswitch
> > systems anytime soon, regardless of what we?re doing / not doing.
> >
> >
> >
> > Ken
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> > Giagnocavo
> > Sent: Tuesday, April 27, 2010 5:06 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> >
> >
> > And I?m merely poking fun at the concept of competitors serious enough to
> > pose a threat, yet dumb enough not to know how to use Google.
> >
> >
> >
> > Honestly, if your competition is sitting there saying ?oh well, I guess
> we
> > gotta dish out half a mil for NexTone? and the only thing that changes
> their
> > mind is ?oh wow, we saw the SDP of Ken?s company and it says freeswitch ?
> > yey, we can save tons of money!? ? well that?s some funny competitors and
> I
> > can?t see them posing any threat regardless if you gave them all your
> > secrets.
> >
> >
> >
> > -Michael
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> > Fulmer
> > Sent: Tuesday, April 27, 2010 3:34 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> >
> >
> > Sorry if I came across harshly toward the group. My response was meant
> for
> > the following remark:
> >
> >
> >
> > ?Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> > are still competition? I mean, you got people sitting around going "hmm
> how
> > can bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"??
> >
> >
> >
> > I was merely trying to ask a question to see if it?s possible. I didn?t
> mean
> > to cause a problem on this user list.
> >
> >
> >
> > Ken
> >
> >
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony
> > Minessale
> > Sent: Tuesday, April 27, 2010 4:08 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> >
> >
> > Ok so I guess that means you don't want to sponsor ClueCon......
> >
> >
> >
> > What's wrong with it is that is your problem not ours.
> >
> > Our problem is to write a full featured soft switch for free for everyone
> to
> > use not just you.
> >
> >
> >
> > Wanting to make FreeSWITCH the world's best kept secret is not on our
> > roadmap so we don't rush out to teach everyone how to hide it but if you
> > look hard enough you'll probably figure it out.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On Tue, Apr 27, 2010 at 3:54 PM, Ken Fulmer
> > <kenfulmer at icstechnologysolutions.com> wrote:
> >
> > We have competitors using vendor solutions that are expensive as hell.
> So,
> > we are trying to counter with a less expensive but powerful solution.
> What
> > exactly is wrong with that? And what is wrong with trying to extend the
> > barrier to entry for our competitors as long as possible? Not sure why
> > there's such a mystery here...
> >
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> >
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> > Giagnocavo
> > Sent: Tuesday, April 27, 2010 3:39 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> > Wait, so your "competition" can't figure out to use FreeSWITCH, yet they
> are
> > still competition? I mean, you got people sitting around going "hmm how
> can
> > bet beat Ken, damn we can't- -- OH FREESWITCH NOW I GET IT!!!"?
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> > Fulmer
> > Sent: Tuesday, April 27, 2010 12:48 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> > Very true. We're just trying to stay one step ahead of the local
> > competition.
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian
> > Kielhofner
> > Sent: Tuesday, April 27, 2010 12:37 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Replace "Freeswitch" in SDP
> >
> > While I understand the request:
> >
> > 1)  They are going to find out anyway.
> > 2)  You are posting to FreeSWITCH-users using (presumably) your real name
> > and corporate e-mail address.
> >
> > If you were looking to hide FreeSWITCH it's a little too late.
> >
> > On Tue, Apr 27, 2010 at 12:06 PM, Ken Fulmer
> > <kenfulmer at icstechnologysolutions.com> wrote:
> >> It's not meant to be offensive or anything. We'd just like to ensure
> >> the competition doesn't know what we're doing for as long as possible.
> >> That's all.
> >>
> >>
> >>
> >> Ken
> >
> > --
> > Kristian Kielhofner
> > http://www.astlinux.org
> > http://blog.krisk.org
> > http://www.star2star.com
> > http://www.submityoursip.com
> > http://www.voalte.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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > 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>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Apr 28 07:01:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 09:01:53 -0500
Subject: [Freeswitch-users] session.steamFile misses DTMF event forfirst
	few seconds
In-Reply-To: <5CE98178D9124659BC930EDDBF0ED450@ws4>
References: <u2r191c3a031004270840o6948472blfa90e9ca0a238914@mail.gmail.com>
	<5CE98178D9124659BC930EDDBF0ED450@ws4>
Message-ID: <v2w191c3a031004280701wcc3d95a6o18c0bbb08b9ae28a@mail.gmail.com>

I told you what to do if you want to have the answer.
do I need to repost it or did you get the last email?


On Wed, Apr 28, 2010 at 5:08 AM, Frank @ Impact <frank at impactfax.com> wrote:

>  Yes. I understand your position of trying to help people using old code.
>  That is why I tried to download the latest code to try to stay current.
>
>
>
> But that is also when I found this problem with the newer code.
>
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Tuesday, April 27, 2010 11:40 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] session.steamFile misses DTMF event
> forfirst few seconds
>
>
>
> 12XXX is so many years old, i wish users who want free help would at least
> stay up to date with the code.
>
> FYI, our repo is on git now and the svn mirror is not updating at the
> moment.
>
> produce a complete minimal script that reproduces your problem and can be
> run on git HEAD (see download instructions to learn how to build with git)
> Use existing sound files from the FS install so we can just run it in our
> lab to reproduce the issue.
>
> Open an issue on http://jira.freeswitch.org and attach the script.
>
>
>  On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank at impactfax.com>
> wrote:
>
> The calls are coming from land based lines.  Traditional POTS (not cable
> company).  Caller is calling into FS and providing those DTMF.  I can
> reproduce on my POTS line (3000' from CO) and there is no discernable
> noise on the line.
>
> The call comes into the media gateway and then goes sip to FS.
>
> I can reproduce the problem just by starting FS version from the latest
> trunk.  And then I can eliminate the problem by restarting the FS
> version 12790.  In both test cases, the media gateway remains constant.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Alberto Escudero
> Sent: Tuesday, April 27, 2010 9:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
> first few seconds
>
> How are you  receiving those DTMFs, inbound or outbound? Are those calls
> coming from a mobile network (GSM). We have experienced lots of problem
> with DTMF detection in noisy lines.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > I recently upgraded from FS 12790M to svn 17188.  When I did, I
> noticed
> > that session.streamFile behaved differently and I started having
> > problems with my IVR app.
> >
> > With the upgraded FS, I have a problem with streamFile no firing on
> the
> > DTMF and calling the callback function for the first few seconds of
> the
> > wav file playback.  It behaves as though it does not hear the DTMFs.
> If
> > I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> > catches the DTMF and all is well.  If I key as soon as I hear the wav
> > file start, streamFile just keeps playing the wav and does not call
> the
> > callback function.
> >
> > When I revert back to the previous version of FS, streamFile always
> > fires the callback right away no matter how quickly I press the first
> > DTMF as the wav file starts to stream out.
> >
> > The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> > 16 bit, mono 8000 Hz
> >
> > The snippet of js code I am using is as follows.
> >
> > if(session.ready()) {
> >     session.answer();
> >     session.sleep(750);
> >     while(session.ready()) {
> >         session.sleep(500);
> >         session.flushDigits(); // clear out input buffers
> >
> >
> >
> if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> > ts_cb,""))===false) {
> >             pin=session.getDigits(pinmax,pinterm,pinwait);
> >         } else {
> >             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
> >         }
> >         // more code here..
> >     }
> >
> > Do I need to change the way I use streamFile in the later release?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Apr 28 07:06:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 09:06:07 -0500
Subject: [Freeswitch-users] Difference between effective-caller-id and
	outbound-caller-id
In-Reply-To: <o2m7abab2411004280515wad8b1732t185e7b93ac2e37cd@mail.gmail.com>
References: <o2m7abab2411004280515wad8b1732t185e7b93ac2e37cd@mail.gmail.com>
Message-ID: <u2u191c3a031004280706p1f6f1a7cp81ece637ece6a9d@mail.gmail.com>

neither one mean anything?

do you mean effective_caller_id_name effective_caller_id_number vs
origination_caller_id_name origination_caller_id_number

if so

effective_caller_id_name/number are variables you set on an inbound channel
so when that inbound channel calls the bridge app to connection to an
outbound channel, it will pass the values in that variable instead of the
actual caller_id fields that were passed by the caller.

origination_caller_id_name/number are variables you set in a dial-string to
control the caller-id
{origination_caller_id_name=test,origination_caller_id_number=51212}sofia/internal/
foo at bar.com

This is independent of the inbound leg and is also usable when there is no
inbound leg such as with the originate cli command.



On Wed, Apr 28, 2010 at 7:15 AM, Frank Church <vfclists at googlemail.com>wrote:

> What are the differences between the two and when is one more
> appropriate than the other?
>
> What are the use cases?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 28 07:43:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 09:43:07 -0500
Subject: [Freeswitch-users] Error with media ptime
In-Reply-To: <p2haa768f581004272119qe1b44579yb07c2d0d63e895ed@mail.gmail.com>
References: <s2naa768f581004272027m15a62473y9defe9764f48ac18@mail.gmail.com>
	<p2haa768f581004272119qe1b44579yb07c2d0d63e895ed@mail.gmail.com>
Message-ID: <v2q191c3a031004280743vf64121f6u4351356a1ac5659e@mail.gmail.com>

yes it will.
whatever you are talking to is broken. It's up to them to fix the broken
timestamps and ptime advertisement.


On Tue, Apr 27, 2010 at 11:19 PM, leonardo alves <telteclistas at gmail.com>wrote:

> Sorry I found this message:
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg10032.html
> and the
> <param name="inbound-codec-negotiation" value="scrooge"/>
> fixed my problem.
>
> Now I have another question.
> What exactly this "scrooge" does ? Is this going to affect the other
> providers that was working ?
> Thanks
> Leonardo
>
>
>
> On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <telteclistas at gmail.com>wrote:
>
>> Hello,
>>
>> I am new to freeswitch and I have just installed the last version of
>> freeswitch. I am doing some tests dialing with sip and when the call is
>> answer I play an file.
>> If I do 10 calls, in 5 or 6 of them I get this error in the console and
>> the audio gets cut like with it was a bandwithd problem.
>>
>> 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use
>> ptime 20 but what they meant to say was 30
>> This issue has so far been identified to happen on the following broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so broken
>> who knows what will happen.
>>
>> Does anyone knows if there is a way to fix this issue ?
>>
>> Thanks
>> Leonardo
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From Prometheus001 at gmx.net  Wed Apr 28 09:03:43 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 28 Apr 2010 18:03:43 +0200
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
Message-ID: <4BD85C5F.3000302@gmx.net>

Hello,

I tried mod_com_g720 with 2 licenses and ran into a problem:

What DOES work with G729:

    * Calling Mailbox (Phone is Snom 360)
    * Calling external Numbers through Patton gatway

What does NOT work

    * Calling another Snom Phone with G.729 enabled (both Phones use
      TLS/SRTP)

Here an exempt from the log calling phone is 200, called phone is Snom 320:
2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare
[G729:18:8000:20]/[G729:18:8000:20]
2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729
2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send
payload to 101
2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP
2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are
unchanged for sofia/internal/sip:208 at 192.168.178.126:5060.
2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE -
0x9a09510 0x8e774c8
2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE
FAILED - 0x8e9f9a8 (nil)
2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729
decoder error!
2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478
sofia/internal/200 at fs00.telefaks.biz ending bridge by request from read
function

g729_status
Permitted G.729AB channels: 2
Encoders in use: 0
Decoders in use: 0

Here is the dialplan
    <extension name="Local_Extension_no_Voicemail">.
      <condition field="destination_number" expression="^\S*$"
break="never">.
        <action application="set" data="transfer_ringback=${us-ring}"/>.
        <action application="set" data="dialed_ext=208"/>.
        <action application="set" data="effective_caller_id_number=200"/>.
        <action application="set" data="effective_caller_id_name=Peter "/>.
        <action application="db"
data="insert/call_return/${dialed_ext}/${caller_id_number}"/>.
        <action application="db"
data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>.
        <action application="set" data="dialed_ext=208"/>.
        <action application="export" data="dialed_ext=208"/>.
        <action application="ring_ready" />.
        <action application="bridge" data="user/208 at my.domain"/>
      </condition>.
    </extension>   

Just tested it with an Aastra Phone without TLS: Same Behaviour.

Anybody has a clue how to solve this?

Best regards
Peter



From Prometheus001 at gmx.net  Wed Apr 28 09:05:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 28 Apr 2010 18:05:16 +0200
Subject: [Freeswitch-users] effective_caller_id_name not working while
 calling multiple endpoints with :_:
In-Reply-To: <4BD16ABE.2030506@gmx.net>
References: <4BCED007.80900@gmx.net>	<66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>	<l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>
	<4BD16ABE.2030506@gmx.net>
Message-ID: <4BD85CBC.4080200@gmx.net>

Nobody has an idea how to solve this? Shall I open a JIRA?

Best regards
Peter

Peter P GMX schrieb:
> Hello Anthony,
>
> I upgraded to newest GIT and tried it
>
> The dialplan now contains the fowllowing:
> <action application="bridge" data="<effective_caller_id_name=My
> Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain"/>
>
> When the dialplan is executed, it seems to be processed correctly:
> EXECUTE sofia/local/06912345678 at 192.168.178.218:5060
> bridge(<effective_caller_id_name=My
> Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain)
> 2010-04-23 10:59:12.479598 [DEBUG] switch_ivr_originate.c:1394 variable
> string 0 = [effective_caller_id_name=My Name]
> 2010-04-23 10:59:12.501255 [DEBUG] switch_ivr_originate.c:1885 variable
> string 0 = [global_to_originate_1=true]
> 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> string 0 = [presence_id=200 at my.domain]
> 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> string 1 = [transfer_fallback_extension=200]
>
> However the INVITE message does not contain the caller_id_name, see below
>
>
> What am I doing wrong?
>
> Best regards
> Peter
>
> U 192.168.178.220:5060 -> 192.168.178.50:3072
> INVITE sip:200 at 192.168.178.50:3072;line=v3bii5l2 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bK4jUFH215p85tr.
> Max-Forwards: 70.
> From: "06912345678" <sip:06912345678 at 192.168.178.220>;tag=a8m8ccQcgjjUg.
> To: <sip:200 at 192.168.178.50:3072;line=v3bii5l2>.
> Call-ID: b6cafa23-c959-122d-4682-080027e51f59.
> CSeq: 129888323 INVITE.
> Contact: <sip:mod_sofia at 192.168.178.220:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 348.
> X-FS-Support: update_display.
> Remote-Party-ID: "06912345678"
> <sip:06912345678 at 192.168.178.220>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1272001222 1272001223 IN IP4 192.168.178.220.
> s=FreeSWITCH.
> c=IN IP4 192.168.178.220.
> t=0 0.
> m=audio 12096 RTP/AVP 9 0 8 99 3 101 13.
> a=rtpmap:9 G722/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:99 SPEEX/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.
>
> Anthony Minessale schrieb:
>   
>> when using enterprise_originate you must use the special leading <>
>> brackets to insert global variables meant for each tier 1 originate
>>
>> <global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/foo at bar.com
>> <mailto:foo at bar.com>,sofia/internal/foo2 at bar.com:_:sofia/internal/foo3 at bar3.com
>> <mailto:foo3 at bar3.com>
>>
>>
>> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone
>> <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>> wrote:
>>
>>     I think you should first thing update to latest GIT :)
>>
>>     David Ponzone  Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service Client IPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  -   www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout message
>>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline
>>     toute responsabilit? au titre de ce message s'il a ?t? alt?r?,
>>     d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>     message, merci de le d?truire imm?diatement et d'avertir
>>     l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :
>>
>>     
>>>     Setting the effective_caller_id_name when dialing multiple endpoints
>>>     with :_: do not seem to work.
>>>     See example:
>>>      <action application="set" data="effective_caller_id_name=MyName"/>
>>>      <action application="bridge"
>>>
>>>     data="user/30 at mydomain.com
>>>     <mailto:user/30 at mydomain.com>:_:user/31 at mydomain.com
>>>     <mailto:user/31 at mydomain.com>:_:user/32 at mydomain.com
>>>     <mailto:user/32 at mydomain.com>:_:user/33 at mydomain.com
>>>     <mailto:user/33 at mydomain.com>:_:user/34 at mydomain.com
>>>     <mailto:user/34 at mydomain.com>"/>
>>>
>>>     Freeswitch tries to set it:
>>>       EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414
>>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060
>>>     set(effective_caller_id_name=MyName)
>>>       2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
>>>     sofia/external/069xxxxxxxx at 10.xx.xx.1414
>>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060 SET
>>>     [effective_caller_id_name]=[MyName]
>>>
>>>     But the SIP messages do not contain the effective_caller_id_name.
>>>
>>>     If we change the ":_:" sperator to "," then the
>>>     effective_caller_id_name
>>>     is correctly submittted (hower I cannot call
>>>     multiple-registrations on
>>>     one number then).
>>>
>>>     We are on
>>>      FreeSWITCH Version 1.0.head (svn-17188)
>>>
>>>     Any ideas how to overcome this? Or shall I open a JIRA?
>>>
>>>     Best regards
>>>     Peter
>>>
>>>     See example SIP message:
>>>
>>>     U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
>>>     INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
>>>     Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
>>>     Max-Forwards: 70.
>>>     From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
>>>     To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
>>>     Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
>>>     CSeq: 129803060 INVITE.
>>>     Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
>>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
>>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>     REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
>>>     Supported: timer, precondition, path, replaces.
>>>     Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>     include-session-description, presence.winfo, message-summary, refer.
>>>     Content-Type: application/sdp.
>>>     Content-Disposition: session.
>>>     Content-Length: 920.
>>>     X-FS-Support: update_display.
>>>     Remote-Party-ID: "069xxxxxxxx"
>>>     <sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
>>>     .
>>>     v=0.
>>>     o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
>>>     s=FreeSWITCH.
>>>     c=IN IP4 10.xx.xx.141.
>>>     t=0 0.
>>>     m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9
>>>     121 3
>>>     101 13.
>>>     a=rtpmap:115 G7221/32000.
>>>     a=fmtp:115 bitrate=48000.
>>>     a=rtpmap:96 AMR/8000.
>>>     a=fmtp:96 octet-align=0.
>>>     a=rtpmap:99 SPEEX/32000.
>>>     a=rtpmap:18 G729/8000.
>>>     a=rtpmap:4 G723/8000.
>>>     a=rtpmap:7 LPC/8000.
>>>     a=rtpmap:124 G726-16/8000.
>>>     a=rtpmap:8 PCMA/8000.
>>>     a=rtpmap:6 DVI4/16000.
>>>     a=rtpmap:123 G726-24/8000.
>>>     a=rtpmap:0 PCMU/8000.
>>>     a=rtpmap:10 L16/22050.
>>>     a=rtpmap:98 iLBC/8000.
>>>     a=fmtp:98 mode
>>>     #
>>>
>>>
>>>
>>>     _______________________________________________
>>>     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/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 28 09:09:24 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Apr 2010 11:09:24 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BD85C5F.3000302@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
Message-ID: <0A28B620-31CF-4935-AD96-9F6763BF0865@freeswitch.org>

what settings do you have in the snom?  I have tested it with my polycom with SRTP so I doubt that is the issue.

/b

On Apr 28, 2010, at 11:03 AM, Peter P GMX wrote:

> 
> What does NOT work
> 
>    * Calling another Snom Phone with G.729 enabled (both Phones use
>      TLS/SRTP)



From mardy at voysys.com  Wed Apr 28 09:30:50 2010
From: mardy at voysys.com (Mardy Marshall)
Date: Wed, 28 Apr 2010 12:30:50 -0400
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<0C713278-DE8E-4339-9844-E3A01FF40E6B@voysys.com>
	<i2p191c3a031004191653oc2bf5752zf8e6df96bae4d7a3@mail.gmail.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
	<i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
	<D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>
	<l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>
Message-ID: <79035715-B003-4BDA-AD3C-B187718E8ADB@voysys.com>

I tried a new build created from git-head and the situation seems to have gotten worse.  In my actual application, which is using the mod_event_socket API to communicate with FreeSwitch, I no longer receive any events indicating that a transfer of the call has occurred.  Also both call legs remain active as FreeSWITCH never terminates the original call leg created for the transfer consultation.

Prior to these latest changes, this is what I would observe:

The station initiating the transfer would dial the destination extension, for example, extension 300, which is then redirected to FreeSWITCH.  FreeSWITCH is configured with the following dialplan entry:

    <extension name="park">
      <condition field="destination_number" expression="^300$">
        <action application="socket" data="192.168.0.16:8086 async full"/>
      </condition>
    </extension>

This initial "consultation" call will then launch my application which will communicate with FreeSWITCH via an outbound event socket connection.  Later,when the station being transferred sends the INVITE w/Replaces to FreeSWITCH, FS would then create a new UUID and bridge that to the original UUID associated with the initiating call leg.  In response to this, FS sends the application a uuid_bridge event which contains the other UUID.  From the application side, I am able to respond to this uuid_bridge event and redirect the application to the new call leg.  The problem is that the original call leg does not get terminated by FS and I haven't been able to terminate it via the API without ending up shutting down the whole session.  To make matters worse, because FS did not terminate the original call leg, the station that initiated the transfer thinks that the transfer failed and sends a BYE to FS which then kills the session and drops the call.  Yuck...

-Mardy



On Apr 27, 2010, at 3:43 PM, Anthony Minessale wrote:

> The problem here is that you were testing it against a 1 legged call (echo application is not bridged to anything)
> 
> This code is usually only encountered when replacing a leg of a call.
> 
> try git rev [8660b6f] or better
> 
> I added a patch so the new channel executes the same app that the old was originally executing and hangs up on the original.
> 
> Note, if you want the transferred call to pick up on the other call to the application in-progress you would have to loop the original call.
> 
> 
> 
> 
> On Tue, Apr 27, 2010 at 11:03 AM, Mardy Marshall <mardy at voysys.com> wrote:
> The REFER exchange occurs on the other side of the proxy between the two Polycom phones.  Since
> FreeSWITCH is the destination of the transfer, it will only see the INVITE w/Replaces.  I've sent you the
> PCAP file which captures the complete SIP exchange.  That should help to clarify.
> 
> -Mardy
> 
> On Apr 27, 2010, at 11:47 AM, Anthony Minessale wrote:
> 
>> There is no sign of a REFER packet or anything indicating an attended transfer in this trace.
>> Did you send the wrong one possibly?
>> 
>> 
>> On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:
>> Here is the setup that was used to reproduce the consultative transfer problem.
>> 
>> There are two boxes, the first is running a proxy based PBX (sipXecs) and the
>> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones, extension
>> 200 and 202, registered with it.  The PBX has configured a mapping rule which
>> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060 which
>> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
>> configured to allow connections from the PBX box via an ACL configuration and
>> the public dialplan includes an "echo" extension:
>> 
>>     <extension name="echo">
>>       <condition field="destination_number" expression="^9996$">
>>         <action application="answer"/>
>>         <action application="echo"/>
>>       </condition>
>>     </extension>
>> 
>> Any of the phones registered with the PBX can dial extension 9996 and be
>> connected to the FreeSWITCH echo application.  But when one phone attempts
>> to transfer another phone to extension 9996 via a consultative transfer,
>> FreeSWITCH does not properly complete the transfer.  You can see in the log
>> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
>> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
>> initiated the transfer.  Without that terminating BYE, the transfer
>> controller thinks that the transfer failed.
>> 
>> The corresponding FreeSWITCH log file - http://pastebin.freeswitch.org/12806
>> 
>> If it will help, I can also forward a corresponding PCAP file.
>> 
>> Thanks
>> 
>> -Mardy
>> 
>> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>> 
>>> I'm trying to understand this:
>>> 
>>> If FS is acting as a phone in your scenario why are you sending a refer to it and not the server?
>>> In most situations there is a b2bua server who routes the calls and takes all the REFER.
>>> Is this one of those PROXY only sip servers?
>>> 
>>> I think you would need to produce a full debug log of this, and if you are using some kind of proxy based setup we would need some way to easily reproduce it or visit your lab because we do not typically use anything of the sort.
>>> 
>>> Execute these commands and reproduce it and capture the whole log and put it on
>>> http://pastebin.freeswitch.org
>>> 
>>> sofia profile internal siptrace on
>>> console loglevel debug
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> instead of emailing again when impatient for an answer (something we frown upon here in this busy list) 
>>> produce a reproducible step by step process to duplicate your issue.  We are trying to help people but we don't have the time to do the leg work for everyone who asks a question when we get hundreds of emails a day.
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>> Just following up...  Does anyone have any suggestions on how to proceed with this?  I've run out of ideas.
>>> 
>>> Thanks,
>>> 
>>> -Mardy
>>> 
>>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>> 
>>>> The phones that I am using are not registered with FS.  They are registered with another proxy based PBX.  I am simply using FS as B2BUA which is also registered with the PBX.  And yes, I can successfully transfer a call to another phone with this setup.
>>>> 
>>>> To simplify things I tried the same scenario using FSComm in place of my own FS application and tried to transfer a call to FSComm with the same results.  And just in case there might be a problem specific to FSComm, I set up a clean install of FS 1.0.6 and tried transferring a call to the FS echo application with the same results.  By the way, I have no problems with blind transfers, only attended transfers.
>>>> 
>>>> -Mardy
>>>> 
>>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>>> 
>>>>> did you try just setting up 2 phones on plain fresh FS install, and calling them normally and transferring them around?
>>>>> That description is still pretty vague? What is an Event Socket application, which has nothing to do with sip and sip transfers, that's a FS protocol.
>>>>> 
>>>>> 
>>>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>> I have two phones (Polycom) and an event_socket application, all of which are using a SIP proxy for call routing.  The first phone calls the second phone.  The second phone then attempts to transfer the call to the FS/event_socket application by first placing the call on hold and then calling the FS application, followed by a consultative transfer.  The REFER dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.  The transferred call leg appears to be answered by FS and the application receives a uuid_bridge event with the UUID of the new call leg.  The problem that I see is that the original call leg, created when the user called the FS application to announce the transfer, does not get canceled by FS and subsequently does not send the BYE back to the Polycom.  Is there something that I need to do at the event_socket application to complete the transfer?  I've tried killing the UUID associated with the first call leg as well as issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>>> 
>>>>> -Mardy
>>>>> 
>>>>> 
>>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>>> 
>>>>>> but what is the client sending the REFER?
>>>>>> 
>>>>>> FS gets refer+replaces all the time, if it's the one where the dest is on another box (aka the nightmare xfer that you should see references to in the debug log if so) then it will not complete until that far end call is answered.
>>>>>> 
>>>>>> FS handles this scenerio for us hundreds of times a day using a wide range of sip devices so perhaps
>>>>>> your UA has an interop problem.
>>>>>> 
>>>>>> 
>>>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>>> 
>>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>>> 
>>>>>>> uuid_simplify will issue the refer...
>>>>>> 
>>>>>> I looked at uuid_simplify and if I understand it correctly it is for use when one wants to act as the transfer controller.  In my case, FS is the transfer destination.  Another phone has already generated the refer and FS has been sent an invite with replaces.
>>>>>> 
>>>>>>> 
>>>>>>> May I ask what application you are developing?
>>>>>> 
>>>>>> An ACD.
>>>>>> 
>>>>>>> 
>>>>>>> Regards,
>>>>>>> Jo?o Mesquita
>>>>>>> FSComm developer
>>>>>>> 
>>>>>>> 
>>>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com> wrote:
>>>>>>> I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm.  (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.)  What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller.  From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active.  I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call.  Is there a specific action that the application must take in order for the transfer to complete?
>>>>>>> 
>>>>>>> -Mardy
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> -- 
>>>>>> Anthony Minessale II
>>>>>> 
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>> 
>>>>>> 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
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> Anthony Minessale II
>>>>> 
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> 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
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Apr 28 09:49:20 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 28 Apr 2010 18:49:20 +0200
Subject: [Freeswitch-users] Good book on (VoIP) telephony?
In-Reply-To: <y2m87f2f3b91004272222tecf30eep37daa7ff0bdb8cbf@mail.gmail.com>
References: <1272412604620-4971798.post@n2.nabble.com>
	<x2u9dc4a1671004271748lafb75ea8s1c8cb69708a9e1ed@mail.gmail.com>
	<y2m87f2f3b91004272222tecf30eep37daa7ff0bdb8cbf@mail.gmail.com>
Message-ID: <s2p7b197bef1004280949pe5f1a0cxb0390effff76c8bd@mail.gmail.com>

On Wed, Apr 28, 2010 at 7:22 AM, Michael Collins <msc at freeswitch.org> wrote:
> I normally don't like "for dummies" books but honestly I have to recommend
> this one:
> http://www.amazon.com/VoIP-Deployment-Dummies-Stephen-Olejniczak/dp/047038543X/ref=pd_sim_b_2

I definitely confirm is a very useful one: gives you an holistic
approach to all you need to know about. After that one, you can go
deep in what interest you the most with the other books. But you'll
gain in having the entire picture in your mind.

-giovanni

>
> In spite of the annoying typos ("Secession Initiation Protocol", "Secession
> Description Protocol", "Session Boarder Controller") I found it to be very
> handy for someone learning about SIP and where it fits into the LAN, WAN,
> etc. plus some handy troubleshooting tips and a nice section on using
> Wireshark to debug VoIP calls.
>
> -MC
>
> On Tue, Apr 27, 2010 at 5:48 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>
>> If you are a new at VoIP look into these publications:
>>
>> http://www.amazon.com/SIP-Understanding-Initiation-Protocol-Telecommunications/dp/1607839954/ref=sr_1_1?ie=UTF8&s=books&qid=1272415143&sr=1-1
>>
>>
>> http://www.amazon.com/SIP-Beyond-VoIP-Communications-Revolution/dp/0974813001/ref=sr_1_10?ie=UTF8&s=books&qid=1272415143&sr=1-10
>>
>>
>> http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741/ref=sr_1_12/189-7099384-6630443?ie=UTF8&s=books&qid=1272415088&sr=1-12
>>
>>
>> http://www.amazon.com/Understanding-Servlets-Artech-House-Telecommunications/dp/159693428X/ref=sr_1_9?ie=UTF8&s=books&qid=1272415143&sr=1-9
>>
>>
>> http://www.amazon.com/Switching-VoIP-Theodore-Wallingford/dp/0596008686/ref=sr_1_1?ie=UTF8&s=books&qid=1272415530&sr=1-1
>>
>> Or look into these Google eBooks links
>>
>> http://books.google.com/books?id=9_wRFy5OGw4C&dq=asterisk+book&printsec=frontcover&source=bn&hl=en&ei=mYXXS6DeG4bc8ASul_SSBw&sa=X&oi=book_result&ct=result&resnum=4&ved=0CCsQ6AEwAw#v=onepage&q=asterisk%20book&f=false
>>
>> -E
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Giovanni Maruzzelli
Cell : +39-347-2665618


From sean at obscuradigital.com  Wed Apr 28 09:54:07 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Wed, 28 Apr 2010 09:54:07 -0700
Subject: [Freeswitch-users] Functionality
Message-ID: <C7FDB63F.902F%sean@obscuradigital.com>

Hello list,

I?ve been looking through the wiki to figure the best way to enable the
receptionist to turn on or off an ivr after hours auto-attendant.  I?m
thinking the receptionist can enter a couple digits to control this
functionality.  Can someone provide an example or show me in the wiki a good
place to figure this out

Thanks
Sean
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From msc at freeswitch.org  Wed Apr 28 10:00:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 10:00:24 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <t2v87f2f3b91004281000n2ee96a1ep51dd9e86ff12f5b0@mail.gmail.com>

Come join us!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28

-MC
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From anthony.minessale at gmail.com  Wed Apr 28 11:59:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 13:59:57 -0500
Subject: [Freeswitch-users] Having a problem with attended transfer when
	FS is the transfer target
In-Reply-To: <79035715-B003-4BDA-AD3C-B187718E8ADB@voysys.com>
References: <B1BA0198-133A-4AD6-85A7-78D0D8A6E48A@voysys.com>
	<1A509938-2FEE-49B8-AFBD-AA2E9CA8A0F6@voysys.com>
	<FAB4AA7C-06AB-49D2-B2A6-843DF516D1C0@voysys.com>
	<j2o191c3a031004210713rc47ce3e0v2b9a653c55173f9c@mail.gmail.com>
	<t2u191c3a031004210823q4a2f883dy6e68a35aa94d6937@mail.gmail.com>
	<30C47F7A-908B-421F-B09D-4476FD4B9291@voysys.com>
	<i2i191c3a031004270847o2d3cf9a6ib6eb91e83c993fda@mail.gmail.com>
	<D6424C9B-BE92-4EFD-9114-FF90F768CBE0@voysys.com>
	<l2i191c3a031004271243y4843530dh810d415a1d94ee89@mail.gmail.com>
	<79035715-B003-4BDA-AD3C-B187718E8ADB@voysys.com>
Message-ID: <x2r191c3a031004281159r11349edub5f8d011df936cda@mail.gmail.com>

I am not sure what you are talking about but the small patch I did was only
to the code that receives an INVITE+replaces and nothing else.

Also, I thought you said you took your custom application out of the mix and
was just doing a basic test with only sipX and FreeSWITCH.

you may want to re-think your approach.....


On Wed, Apr 28, 2010 at 11:30 AM, Mardy Marshall <mardy at voysys.com> wrote:

> I tried a new build created from git-head and the situation seems to have
> gotten worse.  In my actual application, which is using the mod_event_socket
> API to communicate with FreeSwitch, I no longer receive any events
> indicating that a transfer of the call has occurred.  Also both call legs
> remain active as FreeSWITCH never terminates the original call leg created
> for the transfer consultation.
>
> Prior to these latest changes, this is what I would observe:
>
> The station initiating the transfer would dial the destination extension,
> for example, extension 300, which is then redirected to FreeSWITCH.
>  FreeSWITCH is configured with the following dialplan entry:
>
>     <extension name="park">
>       <condition field="destination_number" expression="^300$">
>         <action application="socket" data="192.168.0.16:8086 async full"/>
>       </condition>
>     </extension>
>
> This initial "consultation" call will then launch my application which will
> communicate with FreeSWITCH via an outbound event socket connection.
>  Later,when the station being transferred sends the INVITE w/Replaces to
> FreeSWITCH, FS would then create a new UUID and bridge that to the original
> UUID associated with the initiating call leg.  In response to this, FS sends
> the application a uuid_bridge event which contains the other UUID.  From the
> application side, I am able to respond to this uuid_bridge event and
> redirect the application to the new call leg.  The problem is that the
> original call leg does not get terminated by FS and I haven't been able to
> terminate it via the API without ending up shutting down the whole session.
>  To make matters worse, because FS did not terminate the original call leg,
> the station that initiated the transfer thinks that the transfer failed and
> sends a BYE to FS which then kills the session and drops the call.  Yuck...
>
> -Mardy
>
>
>
> On Apr 27, 2010, at 3:43 PM, Anthony Minessale wrote:
>
> The problem here is that you were testing it against a 1 legged call (echo
> application is not bridged to anything)
>
> This code is usually only encountered when replacing a leg of a call.
>
> try git rev [8660b6f] or better
>
> I added a patch so the new channel executes the same app that the old was
> originally executing and hangs up on the original.
>
> Note, if you want the transferred call to pick up on the other call to the
> application in-progress you would have to loop the original call.
>
>
>
>
> On Tue, Apr 27, 2010 at 11:03 AM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> The REFER exchange occurs on the other side of the proxy between the two
>> Polycom phones.  Since
>> FreeSWITCH is the destination of the transfer, it will only see the INVITE
>> w/Replaces.  I've sent you the
>> PCAP file which captures the complete SIP exchange.  That should help to
>> clarify.
>>
>> -Mardy
>>
>> On Apr 27, 2010, at 11:47 AM, Anthony Minessale wrote:
>>
>> There is no sign of a REFER packet or anything indicating an attended
>> transfer in this trace.
>> Did you send the wrong one possibly?
>>
>>
>> On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>
>>> Here is the setup that was used to reproduce the consultative transfer
>>> problem.
>>>
>>> There are two boxes, the first is running a proxy based PBX (sipXecs) and
>>> the
>>> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones,
>>> extension
>>> 200 and 202, registered with it.  The PBX has configured a mapping rule
>>> which
>>> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060which
>>> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
>>> configured to allow connections from the PBX box via an ACL configuration
>>> and
>>> the public dialplan includes an "echo" extension:
>>>
>>>     <extension name="echo">
>>>       <condition field="destination_number" expression="^9996$">
>>>         <action application="answer"/>
>>>         <action application="echo"/>
>>>       </condition>
>>>     </extension>
>>>
>>> Any of the phones registered with the PBX can dial extension 9996 and be
>>> connected to the FreeSWITCH echo application.  But when one phone
>>> attempts
>>> to transfer another phone to extension 9996 via a consultative transfer,
>>> FreeSWITCH does not properly complete the transfer.  You can see in the
>>> log
>>> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
>>> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
>>> initiated the transfer.  Without that terminating BYE, the transfer
>>> controller thinks that the transfer failed.
>>>
>>> The corresponding FreeSWITCH log file -
>>> http://pastebin.freeswitch.org/12806
>>>
>>> If it will help, I can also forward a corresponding PCAP file.
>>>
>>> Thanks
>>>
>>> -Mardy
>>>
>>> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>>>
>>> I'm trying to understand this:
>>>
>>> If FS is acting as a phone in your scenario why are you sending a refer
>>> to it and not the server?
>>> In most situations there is a b2bua server who routes the calls and takes
>>> all the REFER.
>>> Is this one of those PROXY only sip servers?
>>>
>>> I think you would need to produce a full debug log of this, and if you
>>> are using some kind of proxy based setup we would need some way to easily
>>> reproduce it or visit your lab because we do not typically use anything of
>>> the sort.
>>>
>>> Execute these commands and reproduce it and capture the whole log and put
>>> it on
>>> http://pastebin.freeswitch.org
>>>
>>> sofia profile internal siptrace on
>>> console loglevel debug
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> instead of emailing again when impatient for an answer (something we
>>>> frown upon here in this busy list)
>>>> produce a reproducible step by step process to duplicate your issue.  We
>>>> are trying to help people but we don't have the time to do the leg work for
>>>> everyone who asks a question when we get hundreds of emails a day.
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>
>>>>> Just following up...  Does anyone have any suggestions on how to
>>>>> proceed with this?  I've run out of ideas.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> -Mardy
>>>>>
>>>>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>>>>
>>>>> The phones that I am using are not registered with FS.  They are
>>>>> registered with another proxy based PBX.  I am simply using FS as B2BUA
>>>>> which is also registered with the PBX.  And yes, I can successfully transfer
>>>>> a call to another phone with this setup.
>>>>>
>>>>> To simplify things I tried the same scenario using FSComm in place of
>>>>> my own FS application and tried to transfer a call to FSComm with the same
>>>>> results.  And just in case there might be a problem specific to FSComm, I
>>>>> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
>>>>> echo application with the same results.  By the way, I have no problems with
>>>>> blind transfers, only attended transfers.
>>>>>
>>>>> -Mardy
>>>>>
>>>>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>>>>
>>>>> did you try just setting up 2 phones on plain fresh FS install, and
>>>>> calling them normally and transferring them around?
>>>>> That description is still pretty vague? What is an Event Socket
>>>>> application, which has nothing to do with sip and sip transfers, that's a FS
>>>>> protocol.
>>>>>
>>>>>
>>>>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>
>>>>>> I have two phones (Polycom) and an event_socket application, all of
>>>>>> which are using a SIP proxy for call routing.  The first phone calls the
>>>>>> second phone.  The second phone then attempts to transfer the call to the
>>>>>> FS/event_socket application by first placing the call on hold and then
>>>>>> calling the FS application, followed by a consultative transfer.  The REFER
>>>>>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>>>>>  The transferred call leg appears to be answered by FS and the application
>>>>>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>>>>>> that I see is that the original call leg, created when the user called the
>>>>>> FS application to announce the transfer, does not get canceled by FS and
>>>>>> subsequently does not send the BYE back to the Polycom.  Is there something
>>>>>> that I need to do at the event_socket application to complete the transfer?
>>>>>>  I've tried killing the UUID associated with the first call leg as well as
>>>>>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>>>>
>>>>>> -Mardy
>>>>>>
>>>>>>
>>>>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>>>>
>>>>>> but what is the client sending the REFER?
>>>>>>
>>>>>> FS gets refer+replaces all the time, if it's the one where the dest is
>>>>>> on another box (aka the nightmare xfer that you should see references to in
>>>>>> the debug log if so) then it will not complete until that far end call is
>>>>>> answered.
>>>>>>
>>>>>> FS handles this scenerio for us hundreds of times a day using a wide
>>>>>> range of sip devices so perhaps
>>>>>> your UA has an interop problem.
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>>
>>>>>>>
>>>>>>> On Apr 19, 2010, at 5:12 PM, Jo?o Mesquita wrote:
>>>>>>>
>>>>>>> uuid_simplify will issue the refer...
>>>>>>>
>>>>>>>
>>>>>>> I looked at uuid_simplify and if I understand it correctly it is for
>>>>>>> use when one wants to act as the transfer controller.  In my case, FS is the
>>>>>>> transfer destination.  Another phone has already generated the refer and FS
>>>>>>> has been sent an invite with replaces.
>>>>>>>
>>>>>>>
>>>>>>> May I ask what application you are developing?
>>>>>>>
>>>>>>>
>>>>>>> An ACD.
>>>>>>>
>>>>>>>
>>>>>>> Regards,
>>>>>>> Jo?o Mesquita
>>>>>>> FSComm developer
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>>>>
>>>>>>>> I'm having a problem with attended transfers where the destination
>>>>>>>> of the transfer is a FreeSWITCH based application such as FSComm.  (It
>>>>>>>> should be noted that in my setup the phone performing the transfer and the
>>>>>>>> caller which is being transferred are parties of another SIP server.)  What
>>>>>>>> I see, from a SIP signaling standpoint, is that after FreeSWITCH receives
>>>>>>>> and acknowledges the INVITE w/Replaces it does not terminate the initial
>>>>>>>> call leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>>>>>> application side, FS still thinks that both the initial call leg and
>>>>>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>>>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>>>>>> kill all legs of the call.  Is there a specific action that the application
>>>>>>>> must take in order for the transfer to complete?
>>>>>>>>
>>>>>>>> -Mardy
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> 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>
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> 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>
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 28 13:24:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 15:24:20 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BD85C5F.3000302@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
Message-ID: <y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>

do you have lastest git HEAD ?
can you update and try again?


On Wed, Apr 28, 2010 at 11:03 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I tried mod_com_g720 with 2 licenses and ran into a problem:
>
> What DOES work with G729:
>
>    * Calling Mailbox (Phone is Snom 360)
>    * Calling external Numbers through Patton gatway
>
> What does NOT work
>
>    * Calling another Snom Phone with G.729 enabled (both Phones use
>      TLS/SRTP)
>
> Here an exempt from the log calling phone is 200, called phone is Snom 320:
> 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare
> [G729:18:8000:20]/[G729:18:8000:20]
> 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729
> 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send
> payload to 101
> 2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP
> 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are
> unchanged for sofia/internal/sip:208 at 192.168.178.126:5060.
> 2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE -
> 0x9a09510 0x8e774c8
> 2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE
> FAILED - 0x8e9f9a8 (nil)
> 2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729
> decoder error!
> 2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478
> sofia/internal/200 at fs00.telefaks.biz ending bridge by request from read
> function
>
> g729_status
> Permitted G.729AB channels: 2
> Encoders in use: 0
> Decoders in use: 0
>
> Here is the dialplan
>    <extension name="Local_Extension_no_Voicemail">.
>      <condition field="destination_number" expression="^\S*$"
> break="never">.
>        <action application="set" data="transfer_ringback=${us-ring}"/>.
>        <action application="set" data="dialed_ext=208"/>.
>        <action application="set" data="effective_caller_id_number=200"/>.
>        <action application="set" data="effective_caller_id_name=Peter "/>.
>        <action application="db"
> data="insert/call_return/${dialed_ext}/${caller_id_number}"/>.
>        <action application="db"
> data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>.
>        <action application="set" data="dialed_ext=208"/>.
>        <action application="export" data="dialed_ext=208"/>.
>        <action application="ring_ready" />.
>        <action application="bridge" data="user/208 at my.domain"/>
>      </condition>.
>    </extension>
>
> Just tested it with an Aastra Phone without TLS: Same Behaviour.
>
> Anybody has a clue how to solve this?
>
> 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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nico at vthadden.de  Wed Apr 28 09:55:28 2010
From: nico at vthadden.de (Nicola von Thadden)
Date: Wed, 28 Apr 2010 18:55:28 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
Message-ID: <4BD86880.4090607@vthadden.de>

Hi,

is there anything new since august?
I have the same problem with latest FritzBox firmenware and latest
FreeSwitch installed (via Quick & Dirty install).

FritzBox always sends 30ms in one paket, FS starts with 20 and then
switches to 30 (maybe thats the problem?)

Calling the Demo IVR at 5000 is (with default options and latest FS) no
problem, both uses 30ms ptime, have no idea why.

Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30.

Conferences are also choppy.

Adding <param name="rtp-autofix-timing" value="false" />
to 1000.xml (account used at the FBox) does not help.

Anyone got an idea?

Nico


From anthony.minessale at gmail.com  Wed Apr 28 13:26:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 15:26:54 -0500
Subject: [Freeswitch-users] effective_caller_id_name not working while
	calling multiple endpoints with :_:
In-Reply-To: <4BD85CBC.4080200@gmx.net>
References: <4BCED007.80900@gmx.net>
	<66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>
	<l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>
	<4BD16ABE.2030506@gmx.net> <4BD85CBC.4080200@gmx.net>
Message-ID: <k2q191c3a031004281326t76ff9529g99424cf7f7903a60@mail.gmail.com>

effective_caller_id_name does not go in the brackets
you want origination_caller_id_name and origination_caller_id_number inside
the <> and/or {}



On Wed, Apr 28, 2010 at 11:05 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Nobody has an idea how to solve this? Shall I open a JIRA?
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
> > Hello Anthony,
> >
> > I upgraded to newest GIT and tried it
> >
> > The dialplan now contains the fowllowing:
> > <action application="bridge" data="<effective_caller_id_name=My
> > Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain
> :_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain
> :_:user/230 at my.domain"/>
> >
> > When the dialplan is executed, it seems to be processed correctly:
> > EXECUTE sofia/local/06912345678 at 192.168.178.218:5060
> > bridge(<effective_caller_id_name=My
> > Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain
> :_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain
> :_:user/230 at my.domain)
> > 2010-04-23 10:59:12.479598 [DEBUG] switch_ivr_originate.c:1394 variable
> > string 0 = [effective_caller_id_name=My Name]
> > 2010-04-23 10:59:12.501255 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 0 = [global_to_originate_1=true]
> > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 0 = [presence_id=200 at my.domain]
> > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 1 = [transfer_fallback_extension=200]
> >
> > However the INVITE message does not contain the caller_id_name, see below
> >
> >
> > What am I doing wrong?
> >
> > Best regards
> > Peter
> >
> > U 192.168.178.220:5060 -> 192.168.178.50:3072
> > INVITE sip:200 at 192.168.178.50:3072;line=v3bii5l2 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bK4jUFH215p85tr.
> > Max-Forwards: 70.
> > From: "06912345678" <sip:06912345678 at 192.168.178.220<sip%3A06912345678 at 192.168.178.220>
> >;tag=a8m8ccQcgjjUg.
> > To: <sip:200 at 192.168.178.50:3072;line=v3bii5l2>.
> > Call-ID: b6cafa23-c959-122d-4682-080027e51f59.
> > CSeq: 129888323 INVITE.
> > Contact: <sip:mod_sofia at 192.168.178.220:5060>.
> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer.
> > Content-Type: application/sdp.
> > Content-Disposition: session.
> > Content-Length: 348.
> > X-FS-Support: update_display.
> > Remote-Party-ID: "06912345678"
> > <sip:06912345678 at 192.168.178.220 <sip%3A06912345678 at 192.168.178.220>
> >;party=calling;screen=yes;privacy=off.
> > .
> > v=0.
> > o=FreeSWITCH 1272001222 1272001223 IN IP4 192.168.178.220.
> > s=FreeSWITCH.
> > c=IN IP4 192.168.178.220.
> > t=0 0.
> > m=audio 12096 RTP/AVP 9 0 8 99 3 101 13.
> > a=rtpmap:9 G722/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:99 SPEEX/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.
> >
> > Anthony Minessale schrieb:
> >
> >> when using enterprise_originate you must use the special leading <>
> >> brackets to insert global variables meant for each tier 1 originate
> >>
> >>
> <global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/
> foo at bar.com
> >> <mailto:foo at bar.com>,sofia/internal/foo2 at bar.com:_:sofia/internal/
> foo3 at bar3.com
> >> <mailto:foo3 at bar3.com>
> >>
> >>
> >> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone
> >> <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>> wrote:
> >>
> >>     I think you should first thing update to latest GIT :)
> >>
> >>     David Ponzone  Direction Technique
> >>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> >>     tel:      01 74 03 18 97
> >>     gsm:   06 66 98 76 34
> >>
> >>     Service Client IPeva
> >>     tel:      0811 46 26 26
> >>     www.ipeva.fr  -   www.ipeva-studio.com
> >>
> >>     /Ce message et toutes les pi?ces jointes sont confidentiels et
> >>     ?tablis ? l'intention exclusive de ses destinataires. Toute
> >>     utilisation ou diffusion non autoris?e est interdite. Tout message
> >>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline
> >>     toute responsabilit? au titre de ce message s'il a ?t? alt?r?,
> >>     d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >>     message, merci de le d?truire imm?diatement et d'avertir
> >>     l'exp?diteur./
> >>     /
> >>     /
> >>
> >>
> >>
> >>     Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :
> >>
> >>
> >>>     Setting the effective_caller_id_name when dialing multiple
> endpoints
> >>>     with :_: do not seem to work.
> >>>     See example:
> >>>      <action application="set" data="effective_caller_id_name=MyName"/>
> >>>      <action application="bridge"
> >>>
> >>>     data="user/30 at mydomain.com
> >>>     <mailto:user/30 at mydomain.com>:_:user/31 at mydomain.com
> >>>     <mailto:user/31 at mydomain.com>:_:user/32 at mydomain.com
> >>>     <mailto:user/32 at mydomain.com>:_:user/33 at mydomain.com
> >>>     <mailto:user/33 at mydomain.com>:_:user/34 at mydomain.com
> >>>     <mailto:user/34 at mydomain.com>"/>
> >>>
> >>>     Freeswitch tries to set it:
> >>>       EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414
> >>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060
> >>>     set(effective_caller_id_name=MyName)
> >>>       2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
> >>>     sofia/external/069xxxxxxxx at 10.xx.xx.1414
> >>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060 SET
> >>>     [effective_caller_id_name]=[MyName]
> >>>
> >>>     But the SIP messages do not contain the effective_caller_id_name.
> >>>
> >>>     If we change the ":_:" sperator to "," then the
> >>>     effective_caller_id_name
> >>>     is correctly submittted (hower I cannot call
> >>>     multiple-registrations on
> >>>     one number then).
> >>>
> >>>     We are on
> >>>      FreeSWITCH Version 1.0.head (svn-17188)
> >>>
> >>>     Any ideas how to overcome this? Or shall I open a JIRA?
> >>>
> >>>     Best regards
> >>>     Peter
> >>>
> >>>     See example SIP message:
> >>>
> >>>     U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
> >>>     INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
> >>>     Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
> >>>     Max-Forwards: 70.
> >>>     From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141
> >;tag=5evr6508K9S3K.
> >>>     To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
> >>>     Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> >>>     CSeq: 129803060 INVITE.
> >>>     Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
> >>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> >>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>     REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>     Supported: timer, precondition, path, replaces.
> >>>     Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>     include-session-description, presence.winfo, message-summary,
> refer.
> >>>     Content-Type: application/sdp.
> >>>     Content-Disposition: session.
> >>>     Content-Length: 920.
> >>>     X-FS-Support: update_display.
> >>>     Remote-Party-ID: "069xxxxxxxx"
> >>>     <sip:069xxxxxxxx at 10.xx.xx.141
> >;party=calling;screen=yes;privacy=off.
> >>>     .
> >>>     v=0.
> >>>     o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
> >>>     s=FreeSWITCH.
> >>>     c=IN IP4 10.xx.xx.141.
> >>>     t=0 0.
> >>>     m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9
> >>>     121 3
> >>>     101 13.
> >>>     a=rtpmap:115 G7221/32000.
> >>>     a=fmtp:115 bitrate=48000.
> >>>     a=rtpmap:96 AMR/8000.
> >>>     a=fmtp:96 octet-align=0.
> >>>     a=rtpmap:99 SPEEX/32000.
> >>>     a=rtpmap:18 G729/8000.
> >>>     a=rtpmap:4 G723/8000.
> >>>     a=rtpmap:7 LPC/8000.
> >>>     a=rtpmap:124 G726-16/8000.
> >>>     a=rtpmap:8 PCMA/8000.
> >>>     a=rtpmap:6 DVI4/16000.
> >>>     a=rtpmap:123 G726-24/8000.
> >>>     a=rtpmap:0 PCMU/8000.
> >>>     a=rtpmap:10 L16/22050.
> >>>     a=rtpmap:98 iLBC/8000.
> >>>     a=fmtp:98 mode
> >>>     #
> >>>
> >>>
> >>>
> >>>     _______________________________________________
> >>>     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/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> <mailto: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>
> >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.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>
> >> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:+19193869900
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Apr 28 14:04:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 16:04:08 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD86880.4090607@vthadden.de>
References: <4BD86880.4090607@vthadden.de>
Message-ID: <h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>

Do you understand that the problem is with your device and not FS?
So why do you think something would have changed?

here is what you can do to work around it:

look for this param in you sip profile config:
<param name="inbound-codec-negotiation" value="generous"/>

change it to the following:
<param name="inbound-codec-negotiation" value="scrooge"/>

restart and retry

if this is not enough also edit vars.xml

look for
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

replace it with:

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>

retry




On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico at vthadden.de>wrote:

> Hi,
>
> is there anything new since august?
> I have the same problem with latest FritzBox firmenware and latest
> FreeSwitch installed (via Quick & Dirty install).
>
> FritzBox always sends 30ms in one paket, FS starts with 20 and then
> switches to 30 (maybe thats the problem?)
>
> Calling the Demo IVR at 5000 is (with default options and latest FS) no
> problem, both uses 30ms ptime, have no idea why.
>
> Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox
> 30.
>
> Conferences are also choppy.
>
> Adding <param name="rtp-autofix-timing" value="false" />
> to 1000.xml (account used at the FBox) does not help.
>
> Anyone got an idea?
>
> Nico
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From nico at vthadden.de  Wed Apr 28 14:17:00 2010
From: nico at vthadden.de (Nicola von Thadden)
Date: Wed, 28 Apr 2010 23:17:00 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
References: <4BD86880.4090607@vthadden.de>
	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
Message-ID: <4BD8A5CC.8060107@vthadden.de>

Changing those values didn't help

I wonder why then it works with all other servers.
Also, if I call the 9888 (so 888 at conference.freeswitch.org) it works
without problems, i wonder what they have changed in the config O.o

On 28.04.2010 23:04, Anthony Minessale wrote:
> Do you understand that the problem is with your device and not FS?
> So why do you think something would have changed?
> 
> here is what you can do to work around it:
> 
> look for this param in you sip profile config:
> <param name="inbound-codec-negotiation" value="generous"/>
> 
> change it to the following:
> <param name="inbound-codec-negotiation" value="scrooge"/>
> 
> restart and retry
> 
> if this is not enough also edit vars.xml
> 
> look for 
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> 
> replace it with:
> 
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
> 
> retry
> 
> 
> 
> 
> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico at vthadden.de
> <mailto:nico at vthadden.de>> wrote:
> 
>     Hi,
> 
>     is there anything new since august?
>     I have the same problem with latest FritzBox firmenware and latest
>     FreeSwitch installed (via Quick & Dirty install).
> 
>     FritzBox always sends 30ms in one paket, FS starts with 20 and then
>     switches to 30 (maybe thats the problem?)
> 
>     Calling the Demo IVR at 5000 is (with default options and latest FS) no
>     problem, both uses 30ms ptime, have no idea why.
> 
>     Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime,
>     FBox 30.
> 
>     Conferences are also choppy.
> 
>     Adding <param name="rtp-autofix-timing" value="false" />
>     to 1000.xml (account used at the FBox) does not help.
> 
>     Anyone got an idea?
> 
>     Nico
> 
>     _______________________________________________
>     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/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Apr 28 14:17:31 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 28 Apr 2010 16:17:31 -0500
Subject: [Freeswitch-users] Dial Timeout Issue
Message-ID: <05d601cae718$36799710$a36cc530$@com>

I'm having an issue where originate_timeout doesn't seem to be working as I
think it would.  I am using lcr to get the gateways.  I can bridge to either
gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
for failover and it worked fine.  I decided to change gateway1 to a fake IP
to test failover and it works, but it takes 30 seconds to timeout.  I then
found originate_timeout and set it to 4 seconds, but it doesn't timeout.  It
still takes 30 seconds.  


Here is the snippet from my dialplan:

<action application="set" data="originate_timeout=4"/>
<action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>


Here is the debug that shows it is being set:

EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
EXECUTE sofia/internal/1111 at 192.168.1.108
bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)

Then I get this:
2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]


And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
I would guess.  Am I using the timeout wrong?  Or is there some other
setting I need instead?



From anthony.minessale at gmail.com  Wed Apr 28 14:26:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 16:26:57 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD8A5CC.8060107@vthadden.de>
References: <4BD86880.4090607@vthadden.de>
	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
	<4BD8A5CC.8060107@vthadden.de>
Message-ID: <s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>

nothing it's default config on latest git HEAD


On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de>wrote:

> Changing those values didn't help
>
> I wonder why then it works with all other servers.
> Also, if I call the 9888 (so 888 at conference.freeswitch.org) it works
> without problems, i wonder what they have changed in the config O.o
>
> On 28.04.2010 23:04, Anthony Minessale wrote:
> > Do you understand that the problem is with your device and not FS?
> > So why do you think something would have changed?
> >
> > here is what you can do to work around it:
> >
> > look for this param in you sip profile config:
> > <param name="inbound-codec-negotiation" value="generous"/>
> >
> > change it to the following:
> > <param name="inbound-codec-negotiation" value="scrooge"/>
> >
> > restart and retry
> >
> > if this is not enough also edit vars.xml
> >
> > look for
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >
> > replace it with:
> >
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
> >
> > retry
> >
> >
> >
> >
> > On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico at vthadden.de
> > <mailto:nico at vthadden.de>> wrote:
> >
> >     Hi,
> >
> >     is there anything new since august?
> >     I have the same problem with latest FritzBox firmenware and latest
> >     FreeSwitch installed (via Quick & Dirty install).
> >
> >     FritzBox always sends 30ms in one paket, FS starts with 20 and then
> >     switches to 30 (maybe thats the problem?)
> >
> >     Calling the Demo IVR at 5000 is (with default options and latest FS)
> no
> >     problem, both uses 30ms ptime, have no idea why.
> >
> >     Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime,
> >     FBox 30.
> >
> >     Conferences are also choppy.
> >
> >     Adding <param name="rtp-autofix-timing" value="false" />
> >     to 1000.xml (account used at the FBox) does not help.
> >
> >     Anyone got an idea?
> >
> >     Nico
> >
> >     _______________________________________________
> >     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/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto: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>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.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>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From pjintheusa at gmail.com  Wed Apr 28 14:37:08 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 28 Apr 2010 17:37:08 -0400
Subject: [Freeswitch-users] FS Cluster
Message-ID: <z2y367751821004281437o41339cc2wa578bb12786d40d@mail.gmail.com>

Hi there,

I am sure this has been asked before but I can not find any reference to the
subject in the wiki.

Basically - I have two FreeSWITCH servers, FS1 and FS2. Both sit behind
OpenSIPS for load balancing

Phone A is registered on FS1. A call comes in on FS2 for phone A.

How do I get that call across to FS1.

So far I have the internal profile of FS1 and FS2 pointing to the same ODBC
database.

What else do I have to do?

If anyone can point me in the right direction I would be grateful.

Thanks

Pj
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From anthony.minessale at gmail.com  Wed Apr 28 14:45:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Apr 2010 16:45:34 -0500
Subject: [Freeswitch-users] FS Cluster
In-Reply-To: <z2y367751821004281437o41339cc2wa578bb12786d40d@mail.gmail.com>
References: <z2y367751821004281437o41339cc2wa578bb12786d40d@mail.gmail.com>
Message-ID: <z2j191c3a031004281445z82ed6375vb7b8b2eb9483c1c3@mail.gmail.com>

choose a neutral domain name and set it on both boxes in vars.xml


On Wed, Apr 28, 2010 at 4:37 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I am sure this has been asked before but I can not find any reference to
> the subject in the wiki.
>
> Basically - I have two FreeSWITCH servers, FS1 and FS2. Both sit behind
> OpenSIPS for load balancing
>
> Phone A is registered on FS1. A call comes in on FS2 for phone A.
>
> How do I get that call across to FS1.
>
> So far I have the internal profile of FS1 and FS2 pointing to the same ODBC
> database.
>
> What else do I have to do?
>
> If anyone can point me in the right direction I would be grateful.
>
> Thanks
>
> Pj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Wed Apr 28 15:02:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 15:02:39 -0700
Subject: [Freeswitch-users] say phrase into recordFile using lua script
In-Reply-To: <4BD72711.5060605@cronomagic.com>
References: <4BD72711.5060605@cronomagic.com>
Message-ID: <z2y87f2f3b91004281502n83e38c8duceffed7bf4c719cc@mail.gmail.com>

On Tue, Apr 27, 2010 at 11:04 AM, Jesse Cloutier <jesse at cronomagic.com>wrote:

>  Hello, I am having trouble with a lua script that i am trying to put
> together.
>
> The script basically makes a recording using the "session:recordFile()"
> function, and is terminated by a callback function that returns "break" on
> dtmf "#"
>
> What I would like to do is have the system say the date and time (
> session:say() ) into the recording at the end of the recording. It should be
> triggered by
> the call back.
>
> When I try it out it works great, I hear the system say the date / time,
> except that it is never recorded into the recording.
>
> I have been searching the docs for something that might clear up why its
> not recording but I can't find anything. I thought maybe doing a
> uuid_broadcast would fix the problem
> but there is no builtin function for that in the lua api and I was not able
> to make it work using the "session:execute" function either.
>
> Any help would be really appreaciated!
>
> This is the relavent code, for testing purposes I am having the date spoken
> using an alternative digit, you can also see my attempt to do a
> uuid_broadcast and sched_broadcast commented out:
>
> ######################
> --call back function
> function onInputRecord(s, type, obj) -- function to end recording
>
>   if (type == "dtmf") then
>       if ( obj['digit'] == '1') then
>       curDate = os.date();
>         session:say(curDate, "en", "CURRENT_DATE_TIME", "pronounced");
>
>         --uuidBroadcast = " say::en CURRENT_DATE_TIME pronounced " ..
> os.date() .. " both";
>         --session:execute("uuid_broadcast ".. UUID, uuidBroadcast);
>         --session:execute("sched_broadcast", "data=\"+1
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-hello.wav both\"");
>

uuid_broadcast and sched_broadcast are not dialplan apps that you execute,
they are API commands. You need an API object. Try this:
http://wiki.freeswitch.org/wiki/Lua#For_making_API_calls

Try it out and let us know what happens.
-MC
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From msc at freeswitch.org  Wed Apr 28 15:04:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 15:04:35 -0700
Subject: [Freeswitch-users] Intercom w/ Mute
In-Reply-To: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>
References: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>
Message-ID: <l2r87f2f3b91004281504l6903c54em261b2b25cca6da62@mail.gmail.com>

On Tue, Apr 27, 2010 at 2:29 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Is there support for dialing an extension and automatically answering and
> muting the dialed extension? I found this page on the wiki, but it doesn't
> mention anything about automaticaly muting the call?
>
>      http://wiki.freeswitch.org/wiki/Intercom
>
> Cheers,
> Gabe
>

Just curious - what are you doing that requires a mute? In any case,
possibly the methods presented here would be of use in your scenario:
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
-MC
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From msc at freeswitch.org  Wed Apr 28 17:00:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 17:00:07 -0700
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <05d601cae718$36799710$a36cc530$@com>
References: <05d601cae718$36799710$a36cc530$@com>
Message-ID: <l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>

Is it failing because gateway1 is not responding at all? If so then I'd say
you'd need to use leg_timeout or leg_media_timeout on each leg as needed:

<action application="bridge"
data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>

Check these variables:
http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout

Test it out and let us know.
-MC

On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:

> I'm having an issue where originate_timeout doesn't seem to be working as I
> think it would.  I am using lcr to get the gateways.  I can bridge to
> either
> gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
> for failover and it worked fine.  I decided to change gateway1 to a fake IP
> to test failover and it works, but it takes 30 seconds to timeout.  I then
> found originate_timeout and set it to 4 seconds, but it doesn't timeout.
>  It
> still takes 30 seconds.
>
>
> Here is the snippet from my dialplan:
>
> <action application="set" data="originate_timeout=4"/>
> <action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>
>
>
> Here is the debug that shows it is being set:
>
> EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
> 2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
> sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
> EXECUTE sofia/internal/1111 at 192.168.1.108
>
> bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
> ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>
> Then I get this:
> 2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
> sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
>
>
> And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
> I would guess.  Am I using the timeout wrong?  Or is there some other
> setting I need 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
>
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From msc at freeswitch.org  Wed Apr 28 17:09:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 17:09:27 -0700
Subject: [Freeswitch-users] Functionality
In-Reply-To: <C7FDB63F.902F%sean@obscuradigital.com>
References: <C7FDB63F.902F%sean@obscuradigital.com>
Message-ID: <r2l87f2f3b91004281709me8fc7dcn9c9ab819223c54d5@mail.gmail.com>

Hmm...

Are you using static XML dialplan stuff? Possibly you could use the
mod_limit or db/hash stuff to store the state of the auto attendant and then
have an extension that toggles it? Just throwing ideas out...

-MC

On Wed, Apr 28, 2010 at 9:54 AM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hello list,
>
> I?ve been looking through the wiki to figure the best way to enable the
> receptionist to turn on or off an ivr after hours auto-attendant.  I?m
> thinking the receptionist can enter a couple digits to control this
> functionality.  Can someone provide an example or show me in the wiki a good
> place to figure this out
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Wed Apr 28 17:41:46 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Wed, 28 Apr 2010 17:41:46 -0700
Subject: [Freeswitch-users] Functionality
In-Reply-To: <r2l87f2f3b91004281709me8fc7dcn9c9ab819223c54d5@mail.gmail.com>
Message-ID: <C7FE23DA.906E%sean@obscuradigital.com>

Well I?m new to FS so probably go with the static dialplan approach.  Not to
familiar with the db/hash, but that might work.  Any suggestions how to
write that into the dialplan?

Thanks
Sean


On 4/28/10 5:09 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Hmm...
> 
> Are you using static XML dialplan stuff? Possibly you could use the mod_limit
> or db/hash stuff to store the state of the auto attendant and then have an
> extension that toggles it? Just throwing ideas out...
> 
> -MC
> 
> On Wed, Apr 28, 2010 at 9:54 AM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello list,
>> 
>> I?ve been looking through the wiki to figure the best way to enable the
>> receptionist to turn on or off an ivr after hours auto-attendant. ?I?m
>> thinking the receptionist can enter a couple digits to control this
>> functionality. ?Can someone provide an example or show me in the wiki a good
>> place to figure this out
>> 
>> Thanks
>> Sean 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gkuri at ieee.org  Wed Apr 28 17:43:17 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 28 Apr 2010 17:43:17 -0700
Subject: [Freeswitch-users] Intercom w/ Mute
In-Reply-To: <l2r87f2f3b91004281504l6903c54em261b2b25cca6da62@mail.gmail.com>
References: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>
	<l2r87f2f3b91004281504l6903c54em261b2b25cca6da62@mail.gmail.com>
Message-ID: <l2z8b1c9cda1004281743gf0f58c48o74892654c1c647a4@mail.gmail.com>

Well, I guess you can say the request is stemming from a matter of "privacy"
;) ...

A user is paranoid someone can dial their extension via intercom and not
realize what's going on and allow the user to listen in to what's going on
in their office without them knowing. The intercom w/ mute allows the user's
phone to be automatically muted until the dialed user un-mutes it. I think
it's pretty common in commercial intercom implementations, although, I'm not
sure if it's a proprietary feature or how it's even implemented at the SIP
level.

Cheers,
Gabe

On Wed, Apr 28, 2010 at 3:04 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Apr 27, 2010 at 2:29 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
>> Is there support for dialing an extension and automatically answering and
>> muting the dialed extension? I found this page on the wiki, but it doesn't
>> mention anything about automaticaly muting the call?
>>
>>      http://wiki.freeswitch.org/wiki/Intercom
>>
>> Cheers,
>> Gabe
>>
>
> Just curious - what are you doing that requires a mute? In any case,
> possibly the methods presented here would be of use in your scenario:
> http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
> -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 jmesquita at freeswitch.org  Wed Apr 28 19:49:48 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 28 Apr 2010 23:49:48 -0300
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
References: <05d601cae718$36799710$a36cc530$@com>
	<l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
Message-ID: <r2oebbc2d551004281949v678e0ecfu1971c7aa460642e6@mail.gmail.com>

MSC,

I think you mean leg_progress_timeout instead of leg_media_timeout, right?

And actually, this is one good question. What is the difference between
leg_timeout and leg_progress_timeout? leg_timeout if till call completes and
leg_progress_timeout is only til 180/183? Is that the correct understanding?

JM


On Wed, Apr 28, 2010 at 9:00 PM, Michael Collins <msc at freeswitch.org> wrote:

> Is it failing because gateway1 is not responding at all? If so then I'd say
> you'd need to use leg_timeout or leg_media_timeout on each leg as needed:
>
> <action application="bridge"
> data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>
>
> Check these variables:
> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout
>
> Test it out and let us know.
> -MC
>
>
> On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:
>
>> I'm having an issue where originate_timeout doesn't seem to be working as
>> I
>> think it would.  I am using lcr to get the gateways.  I can bridge to
>> either
>> gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
>> for failover and it worked fine.  I decided to change gateway1 to a fake
>> IP
>> to test failover and it works, but it takes 30 seconds to timeout.  I then
>> found originate_timeout and set it to 4 seconds, but it doesn't timeout.
>>  It
>> still takes 30 seconds.
>>
>>
>> Here is the snippet from my dialplan:
>>
>> <action application="set" data="originate_timeout=4"/>
>> <action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>
>>
>>
>> Here is the debug that shows it is being set:
>>
>> EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
>> 2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
>> sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
>> EXECUTE sofia/internal/1111 at 192.168.1.108
>>
>> bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
>> ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>>
>> Then I get this:
>> 2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
>> sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
>>
>>
>> And it successfully goes out gateway2, but it is 30 seconds later, not 4
>> as
>> I would guess.  Am I using the timeout wrong?  Or is there some other
>> setting I need 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Wed Apr 28 22:35:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 29 Apr 2010 08:35:23 +0300
Subject: [Freeswitch-users] Intercom w/ Mute
In-Reply-To: <l2z8b1c9cda1004281743gf0f58c48o74892654c1c647a4@mail.gmail.com>
References: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>
	<l2r87f2f3b91004281504l6903c54em261b2b25cca6da62@mail.gmail.com>
	<l2z8b1c9cda1004281743gf0f58c48o74892654c1c647a4@mail.gmail.com>
Message-ID: <y2ld361ca7b1004282235s3c71af96z96c28bc93ff26bbe@mail.gmail.com>

That's how it is usually implemented on the "old key systems",; people are
used to it and want it.

When you call someone via the intercom the remote side answers automatically
but the microphone is muted. He/she has to pickup the handset/speaker to
have two way audio.

                       Regards, __Yehavi:



 2010/4/29 Gabriel Kuri <gkuri at ieee.org>

> Well, I guess you can say the request is stemming from a matter of
> "privacy" ;) ...
>
> A user is paranoid someone can dial their extension via intercom and not
> realize what's going on and allow the user to listen in to what's going on
> in their office without them knowing. The intercom w/ mute allows the user's
> phone to be automatically muted until the dialed user un-mutes it. I think
> it's pretty common in commercial intercom implementations, although, I'm not
> sure if it's a proprietary feature or how it's even implemented at the SIP
> level.
>
> Cheers,
> Gabe
>
>   On Wed, Apr 28, 2010 at 3:04 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>   On Tue, Apr 27, 2010 at 2:29 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>>> Is there support for dialing an extension and automatically answering and
>>> muting the dialed extension? I found this page on the wiki, but it doesn't
>>> mention anything about automaticaly muting the call?
>>>
>>>      http://wiki.freeswitch.org/wiki/Intercom
>>>
>>> Cheers,
>>> Gabe
>>>
>>
>> Just curious - what are you doing that requires a mute? In any case,
>> possibly the methods presented here would be of use in your scenario:
>> http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
>> -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 babak.freeswitch at gmail.com  Wed Apr 28 23:58:08 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 29 Apr 2010 11:28:08 +0430
Subject: [Freeswitch-users] mod_shout error on recording session!
Message-ID: <z2h1fdf92721004282358pe0a55798h37b1496388c6c8ac@mail.gmail.com>

Hi
I'm using mod_shout in mod_managed to record calls in mp3 format but after 4
or 5 seconds I get this:

2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 LAME 3.97 32bits (
http://www.m
p3dev.org/)
*Assertion failed: enn >= 0, file ..\..\lame-3.97\libmp3lame\psymodel.c,
line 497*

2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 Using polyphase lowpass
filter
, transition band:  3903 Hz -  4000 Hz
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From babak.freeswitch at gmail.com  Thu Apr 29 01:57:29 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 29 Apr 2010 13:27:29 +0430
Subject: [Freeswitch-users] build error
Message-ID: <i2y1fdf92721004290157jb0ca644cw599acb9e43d80362@mail.gmail.com>

I'm trying to build the lates svn and I got these errors
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065:
'msg_unknown_hash' : undeclared identifier
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From peter.olsson at visionutveckling.se  Thu Apr 29 02:08:32 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 29 Apr 2010 11:08:32 +0200
Subject: [Freeswitch-users] build error
In-Reply-To: <i2y1fdf92721004290157jb0ca644cw599acb9e43d80362@mail.gmail.com>
References: <i2y1fdf92721004290157jb0ca644cw599acb9e43d80362@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55777E0FE0@cooper>

It seems you are running in Windows, with a git checkout. Please make sure to set autocrlf=false in git, or else it will mess up som of the files. Change the setting and then check out everything from start.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r babak yakhchali
Skickat: den 29 april 2010 10:57
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] build error

I'm trying to build the lates svn and I got these errors
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_hash' : undeclared identifier
!DSPAM:4bd94c2732931909018391!
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From saeedahmad1981 at gmail.com  Thu Apr 29 03:10:05 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 29 Apr 2010 12:10:05 +0200
Subject: [Freeswitch-users] Accessing SQLITE core.db
Message-ID: <q2kc96b95651004290310va9fe1cdfn133c4073dd82ec0@mail.gmail.com>

Hi,

Is it safe to access sqlite db using PHP, when there are live calls on FS?

I am just sending selects to 'channels' table to view live calls.

Thanks
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From justin at ejtown.org  Thu Apr 29 03:27:56 2010
From: justin at ejtown.org (Justin B Newman)
Date: Thu, 29 Apr 2010 06:27:56 -0400
Subject: [Freeswitch-users] Accessing SQLITE core.db
In-Reply-To: <q2kc96b95651004290310va9fe1cdfn133c4073dd82ec0@mail.gmail.com>
References: <q2kc96b95651004290310va9fe1cdfn133c4073dd82ec0@mail.gmail.com>
Message-ID: <x2g27b0d5bc1004290327t8e8d59ffn59dd994032e60b7@mail.gmail.com>

On Thu, Apr 29, 2010 at 6:10 AM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>
> Is it safe to access sqlite db using PHP, when there are live calls on FS?
>
> I am just sending selects to 'channels' table to view live calls.
>

http://www.sqlite.org/faq.html#q5

-jbn


From saeedahmad1981 at gmail.com  Thu Apr 29 03:47:37 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 29 Apr 2010 12:47:37 +0200
Subject: [Freeswitch-users] Accessing SQLITE core.db
In-Reply-To: <x2g27b0d5bc1004290327t8e8d59ffn59dd994032e60b7@mail.gmail.com>
References: <q2kc96b95651004290310va9fe1cdfn133c4073dd82ec0@mail.gmail.com> 
	<x2g27b0d5bc1004290327t8e8d59ffn59dd994032e60b7@mail.gmail.com>
Message-ID: <p2qc96b95651004290347la4df1c24t8c02d2d9dac9de74@mail.gmail.com>

Thanks,

Since i am using it on Centos, so it seems that its safe to send 'select'
query to core.db.



On Thu, Apr 29, 2010 at 12:27 PM, Justin B Newman <justin at ejtown.org> wrote:

> On Thu, Apr 29, 2010 at 6:10 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
> >
> > Is it safe to access sqlite db using PHP, when there are live calls on
> FS?
> >
> > I am just sending selects to 'channels' table to view live calls.
> >
>
> http://www.sqlite.org/faq.html#q5
>
> -jbn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andy at fabulous4.co.uk  Thu Apr 29 04:48:41 2010
From: andy at fabulous4.co.uk (Andy)
Date: Thu, 29 Apr 2010 12:48:41 +0100
Subject: [Freeswitch-users] mod_shout recordFile bitrate
Message-ID: <A4E9548DC539423A9F7D9555E727E1D6@D810>

Hi folks,
 
I'm using  session.recordFile to push the audio from my calls to an icecast
server. As far as I can tell the bitrate of the audio I'm sending is only
64kbps but the info sent to icecast about the stream always says 24000kbps
which is causing me problems. 
 
Is there anyway I can correct the value sent in the info packet without
editing the source code of mod_shout? 
 
This may be a question for the developers list but, If I did decide to edit
the code in mod_shout.c is there a way I can set this to the actual bit rate
for the stream rather than hardcoding it to 64kbps which is actually always
true with my application.
 
Cheers
Andy
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From rupa at rupa.com  Thu Apr 29 05:47:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 07:47:17 -0500
Subject: [Freeswitch-users] mod_shout error on recording session!
In-Reply-To: <z2h1fdf92721004282358pe0a55798h37b1496388c6c8ac@mail.gmail.com>
References: <z2h1fdf92721004282358pe0a55798h37b1496388c6c8ac@mail.gmail.com>
Message-ID: <v2yb39273531004290547z89f94dd0nfb93374e32c942d9@mail.gmail.com>

There is something wrong with the underlying mp3 that lame (the mp3 decoder)
is catching.  Not much we can do about that and demonstrates the risk of
using mp3.

On Thu, Apr 29, 2010 at 1:58 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm using mod_shout in mod_managed to record calls in mp3 format but after
> 4 or 5 seconds I get this:
>
> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 LAME 3.97 32bits (
> http://www.m
> p3dev.org/)
> *Assertion failed: enn >= 0, file ..\..\lame-3.97\libmp3lame\psymodel.c,
> line 497*
>
> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 Using polyphase lowpass
> filter
> , transition band:  3903 Hz -  4000 Hz
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From fs-list at communicatefreely.net  Thu Apr 29 05:58:48 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 29 Apr 2010 08:58:48 -0400
Subject: [Freeswitch-users] Intercom w/ Mute
In-Reply-To: <y2ld361ca7b1004282235s3c71af96z96c28bc93ff26bbe@mail.gmail.com>
References: <j2j8b1c9cda1004271429y59ec999bna82e57b79b0347fc@mail.gmail.com>	<l2r87f2f3b91004281504l6903c54em261b2b25cca6da62@mail.gmail.com>	<l2z8b1c9cda1004281743gf0f58c48o74892654c1c647a4@mail.gmail.com>
	<y2ld361ca7b1004282235s3c71af96z96c28bc93ff26bbe@mail.gmail.com>
Message-ID: <4BD98288.7020701@communicatefreely.net>

With the Aastra phones, and I expect with others, you can set that in the phone's config.

There is usually a config option that tells the phone to answer an intercom call with the microphone
muted.  You can build an option in your provisioning system that sets the behaviour on a per
extension basis.

It wouldn't be something you can set on a per-call basis with this method, but it doesn't sound like
that is a problem in this case.

-Tim

Yehavi Bourvine wrote:
> That's how it is usually implemented on the "old key systems",; people
> are used to it and want it.
>  
> When you call someone via the intercom the remote side answers
> automatically but the microphone is muted. He/she has to pickup the
> handset/speaker to have two way audio.
>  
>                        Regards, __Yehavi:
> 
> 
>  
> 2010/4/29 Gabriel Kuri <gkuri at ieee.org <mailto:gkuri at ieee.org>>
> 
>     Well, I guess you can say the request is stemming from a matter of
>     "privacy" ;) ...
> 
>     A user is paranoid someone can dial their extension via intercom and
>     not realize what's going on and allow the user to listen in to
>     what's going on in their office without them knowing. The intercom
>     w/ mute allows the user's phone to be automatically muted until the
>     dialed user un-mutes it. I think it's pretty common in commercial
>     intercom implementations, although, I'm not sure if it's a
>     proprietary feature or how it's even implemented at the SIP level.
> 
>     Cheers,
>     Gabe
> 
>     On Wed, Apr 28, 2010 at 3:04 PM, Michael Collins <msc at freeswitch.org
>     <mailto:msc at freeswitch.org>> wrote:
> 
> 
> 
>         On Tue, Apr 27, 2010 at 2:29 PM, Gabriel Kuri <gkuri at ieee.org
>         <mailto:gkuri at ieee.org>> wrote:
> 
>             Is there support for dialing an extension and automatically
>             answering and muting the dialed extension? I found this page
>             on the wiki, but it doesn't mention anything about
>             automaticaly muting the call?
> 
>                  http://wiki.freeswitch.org/wiki/Intercom
> 
>             Cheers,
>             Gabe
> 
> 
>         Just curious - what are you doing that requires a mute? In any
>         case, possibly the methods presented here would be of use in
>         your scenario:
>         http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
>         -MC
> 
> 
>         _______________________________________________
>         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 <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 <http://www.freeswitch.org/>
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From david.ponzone at gmail.com  Thu Apr 29 06:01:01 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 29 Apr 2010 15:01:01 +0200
Subject: [Freeswitch-users] mod_shout error on recording session!
In-Reply-To: <v2yb39273531004290547z89f94dd0nfb93374e32c942d9@mail.gmail.com>
References: <z2h1fdf92721004282358pe0a55798h37b1496388c6c8ac@mail.gmail.com>
	<v2yb39273531004290547z89f94dd0nfb93374e32c942d9@mail.gmail.com>
Message-ID: <EA8062A6-4048-4724-A524-06608E3F5ADE@gmail.com>

Rupa,

I think he said he was recording.
So there is no underlying mp3, isnt there ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 29/04/2010 ? 14:47, Rupa Schomaker a ?crit :

> There is something wrong with the underlying mp3 that lame (the mp3  
> decoder) is catching.  Not much we can do about that and  
> demonstrates the risk of using mp3.
>
> On Thu, Apr 29, 2010 at 1:58 AM, babak yakhchali <babak.freeswitch at gmail.com 
> > wrote:
> Hi
> I'm using mod_shout in mod_managed to record calls in mp3 format but  
> after 4 or 5 seconds I get this:
>
> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 LAME 3.97 32bits (http://www 
> .m
> p3dev.org/)
> Assertion failed: enn >= 0, file ..\..\lame-3.97\libmp3lame 
> \psymodel.c, line 497
>
> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 Using polyphase  
> lowpass filter
> , transition band:  3903 Hz -  4000 Hz
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Thu Apr 29 06:27:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 08:27:48 -0500
Subject: [Freeswitch-users] mod_shout error on recording session!
In-Reply-To: <EA8062A6-4048-4724-A524-06608E3F5ADE@gmail.com>
References: <z2h1fdf92721004282358pe0a55798h37b1496388c6c8ac@mail.gmail.com>
	<v2yb39273531004290547z89f94dd0nfb93374e32c942d9@mail.gmail.com>
	<EA8062A6-4048-4724-A524-06608E3F5ADE@gmail.com>
Message-ID: <y2sb39273531004290627j8920314ah7bc1f34de8af92a@mail.gmail.com>

ack, good point.  You are right.

babak, can you record to wav and then a) listen to the wav and ensure it
sounds like it should and then b) try to convert that to mp3 using
commandline lame?

On Thu, Apr 29, 2010 at 8:01 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Rupa,
>
> I think he said he was recording.
> So there is no underlying mp3, isnt there ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 29/04/2010 ? 14:47, Rupa Schomaker a ?crit :
>
> There is something wrong with the underlying mp3 that lame (the mp3
> decoder) is catching.  Not much we can do about that and demonstrates the
> risk of using mp3.
>
> On Thu, Apr 29, 2010 at 1:58 AM, babak yakhchali <
> babak.freeswitch at gmail.com> wrote:
>
>> Hi
>> I'm using mod_shout in mod_managed to record calls in mp3 format but after
>> 4 or 5 seconds I get this:
>>
>> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 LAME 3.97 32bits (
>> http://www.m
>> p3dev.org/)
>> *Assertion failed: enn >= 0, file ..\..\lame-3.97\libmp3lame\psymodel.c,
>> line 497*
>>
>> 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 Using polyphase lowpass
>> filter
>> , transition band:  3903 Hz -  4000 Hz
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Thu Apr 29 06:32:10 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 08:32:10 -0500
Subject: [Freeswitch-users] mod_shout recordFile bitrate
In-Reply-To: <A4E9548DC539423A9F7D9555E727E1D6@D810>
References: <A4E9548DC539423A9F7D9555E727E1D6@D810>
Message-ID: <h2nb39273531004290632zbadb815fz2c72c4bd1e010a8a@mail.gmail.com>

The bitrate is "hard coded" to 24k (mod_shout.c:769).  When you say the
bitrate of the audio you are sending is 64k, what codec?  mp3 is a
compressed format.  Trying to match the bitrate of either a uncompressed
codec or other compression is not desirable.  Instead choose a bitrate that
is acoustically transparent and still low enough to be useful.

On Thu, Apr 29, 2010 at 6:48 AM, Andy <andy at fabulous4.co.uk> wrote:

>  Hi folks,
>
> I'm using  session.recordFile to push the audio from my calls to an icecast
> server. As far as I can tell the bitrate of the audio I'm sending is only
> 64kbps but the info sent to icecast about the stream always says 24000kbps
> which is causing me problems.
>
> Is there anyway I can correct the value sent in the info packet without
> editing the source code of mod_shout?
>
> This may be a question for the developers list but, If I did decide to edit
> the code in mod_shout.c is there a way I can set this to the actual bit rate
> for the stream rather than hardcoding it to 64kbps which is actually always
> true with my application.
>
> Cheers
> Andy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From Prometheus001 at gmx.net  Thu Apr 29 06:44:55 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 29 Apr 2010 15:44:55 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>
	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
Message-ID: <4BD98D57.7030100@gmx.net>

Hello Nicola,

we had the same problem with the FritzBoxes. Someweher in the internet
we found an advice to patch Freeswitch. See my previous comment on this
patch here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html

Best regards
Peter


Anthony Minessale schrieb:
> nothing it's default config on latest git HEAD
>
>
> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
> <mailto:nico at vthadden.de>> wrote:
>
>     Changing those values didn't help
>
>     I wonder why then it works with all other servers.
>     Also, if I call the 9888 (so 888 at conference.freeswitch.org
>     <mailto:888 at conference.freeswitch.org>) it works
>     without problems, i wonder what they have changed in the config O.o
>
>     On 28.04.2010 23:04, Anthony Minessale wrote:
>     > Do you understand that the problem is with your device and not FS?
>     > So why do you think something would have changed?
>     >
>     > here is what you can do to work around it:
>     >
>     > look for this param in you sip profile config:
>     > <param name="inbound-codec-negotiation" value="generous"/>
>     >
>     > change it to the following:
>     > <param name="inbound-codec-negotiation" value="scrooge"/>
>     >
>     > restart and retry
>     >
>     > if this is not enough also edit vars.xml
>     >
>     > look for
>     > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>     >
>     > replace it with:
>     >
>     > <X-PRE-PROCESS cmd="set"
>     data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>     >
>     > retry
>     >
>     >
>     >
>     >
>     > On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>     <nico at vthadden.de <mailto:nico at vthadden.de>
>     > <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>     >
>     >     Hi,
>     >
>     >     is there anything new since august?
>     >     I have the same problem with latest FritzBox firmenware and
>     latest
>     >     FreeSwitch installed (via Quick & Dirty install).
>     >
>     >     FritzBox always sends 30ms in one paket, FS starts with 20
>     and then
>     >     switches to 30 (maybe thats the problem?)
>     >
>     >     Calling the Demo IVR at 5000 is (with default options and
>     latest FS) no
>     >     problem, both uses 30ms ptime, have no idea why.
>     >
>     >     Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>     ptime,
>     >     FBox 30.
>     >
>     >     Conferences are also choppy.
>     >
>     >     Adding <param name="rtp-autofix-timing" value="false" />
>     >     to 1000.xml (account used at the FBox) does not help.
>     >
>     >     Anyone got an idea?
>     >
>     >     Nico
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto: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/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     >
>     >
>     > _______________________________________________
>     > 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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From mrene_lists at avgs.ca  Thu Apr 29 06:52:50 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 29 Apr 2010 09:52:50 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD98D57.7030100@gmx.net>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>
	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
	<4BD98D57.7030100@gmx.net>
Message-ID: <1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>

If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))

that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)

Perhaps didn't you rescan the sofia profile after your config change?


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-29, at 9:44 AM, Peter P GMX wrote:

> Hello Nicola,
> 
> we had the same problem with the FritzBoxes. Someweher in the internet
> we found an advice to patch Freeswitch. See my previous comment on this
> patch here:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
> 
> Best regards
> Peter
> 
> 
> Anthony Minessale schrieb:
>> nothing it's default config on latest git HEAD
>> 
>> 
>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
>> <mailto:nico at vthadden.de>> wrote:
>> 
>>    Changing those values didn't help
>> 
>>    I wonder why then it works with all other servers.
>>    Also, if I call the 9888 (so 888 at conference.freeswitch.org
>>    <mailto:888 at conference.freeswitch.org>) it works
>>    without problems, i wonder what they have changed in the config O.o
>> 
>>    On 28.04.2010 23:04, Anthony Minessale wrote:
>>> Do you understand that the problem is with your device and not FS?
>>> So why do you think something would have changed?
>>> 
>>> here is what you can do to work around it:
>>> 
>>> look for this param in you sip profile config:
>>> <param name="inbound-codec-negotiation" value="generous"/>
>>> 
>>> change it to the following:
>>> <param name="inbound-codec-negotiation" value="scrooge"/>
>>> 
>>> restart and retry
>>> 
>>> if this is not enough also edit vars.xml
>>> 
>>> look for
>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>> 
>>> replace it with:
>>> 
>>> <X-PRE-PROCESS cmd="set"
>>    data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>>> 
>>> retry
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>>    <nico at vthadden.de <mailto:nico at vthadden.de>
>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>>> 
>>>    Hi,
>>> 
>>>    is there anything new since august?
>>>    I have the same problem with latest FritzBox firmenware and
>>    latest
>>>    FreeSwitch installed (via Quick & Dirty install).
>>> 
>>>    FritzBox always sends 30ms in one paket, FS starts with 20
>>    and then
>>>    switches to 30 (maybe thats the problem?)
>>> 
>>>    Calling the Demo IVR at 5000 is (with default options and
>>    latest FS) no
>>>    problem, both uses 30ms ptime, have no idea why.
>>> 
>>>    Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>>    ptime,
>>>    FBox 30.
>>> 
>>>    Conferences are also choppy.
>>> 
>>>    Adding <param name="rtp-autofix-timing" value="false" />
>>>    to 1000.xml (account used at the FBox) does not help.
>>> 
>>>    Anyone got an idea?
>>> 
>>>    Nico
>>> 
>>>    _______________________________________________
>>>    FreeSWITCH-users mailing list
>>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>    <mailto: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/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>    <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> <mailto:MSN%3Aanthony_minessale at hotmail.com
>>    <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>    <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>    <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>> IRC: irc.freenode.net <http://irc.freenode.net>
>>    <http://irc.freenode.net> #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>    <mailto:sip%3A888 at conference.freeswitch.org>
>>> <mailto:sip%3A888 at conference.freeswitch.org
>>    <mailto:sip%253A888 at conference.freeswitch.org>>
>>> googletalk:conf+888 at conference.freeswitch.org
>>    <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>    <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>> pstn:+19193869900
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> 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/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> 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>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> ------------------------------------------------------------------------
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From andy at fabulous4.co.uk  Thu Apr 29 07:03:30 2010
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Thu, 29 Apr 2010 15:03:30 +0100
Subject: [Freeswitch-users] mod_shout recordFile bitrate
In-Reply-To: <h2nb39273531004290632zbadb815fz2c72c4bd1e010a8a@mail.gmail.com>
References: <A4E9548DC539423A9F7D9555E727E1D6@D810>
	<h2nb39273531004290632zbadb815fz2c72c4bd1e010a8a@mail.gmail.com>
Message-ID: <u2oe2bf58d11004290703xa304cc05p16493e36301eff2c@mail.gmail.com>

Thanks for the reply, I'm using session.recordFile and mp3 format. I've set
the sample rate for the audio in the call to 11kHz which results in a
bitrate of 16kbps (sorry I reported this wrongly in the first email). I'm
assuming the bitrate of the icecast stream will match the bitrate of the
audio in the call so it would be great if I could set the info sent to
icecast to 16kbps instead of 24000kbps as at present.

It appears this is not possible. No worries, I'll just change the hardcoded
value in my install.

Thanks

On 29 April 2010 14:32, Rupa Schomaker <rupa at rupa.com> wrote:

> The bitrate is "hard coded" to 24k (mod_shout.c:769).  When you say the
> bitrate of the audio you are sending is 64k, what codec?  mp3 is a
> compressed format.  Trying to match the bitrate of either a uncompressed
> codec or other compression is not desirable.  Instead choose a bitrate that
> is acoustically transparent and still low enough to be useful.
>
>   On Thu, Apr 29, 2010 at 6:48 AM, Andy <andy at fabulous4.co.uk> wrote:
>
>>   Hi folks,
>>
>> I'm using  session.recordFile to push the audio from my calls to an
>> icecast server. As far as I can tell the bitrate of the audio I'm sending is
>> only 64kbps but the info sent to icecast about the stream always says
>> 24000kbps which is causing me problems.
>>
>> Is there anyway I can correct the value sent in the info packet without
>> editing the source code of mod_shout?
>>
>> This may be a question for the developers list but, If I did decide to
>> edit the code in mod_shout.c is there a way I can set this to the actual bit
>> rate for the stream rather than hardcoding it to 64kbps which is actually
>> always true with my application.
>>
>> Cheers
>> Andy
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 29 07:08:37 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 29 Apr 2010 16:08:37 +0200
Subject: [Freeswitch-users] effective_caller_id_name not working while
 calling multiple endpoints with :_:
In-Reply-To: <k2q191c3a031004281326t76ff9529g99424cf7f7903a60@mail.gmail.com>
References: <4BCED007.80900@gmx.net>	<66FBE283-BEE7-4818-95D8-152F1D0A287A@gmail.com>	<l2h191c3a031004210639vc7eb56adjb2afa086cbd134cf@mail.gmail.com>	<4BD16ABE.2030506@gmx.net>
	<4BD85CBC.4080200@gmx.net>
	<k2q191c3a031004281326t76ff9529g99424cf7f7903a60@mail.gmail.com>
Message-ID: <4BD992E5.8010107@gmx.net>

Thanks Anthony, origination_caller_id_name did the trick!


Anthony Minessale schrieb:
> effective_caller_id_name does not go in the brackets
> you want origination_caller_id_name and origination_caller_id_number
> inside the <> and/or {}
>
>
>
> On Wed, Apr 28, 2010 at 11:05 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Nobody has an idea how to solve this? Shall I open a JIRA?
>
>     Best regards
>     Peter
>
>     Peter P GMX schrieb:
>     > Hello Anthony,
>     >
>     > I upgraded to newest GIT and tried it
>     >
>     > The dialplan now contains the fowllowing:
>     > <action application="bridge" data="<effective_caller_id_name=My
>     >
>     Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain"/>
>     >
>     > When the dialplan is executed, it seems to be processed correctly:
>     > EXECUTE sofia/local/06912345678 at 192.168.178.218:5060
>     <http://06912345678 at 192.168.178.218:5060>
>     > bridge(<effective_caller_id_name=My
>     >
>     Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain:_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain:_:user/230 at my.domain)
>     > 2010-04-23 10:59:12.479598 [DEBUG] switch_ivr_originate.c:1394
>     variable
>     > string 0 = [effective_caller_id_name=My Name]
>     > 2010-04-23 10:59:12.501255 [DEBUG] switch_ivr_originate.c:1885
>     variable
>     > string 0 = [global_to_originate_1=true]
>     > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885
>     variable
>     > string 0 = [presence_id=200 at my.domain]
>     > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885
>     variable
>     > string 1 = [transfer_fallback_extension=200]
>     >
>     > However the INVITE message does not contain the caller_id_name,
>     see below
>     >
>     >
>     > What am I doing wrong?
>     >
>     > Best regards
>     > Peter
>     >
>     > U 192.168.178.220:5060 <http://192.168.178.220:5060> ->
>     192.168.178.50:3072 <http://192.168.178.50:3072>
>     > INVITE sip:200 at 192.168.178.50:3072;line=v3bii5l2 SIP/2.0.
>     > Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bK4jUFH215p85tr.
>     > Max-Forwards: 70.
>     > From: "06912345678" <sip:06912345678 at 192.168.178.220
>     <mailto:sip%3A06912345678 at 192.168.178.220>>;tag=a8m8ccQcgjjUg.
>     > To: <sip:200 at 192.168.178.50:3072;line=v3bii5l2>.
>     > Call-ID: b6cafa23-c959-122d-4682-080027e51f59.
>     > CSeq: 129888323 INVITE.
>     > Contact: <sip:mod_sofia at 192.168.178.220:5060
>     <http://sip:mod_sofia at 192.168.178.220:5060>>.
>     > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-.
>     > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>     > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
>     > Supported: timer, precondition, path, replaces.
>     > Allow-Events: talk, hold, presence, dialog, line-seize,
>     call-info, sla,
>     > include-session-description, presence.winfo, message-summary, refer.
>     > Content-Type: application/sdp.
>     > Content-Disposition: session.
>     > Content-Length: 348.
>     > X-FS-Support: update_display.
>     > Remote-Party-ID: "06912345678"
>     > <sip:06912345678 at 192.168.178.220
>     <mailto:sip%3A06912345678 at 192.168.178.220>>;party=calling;screen=yes;privacy=off.
>     > .
>     > v=0.
>     > o=FreeSWITCH 1272001222 1272001223 IN IP4 192.168.178.220.
>     > s=FreeSWITCH.
>     > c=IN IP4 192.168.178.220.
>     > t=0 0.
>     > m=audio 12096 RTP/AVP 9 0 8 99 3 101 13.
>     > a=rtpmap:9 G722/8000.
>     > a=rtpmap:0 PCMU/8000.
>     > a=rtpmap:8 PCMA/8000.
>     > a=rtpmap:99 SPEEX/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.
>     >
>     > Anthony Minessale schrieb:
>     >
>     >> when using enterprise_originate you must use the special leading <>
>     >> brackets to insert global variables meant for each tier 1 originate
>     >>
>     >>
>     <global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/foo at bar.com
>     <mailto:foo at bar.com>
>     >> <mailto:foo at bar.com
>     <mailto:foo at bar.com>>,sofia/internal/foo2 at bar.com:_:sofia/internal/foo3 at bar3.com
>     <mailto:foo3 at bar3.com>
>     >> <mailto:foo3 at bar3.com <mailto:foo3 at bar3.com>>
>     >>
>     >>
>     >> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone
>     >> <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>
>     <mailto:david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>>>
>     wrote:
>     >>
>     >>     I think you should first thing update to latest GIT :)
>     >>
>     >>     David Ponzone  Direction Technique
>     >>     email: david.ponzone at ipeva.fr
>     <mailto:david.ponzone at ipeva.fr> <mailto:david.ponzone at ipeva.fr
>     <mailto:david.ponzone at ipeva.fr>>
>     >>     tel:      01 74 03 18 97
>     >>     gsm:   06 66 98 76 34
>     >>
>     >>     Service Client IPeva
>     >>     tel:      0811 46 26 26
>     >>     www.ipeva.fr <http://www.ipeva.fr>  -  
>     www.ipeva-studio.com <http://www.ipeva-studio.com>
>     >>
>     >>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>     >>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>     >>     utilisation ou diffusion non autoris?e est interdite. Tout
>     message
>     >>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline
>     >>     toute responsabilit? au titre de ce message s'il a ?t? alt?r?,
>     >>     d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>     >>     message, merci de le d?truire imm?diatement et d'avertir
>     >>     l'exp?diteur./
>     >>     /
>     >>     /
>     >>
>     >>
>     >>
>     >>     Le 21/04/2010 ? 12:14, Peter P GMX a ?crit :
>     >>
>     >>
>     >>>     Setting the effective_caller_id_name when dialing multiple
>     endpoints
>     >>>     with :_: do not seem to work.
>     >>>     See example:
>     >>>      <action application="set"
>     data="effective_caller_id_name=MyName"/>
>     >>>      <action application="bridge"
>     >>>
>     >>>     data="user/30 at mydomain.com <mailto:30 at mydomain.com>
>     >>>     <mailto:user <mailto:user>/30 at mydomain.com
>     <mailto:30 at mydomain.com>>:_:user/31 at mydomain.com
>     <mailto:31 at mydomain.com>
>     >>>     <mailto:user <mailto:user>/31 at mydomain.com
>     <mailto:31 at mydomain.com>>:_:user/32 at mydomain.com
>     <mailto:32 at mydomain.com>
>     >>>     <mailto:user <mailto:user>/32 at mydomain.com
>     <mailto:32 at mydomain.com>>:_:user/33 at mydomain.com
>     <mailto:33 at mydomain.com>
>     >>>     <mailto:user <mailto:user>/33 at mydomain.com
>     <mailto:33 at mydomain.com>>:_:user/34 at mydomain.com
>     <mailto:34 at mydomain.com>
>     >>>     <mailto:user <mailto:user>/34 at mydomain.com
>     <mailto:34 at mydomain.com>>"/>
>     >>>
>     >>>     Freeswitch tries to set it:
>     >>>       EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414
>     >>>     <mailto:sofia
>     <mailto:sofia>/external/069xxxxxxxx at 10.xx.xx.1414>:5060
>     >>>     set(effective_caller_id_name=MyName)
>     >>>       2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
>     >>>     sofia/external/069xxxxxxxx at 10.xx.xx.1414
>     >>>     <mailto:sofia
>     <mailto:sofia>/external/069xxxxxxxx at 10.xx.xx.1414>:5060 SET
>     >>>     [effective_caller_id_name]=[MyName]
>     >>>
>     >>>     But the SIP messages do not contain the
>     effective_caller_id_name.
>     >>>
>     >>>     If we change the ":_:" sperator to "," then the
>     >>>     effective_caller_id_name
>     >>>     is correctly submittted (hower I cannot call
>     >>>     multiple-registrations on
>     >>>     one number then).
>     >>>
>     >>>     We are on
>     >>>      FreeSWITCH Version 1.0.head (svn-17188)
>     >>>
>     >>>     Any ideas how to overcome this? Or shall I open a JIRA?
>     >>>
>     >>>     Best regards
>     >>>     Peter
>     >>>
>     >>>     See example SIP message:
>     >>>
>     >>>     U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
>     >>>     INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
>     >>>     Via: SIP/2.0/UDP
>     10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
>     >>>     Max-Forwards: 70.
>     >>>     From: "069xxxxxxxx"
>     <sip:069xxxxxxxx at 10.xx.xx.141>;tag=5evr6508K9S3K.
>     >>>     To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
>     >>>     Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
>     >>>     CSeq: 129803060 INVITE.
>     >>>     Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
>     >>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
>     >>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO,
>     >>>     REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
>     >>>     Supported: timer, precondition, path, replaces.
>     >>>     Allow-Events: talk, presence, dialog, line-seize,
>     call-info, sla,
>     >>>     include-session-description, presence.winfo,
>     message-summary, refer.
>     >>>     Content-Type: application/sdp.
>     >>>     Content-Disposition: session.
>     >>>     Content-Length: 920.
>     >>>     X-FS-Support: update_display.
>     >>>     Remote-Party-ID: "069xxxxxxxx"
>     >>>    
>     <sip:069xxxxxxxx at 10.xx.xx.141>;party=calling;screen=yes;privacy=off.
>     >>>     .
>     >>>     v=0.
>     >>>     o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
>     >>>     s=FreeSWITCH.
>     >>>     c=IN IP4 10.xx.xx.141.
>     >>>     t=0 0.
>     >>>     m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98
>     122 9
>     >>>     121 3
>     >>>     101 13.
>     >>>     a=rtpmap:115 G7221/32000.
>     >>>     a=fmtp:115 bitrate=48000.
>     >>>     a=rtpmap:96 AMR/8000.
>     >>>     a=fmtp:96 octet-align=0.
>     >>>     a=rtpmap:99 SPEEX/32000.
>     >>>     a=rtpmap:18 G729/8000.
>     >>>     a=rtpmap:4 G723/8000.
>     >>>     a=rtpmap:7 LPC/8000.
>     >>>     a=rtpmap:124 G726-16/8000.
>     >>>     a=rtpmap:8 PCMA/8000.
>     >>>     a=rtpmap:6 DVI4/16000.
>     >>>     a=rtpmap:123 G726-24/8000.
>     >>>     a=rtpmap:0 PCMU/8000.
>     >>>     a=rtpmap:10 L16/22050.
>     >>>     a=rtpmap:98 iLBC/8000.
>     >>>     a=fmtp:98 mode
>     >>>     #
>     >>>
>     >>>
>     >>>
>     >>>     _______________________________________________
>     >>>     FreeSWITCH-users mailing list
>     >>>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>>     <mailto: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>
>     >>     <mailto: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/
>     >> Twitter: http://twitter.com/FreeSWITCH_wire
>     >>
>     >> AIM: anthm
>     >> MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     >> <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     >> IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >>
>     >> FreeSWITCH Developer Conference
>     >> sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     >> <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     >> googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     >> pstn:+19193869900
>     >>
>     ------------------------------------------------------------------------
>     >>
>     >> _______________________________________________
>     >> 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
>     <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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From aep.lists at it46.se  Thu Apr 29 07:09:28 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Thu, 29 Apr 2010 16:09:28 +0200
Subject: [Freeswitch-users] event after hangup (Javascript)
Message-ID: <b7818a7faca042bfd028f33900ed0d0d.squirrel@correo.nodo50.org>

Hi,

Is there a way to trigger a customized event after the user hangup the call?

Although i define the
function on_hangup(hup_session, how)

it seems i can not trigger and event inside of it although i can run logging.

Hints are welcome.

-aep
-- 
Stopping junk mailers is good for the environment





From Prometheus001 at gmx.net  Thu Apr 29 07:18:22 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 29 Apr 2010 16:18:22 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>
Message-ID: <4BD9952E.604@gmx.net>

Thanks for the hint,

should "rtp-autofix-timing" be set to true or false in that case?

Also: What is the difference to "rtp auto adjust"?

I would like to update the wiki on that, when this is clearified. The
Fritzbox is the most common ATA here in Europe.

Best rgeards
Peter

Mathieu Rene schrieb:
> If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
>
> that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
>
> Perhaps didn't you rescan the sofia profile after your config change?
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
>
>   
>> Hello Nicola,
>>
>> we had the same problem with the FritzBoxes. Someweher in the internet
>> we found an advice to patch Freeswitch. See my previous comment on this
>> patch here:
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
>>
>> Best regards
>> Peter
>>
>>
>> Anthony Minessale schrieb:
>>     
>>> nothing it's default config on latest git HEAD
>>>
>>>
>>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
>>> <mailto:nico at vthadden.de>> wrote:
>>>
>>>    Changing those values didn't help
>>>
>>>    I wonder why then it works with all other servers.
>>>    Also, if I call the 9888 (so 888 at conference.freeswitch.org
>>>    <mailto:888 at conference.freeswitch.org>) it works
>>>    without problems, i wonder what they have changed in the config O.o
>>>
>>>    On 28.04.2010 23:04, Anthony Minessale wrote:
>>>       
>>>> Do you understand that the problem is with your device and not FS?
>>>> So why do you think something would have changed?
>>>>
>>>> here is what you can do to work around it:
>>>>
>>>> look for this param in you sip profile config:
>>>> <param name="inbound-codec-negotiation" value="generous"/>
>>>>
>>>> change it to the following:
>>>> <param name="inbound-codec-negotiation" value="scrooge"/>
>>>>
>>>> restart and retry
>>>>
>>>> if this is not enough also edit vars.xml
>>>>
>>>> look for
>>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>>>
>>>> replace it with:
>>>>
>>>> <X-PRE-PROCESS cmd="set"
>>>>         
>>>    data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>>>       
>>>> retry
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>>>>         
>>>    <nico at vthadden.de <mailto:nico at vthadden.de>
>>>       
>>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>>>>
>>>>    Hi,
>>>>
>>>>    is there anything new since august?
>>>>    I have the same problem with latest FritzBox firmenware and
>>>>         
>>>    latest
>>>       
>>>>    FreeSwitch installed (via Quick & Dirty install).
>>>>
>>>>    FritzBox always sends 30ms in one paket, FS starts with 20
>>>>         
>>>    and then
>>>       
>>>>    switches to 30 (maybe thats the problem?)
>>>>
>>>>    Calling the Demo IVR at 5000 is (with default options and
>>>>         
>>>    latest FS) no
>>>       
>>>>    problem, both uses 30ms ptime, have no idea why.
>>>>
>>>>    Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>>>>         
>>>    ptime,
>>>       
>>>>    FBox 30.
>>>>
>>>>    Conferences are also choppy.
>>>>
>>>>    Adding <param name="rtp-autofix-timing" value="false" />
>>>>    to 1000.xml (account used at the FBox) does not help.
>>>>
>>>>    Anyone got an idea?
>>>>
>>>>    Nico
>>>>
>>>>    _______________________________________________
>>>>    FreeSWITCH-users mailing list
>>>>    FreeSWITCH-users at lists.freeswitch.org
>>>>         
>>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>       
>>>>    <mailto: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/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>>         
>>>    <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>       
>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>>         
>>>    <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>>       
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>         
>>>    <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>       
>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>>         
>>>    <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>>       
>>>> IRC: irc.freenode.net <http://irc.freenode.net>
>>>>         
>>>    <http://irc.freenode.net> #freeswitch
>>>       
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>>         
>>>    <mailto:sip%3A888 at conference.freeswitch.org>
>>>       
>>>> <mailto:sip%3A888 at conference.freeswitch.org
>>>>         
>>>    <mailto:sip%253A888 at conference.freeswitch.org>>
>>>       
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>         
>>>    <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>       
>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>>         
>>>    <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>>       
>>>> pstn:+19193869900
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> 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>
>>> googletalk:conf+888 at conference.freeswitch.org
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From david.ponzone at gmail.com  Thu Apr 29 07:31:44 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 29 Apr 2010 16:31:44 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD9952E.604@gmx.net>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>
	<4BD9952E.604@gmx.net>
Message-ID: <177EA315-FD1B-4F38-96AC-9E1185C4D3E3@gmail.com>

Peter,

I think Math's advice was to set rtp-autofix-timing to false, because  
in some situations, the smartness in FS can cause issues.

rtp-auto-adjust has perhaps a misleading name.
It's sometimes called auto-NAT or NAT autolearn.
it's the ability for FreeSWITCH to adjust dynamically itself to the  
received RTP stream on a port, instead of what it was expecting from  
the SDP sent by the remote party.
it does that by learning the source IP/port of the 20 first packets.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/04/2010 ? 16:18, Peter P GMX a ?crit :

> Thanks for the hint,
>
> should "rtp-autofix-timing" be set to true or false in that case?
>
> Also: What is the difference to "rtp auto adjust"?
>
> I would like to update the wiki on that, when this is clearified. The
> Fritzbox is the most common ATA here in Europe.
>
> Best rgeards
> Peter
>
> Mathieu Rene schrieb:
>> If you look at the code a bit further, you'll notice an if  
>> (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
>>
>> that flag is set depending on the value of the "rtp-autofix-timing"  
>> parameter in the sip profile, no need to patch anything :)
>>
>> Perhaps didn't you rescan the sofia profile after your config change?
>>
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
>>
>>
>>> Hello Nicola,
>>>
>>> we had the same problem with the FritzBoxes. Someweher in the  
>>> internet
>>> we found an advice to patch Freeswitch. See my previous comment on  
>>> this
>>> patch here:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Anthony Minessale schrieb:
>>>
>>>> nothing it's default config on latest git HEAD
>>>>
>>>>
>>>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
>>>> <mailto:nico at vthadden.de>> wrote:
>>>>
>>>>   Changing those values didn't help
>>>>
>>>>   I wonder why then it works with all other servers.
>>>>   Also, if I call the 9888 (so 888 at conference.freeswitch.org
>>>>   <mailto:888 at conference.freeswitch.org>) it works
>>>>   without problems, i wonder what they have changed in the config  
>>>> O.o
>>>>
>>>>   On 28.04.2010 23:04, Anthony Minessale wrote:
>>>>
>>>>> Do you understand that the problem is with your device and not FS?
>>>>> So why do you think something would have changed?
>>>>>
>>>>> here is what you can do to work around it:
>>>>>
>>>>> look for this param in you sip profile config:
>>>>> <param name="inbound-codec-negotiation" value="generous"/>
>>>>>
>>>>> change it to the following:
>>>>> <param name="inbound-codec-negotiation" value="scrooge"/>
>>>>>
>>>>> restart and retry
>>>>>
>>>>> if this is not enough also edit vars.xml
>>>>>
>>>>> look for
>>>>> <X-PRE-PROCESS cmd="set"  
>>>>> data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>>>>
>>>>> replace it with:
>>>>>
>>>>> <X-PRE-PROCESS cmd="set"
>>>>>
>>>>   data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>>>>
>>>>> retry
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>>>>>
>>>>   <nico at vthadden.de <mailto:nico at vthadden.de>
>>>>
>>>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>>>>>
>>>>>   Hi,
>>>>>
>>>>>   is there anything new since august?
>>>>>   I have the same problem with latest FritzBox firmenware and
>>>>>
>>>>   latest
>>>>
>>>>>   FreeSwitch installed (via Quick & Dirty install).
>>>>>
>>>>>   FritzBox always sends 30ms in one paket, FS starts with 20
>>>>>
>>>>   and then
>>>>
>>>>>   switches to 30 (maybe thats the problem?)
>>>>>
>>>>>   Calling the Demo IVR at 5000 is (with default options and
>>>>>
>>>>   latest FS) no
>>>>
>>>>>   problem, both uses 30ms ptime, have no idea why.
>>>>>
>>>>>   Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>>>>>
>>>>   ptime,
>>>>
>>>>>   FBox 30.
>>>>>
>>>>>   Conferences are also choppy.
>>>>>
>>>>>   Adding <param name="rtp-autofix-timing" value="false" />
>>>>>   to 1000.xml (account used at the FBox) does not help.
>>>>>
>>>>>   Anyone got an idea?
>>>>>
>>>>>   Nico
>>>>>
>>>>>   _______________________________________________
>>>>>   FreeSWITCH-users mailing list
>>>>>   FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>
>>>>>   <mailto: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/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>>
>>>>   <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>>
>>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>>>
>>>>   <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>>>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>
>>>>   <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>>
>>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>>>
>>>>   <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>>>
>>>>> IRC: irc.freenode.net <http://irc.freenode.net>
>>>>>
>>>>   <http://irc.freenode.net> #freeswitch
>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>>
>>>>   <mailto:sip%3A888 at conference.freeswitch.org>
>>>>
>>>>> <mailto:sip%3A888 at conference.freeswitch.org
>>>>>
>>>>   <mailto:sip%253A888 at conference.freeswitch.org>>
>>>>
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>
>>>>   <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>
>>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>>>
>>>>   <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>>>
>>>>> pstn:+19193869900
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Thu Apr 29 07:53:06 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 29 Apr 2010 10:53:06 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD9952E.604@gmx.net>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>
	<4BD9952E.604@gmx.net>
Message-ID: <1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>

auto-adjust will automatically switch the IP the RTP is being sent to based on the received packet's IP header. autofix timing mesures the received payload size and compares it with how much "<ptime> ms " is supposed to take. If there is a difference it thins the device is broken (a lot of UAs lie about their ptime in the SDP) and adjusts the codec's ptime in order to interop.

To be the equivalent of your patch, it would have to be set to false.

Out of curiosity, what codec is being used by fritzbox when the problem occurs?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-29, at 10:18 AM, Peter P GMX wrote:

> Thanks for the hint,
> 
> should "rtp-autofix-timing" be set to true or false in that case?
> 
> Also: What is the difference to "rtp auto adjust"?
> 
> I would like to update the wiki on that, when this is clearified. The
> Fritzbox is the most common ATA here in Europe.
> 
> Best rgeards
> Peter
> 
> Mathieu Rene schrieb:
>> If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
>> 
>> that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
>> 
>> Perhaps didn't you rescan the sofia profile after your config change?
>> 
>> 
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>> 
>> 
>> 
>> 
>> On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
>> 
>> 
>>> Hello Nicola,
>>> 
>>> we had the same problem with the FritzBoxes. Someweher in the internet
>>> we found an advice to patch Freeswitch. See my previous comment on this
>>> patch here:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
>>> 
>>> Best regards
>>> Peter
>>> 
>>> 
>>> Anthony Minessale schrieb:
>>> 
>>>> nothing it's default config on latest git HEAD
>>>> 
>>>> 
>>>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
>>>> <mailto:nico at vthadden.de>> wrote:
>>>> 
>>>>   Changing those values didn't help
>>>> 
>>>>   I wonder why then it works with all other servers.
>>>>   Also, if I call the 9888 (so 888 at conference.freeswitch.org
>>>>   <mailto:888 at conference.freeswitch.org>) it works
>>>>   without problems, i wonder what they have changed in the config O.o
>>>> 
>>>>   On 28.04.2010 23:04, Anthony Minessale wrote:
>>>> 
>>>>> Do you understand that the problem is with your device and not FS?
>>>>> So why do you think something would have changed?
>>>>> 
>>>>> here is what you can do to work around it:
>>>>> 
>>>>> look for this param in you sip profile config:
>>>>> <param name="inbound-codec-negotiation" value="generous"/>
>>>>> 
>>>>> change it to the following:
>>>>> <param name="inbound-codec-negotiation" value="scrooge"/>
>>>>> 
>>>>> restart and retry
>>>>> 
>>>>> if this is not enough also edit vars.xml
>>>>> 
>>>>> look for
>>>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>>>> 
>>>>> replace it with:
>>>>> 
>>>>> <X-PRE-PROCESS cmd="set"
>>>>> 
>>>>   data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>>>> 
>>>>> retry
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>>>>> 
>>>>   <nico at vthadden.de <mailto:nico at vthadden.de>
>>>> 
>>>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>>>>> 
>>>>>   Hi,
>>>>> 
>>>>>   is there anything new since august?
>>>>>   I have the same problem with latest FritzBox firmenware and
>>>>> 
>>>>   latest
>>>> 
>>>>>   FreeSwitch installed (via Quick & Dirty install).
>>>>> 
>>>>>   FritzBox always sends 30ms in one paket, FS starts with 20
>>>>> 
>>>>   and then
>>>> 
>>>>>   switches to 30 (maybe thats the problem?)
>>>>> 
>>>>>   Calling the Demo IVR at 5000 is (with default options and
>>>>> 
>>>>   latest FS) no
>>>> 
>>>>>   problem, both uses 30ms ptime, have no idea why.
>>>>> 
>>>>>   Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>>>>> 
>>>>   ptime,
>>>> 
>>>>>   FBox 30.
>>>>> 
>>>>>   Conferences are also choppy.
>>>>> 
>>>>>   Adding <param name="rtp-autofix-timing" value="false" />
>>>>>   to 1000.xml (account used at the FBox) does not help.
>>>>> 
>>>>>   Anyone got an idea?
>>>>> 
>>>>>   Nico
>>>>> 
>>>>>   _______________________________________________
>>>>>   FreeSWITCH-users mailing list
>>>>>   FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> 
>>>>>   <mailto: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/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> 
>>>>   <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>> 
>>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>>> 
>>>>   <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>>> 
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> 
>>>>   <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>> 
>>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>>> 
>>>>   <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>>> 
>>>>> IRC: irc.freenode.net <http://irc.freenode.net>
>>>>> 
>>>>   <http://irc.freenode.net> #freeswitch
>>>> 
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> 
>>>>   <mailto:sip%3A888 at conference.freeswitch.org>
>>>> 
>>>>> <mailto:sip%3A888 at conference.freeswitch.org
>>>>> 
>>>>   <mailto:sip%253A888 at conference.freeswitch.org>>
>>>> 
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> 
>>>>   <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> 
>>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>>> 
>>>>   <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>>> 
>>>>> pstn:+19193869900
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 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/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> 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>
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>> ------------------------------------------------------------------------
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 29 08:01:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Apr 2010 10:01:06 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>
References: <4BD86880.4090607@vthadden.de>
	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
	<4BD8A5CC.8060107@vthadden.de>
	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca> <4BD9952E.604@gmx.net>
	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>
Message-ID: <t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>

Does anyone ever report this issue to Fritzbox or is too hard
to conceive that they have the bug in this situation?


On Thu, Apr 29, 2010 at 9:53 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> auto-adjust will automatically switch the IP the RTP is being sent to based
> on the received packet's IP header. autofix timing mesures the received
> payload size and compares it with how much "<ptime> ms " is supposed to
> take. If there is a difference it thins the device is broken (a lot of UAs
> lie about their ptime in the SDP) and adjusts the codec's ptime in order to
> interop.
>
> To be the equivalent of your patch, it would have to be set to false.
>
> Out of curiosity, what codec is being used by fritzbox when the problem
> occurs?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-29, at 10:18 AM, Peter P GMX wrote:
>
> > Thanks for the hint,
> >
> > should "rtp-autofix-timing" be set to true or false in that case?
> >
> > Also: What is the difference to "rtp auto adjust"?
> >
> > I would like to update the wiki on that, when this is clearified. The
> > Fritzbox is the most common ATA here in Europe.
> >
> > Best rgeards
> > Peter
> >
> > Mathieu Rene schrieb:
> >> If you look at the code a bit further, you'll notice an if
> (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
> >>
> >> that flag is set depending on the value of the "rtp-autofix-timing"
> parameter in the sip profile, no need to patch anything :)
> >>
> >> Perhaps didn't you rescan the sofia profile after your config change?
> >>
> >>
> >> Mathieu Rene
> >> Avant-Garde Solutions Inc
> >> Office: + 1 (514) 664-1044 x100
> >> Cell: +1 (514) 664-1044 x200
> >> mrene at avgs.ca
> >>
> >>
> >>
> >>
> >> On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
> >>
> >>
> >>> Hello Nicola,
> >>>
> >>> we had the same problem with the FritzBoxes. Someweher in the internet
> >>> we found an advice to patch Freeswitch. See my previous comment on this
> >>> patch here:
> >>>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
> >>>
> >>> Best regards
> >>> Peter
> >>>
> >>>
> >>> Anthony Minessale schrieb:
> >>>
> >>>> nothing it's default config on latest git HEAD
> >>>>
> >>>>
> >>>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico at vthadden.de
> >>>> <mailto:nico at vthadden.de>> wrote:
> >>>>
> >>>>   Changing those values didn't help
> >>>>
> >>>>   I wonder why then it works with all other servers.
> >>>>   Also, if I call the 9888 (so 888 at conference.freeswitch.org
> >>>>   <mailto:888 at conference.freeswitch.org>) it works
> >>>>   without problems, i wonder what they have changed in the config O.o
> >>>>
> >>>>   On 28.04.2010 23:04, Anthony Minessale wrote:
> >>>>
> >>>>> Do you understand that the problem is with your device and not FS?
> >>>>> So why do you think something would have changed?
> >>>>>
> >>>>> here is what you can do to work around it:
> >>>>>
> >>>>> look for this param in you sip profile config:
> >>>>> <param name="inbound-codec-negotiation" value="generous"/>
> >>>>>
> >>>>> change it to the following:
> >>>>> <param name="inbound-codec-negotiation" value="scrooge"/>
> >>>>>
> >>>>> restart and retry
> >>>>>
> >>>>> if this is not enough also edit vars.xml
> >>>>>
> >>>>> look for
> >>>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >>>>>
> >>>>> replace it with:
> >>>>>
> >>>>> <X-PRE-PROCESS cmd="set"
> >>>>>
> >>>>   data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
> >>>>
> >>>>> retry
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
> >>>>>
> >>>>   <nico at vthadden.de <mailto:nico at vthadden.de>
> >>>>
> >>>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
> >>>>>
> >>>>>   Hi,
> >>>>>
> >>>>>   is there anything new since august?
> >>>>>   I have the same problem with latest FritzBox firmenware and
> >>>>>
> >>>>   latest
> >>>>
> >>>>>   FreeSwitch installed (via Quick & Dirty install).
> >>>>>
> >>>>>   FritzBox always sends 30ms in one paket, FS starts with 20
> >>>>>
> >>>>   and then
> >>>>
> >>>>>   switches to 30 (maybe thats the problem?)
> >>>>>
> >>>>>   Calling the Demo IVR at 5000 is (with default options and
> >>>>>
> >>>>   latest FS) no
> >>>>
> >>>>>   problem, both uses 30ms ptime, have no idea why.
> >>>>>
> >>>>>   Calling the tetris-demo at 9998 is chopped up, FS using 20ms
> >>>>>
> >>>>   ptime,
> >>>>
> >>>>>   FBox 30.
> >>>>>
> >>>>>   Conferences are also choppy.
> >>>>>
> >>>>>   Adding <param name="rtp-autofix-timing" value="false" />
> >>>>>   to 1000.xml (account used at the FBox) does not help.
> >>>>>
> >>>>>   Anyone got an idea?
> >>>>>
> >>>>>   Nico
> >>>>>
> >>>>>   _______________________________________________
> >>>>>   FreeSWITCH-users mailing list
> >>>>>   FreeSWITCH-users at lists.freeswitch.org
> >>>>>
> >>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>>
> >>>>>   <mailto: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/
> >>>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>>
> >>>>> AIM: anthm
> >>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>>>
> >>>>   <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >>>>
> >>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >>>>>
> >>>>   <mailto: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>
> >>>>>
> >>>>   <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >>>>
> >>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >>>>>
> >>>>   <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >>>>
> >>>>> IRC: irc.freenode.net <http://irc.freenode.net>
> >>>>>
> >>>>   <http://irc.freenode.net> #freeswitch
> >>>>
> >>>>> FreeSWITCH Developer Conference
> >>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>>>
> >>>>   <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >>>>
> >>>>> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >>>>>
> >>>>   <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >>>>
> >>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>>>
> >>>>   <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >>>>
> >>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >>>>>
> >>>>   <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >>>>
> >>>>> pstn:+19193869900
> >>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> 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/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> <mailto: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>
> >>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.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>
> >>>> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >>>> pstn:+19193869900
> >>>>
> ------------------------------------------------------------------------
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From saeedahmad1981 at gmail.com  Thu Apr 29 08:34:06 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 29 Apr 2010 17:34:06 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>
References: <4BD86880.4090607@vthadden.de>
	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
	<4BD8A5CC.8060107@vthadden.de>
	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca> 
	<4BD9952E.604@gmx.net> <1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca> 
	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>
Message-ID: <z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>

i think usually not, because with FS sometimes you get the patch within
minutes, which you'll never get somewhere else :-)

On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Does anyone ever report this issue to Fritzbox or is too hard
> to conceive that they have the bug in this situation?
>
>
> On Thu, Apr 29, 2010 at 9:53 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> auto-adjust will automatically switch the IP the RTP is being sent to
>> based on the received packet's IP header. autofix timing mesures the
>> received payload size and compares it with how much "<ptime> ms " is
>> supposed to take. If there is a difference it thins the device is broken (a
>> lot of UAs lie about their ptime in the SDP) and adjusts the codec's ptime
>> in order to interop.
>>
>> To be the equivalent of your patch, it would have to be set to false.
>>
>> Out of curiosity, what codec is being used by fritzbox when the problem
>> occurs?
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 2010-04-29, at 10:18 AM, Peter P GMX wrote:
>>
>> > Thanks for the hint,
>> >
>> > should "rtp-autofix-timing" be set to true or false in that case?
>> >
>> > Also: What is the difference to "rtp auto adjust"?
>> >
>> > I would like to update the wiki on that, when this is clearified. The
>> > Fritzbox is the most common ATA here in Europe.
>> >
>> > Best rgeards
>> > Peter
>> >
>> > Mathieu Rene schrieb:
>> >> If you look at the code a bit further, you'll notice an if
>> (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
>> >>
>> >> that flag is set depending on the value of the "rtp-autofix-timing"
>> parameter in the sip profile, no need to patch anything :)
>> >>
>> >> Perhaps didn't you rescan the sofia profile after your config change?
>> >>
>> >>
>> >> Mathieu Rene
>> >> Avant-Garde Solutions Inc
>> >> Office: + 1 (514) 664-1044 x100
>> >> Cell: +1 (514) 664-1044 x200
>> >> mrene at avgs.ca
>> >>
>> >>
>> >>
>> >>
>> >> On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
>> >>
>> >>
>> >>> Hello Nicola,
>> >>>
>> >>> we had the same problem with the FritzBoxes. Someweher in the internet
>> >>> we found an advice to patch Freeswitch. See my previous comment on
>> this
>> >>> patch here:
>> >>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.html
>> >>>
>> >>> Best regards
>> >>> Peter
>> >>>
>> >>>
>> >>> Anthony Minessale schrieb:
>> >>>
>> >>>> nothing it's default config on latest git HEAD
>> >>>>
>> >>>>
>> >>>> On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <
>> nico at vthadden.de
>> >>>> <mailto:nico at vthadden.de>> wrote:
>> >>>>
>> >>>>   Changing those values didn't help
>> >>>>
>> >>>>   I wonder why then it works with all other servers.
>> >>>>   Also, if I call the 9888 (so 888 at conference.freeswitch.org
>> >>>>   <mailto:888 at conference.freeswitch.org>) it works
>> >>>>   without problems, i wonder what they have changed in the config O.o
>> >>>>
>> >>>>   On 28.04.2010 23:04, Anthony Minessale wrote:
>> >>>>
>> >>>>> Do you understand that the problem is with your device and not FS?
>> >>>>> So why do you think something would have changed?
>> >>>>>
>> >>>>> here is what you can do to work around it:
>> >>>>>
>> >>>>> look for this param in you sip profile config:
>> >>>>> <param name="inbound-codec-negotiation" value="generous"/>
>> >>>>>
>> >>>>> change it to the following:
>> >>>>> <param name="inbound-codec-negotiation" value="scrooge"/>
>> >>>>>
>> >>>>> restart and retry
>> >>>>>
>> >>>>> if this is not enough also edit vars.xml
>> >>>>>
>> >>>>> look for
>> >>>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> >>>>>
>> >>>>> replace it with:
>> >>>>>
>> >>>>> <X-PRE-PROCESS cmd="set"
>> >>>>>
>> >>>>   data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i"/>
>> >>>>
>> >>>>> retry
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>> On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
>> >>>>>
>> >>>>   <nico at vthadden.de <mailto:nico at vthadden.de>
>> >>>>
>> >>>>> <mailto:nico at vthadden.de <mailto:nico at vthadden.de>>> wrote:
>> >>>>>
>> >>>>>   Hi,
>> >>>>>
>> >>>>>   is there anything new since august?
>> >>>>>   I have the same problem with latest FritzBox firmenware and
>> >>>>>
>> >>>>   latest
>> >>>>
>> >>>>>   FreeSwitch installed (via Quick & Dirty install).
>> >>>>>
>> >>>>>   FritzBox always sends 30ms in one paket, FS starts with 20
>> >>>>>
>> >>>>   and then
>> >>>>
>> >>>>>   switches to 30 (maybe thats the problem?)
>> >>>>>
>> >>>>>   Calling the Demo IVR at 5000 is (with default options and
>> >>>>>
>> >>>>   latest FS) no
>> >>>>
>> >>>>>   problem, both uses 30ms ptime, have no idea why.
>> >>>>>
>> >>>>>   Calling the tetris-demo at 9998 is chopped up, FS using 20ms
>> >>>>>
>> >>>>   ptime,
>> >>>>
>> >>>>>   FBox 30.
>> >>>>>
>> >>>>>   Conferences are also choppy.
>> >>>>>
>> >>>>>   Adding <param name="rtp-autofix-timing" value="false" />
>> >>>>>   to 1000.xml (account used at the FBox) does not help.
>> >>>>>
>> >>>>>   Anyone got an idea?
>> >>>>>
>> >>>>>   Nico
>> >>>>>
>> >>>>>   _______________________________________________
>> >>>>>   FreeSWITCH-users mailing list
>> >>>>>   FreeSWITCH-users at lists.freeswitch.org
>> >>>>>
>> >>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >>>>
>> >>>>>   <mailto: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/
>> >>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>>
>> >>>>> AIM: anthm
>> >>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>>>
>> >>>>   <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >>>>
>> >>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >>>>>
>> >>>>   <mailto: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>
>> >>>>>
>> >>>>   <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >>>>
>> >>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >>>>>
>> >>>>   <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
>> >>
>> >>>>
>> >>>>> IRC: irc.freenode.net <http://irc.freenode.net>
>> >>>>>
>> >>>>   <http://irc.freenode.net> #freeswitch
>> >>>>
>> >>>>> FreeSWITCH Developer Conference
>> >>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>>>>
>> >>>>   <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> >>>>
>> >>>>> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >>>>>
>> >>>>   <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
>> >>
>> >>>>
>> >>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>>>>
>> >>>>   <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >>>>
>> >>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >>>>>
>> >>>>   <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
>> >>
>> >>>>
>> >>>>> pstn:+19193869900
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> 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/
>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>
>> >>>> AIM: anthm
>> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>> <mailto: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>
>> >>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.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>
>> >>>> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >>>> pstn:+19193869900
>> >>>>
>> ------------------------------------------------------------------------
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr 29 08:40:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Apr 2010 10:40:37 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
References: <4BD86880.4090607@vthadden.de>
	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>
	<4BD8A5CC.8060107@vthadden.de>
	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>
	<4BD98D57.7030100@gmx.net>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>
	<4BD9952E.604@gmx.net>
	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>
	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>
	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
Message-ID: <2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>

OK how many of these devices are we talking about that you have?

/b

On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:

> i think usually not, because with FS sometimes you get the patch within minutes, which you'll never get somewhere else :-)
> 
> On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation?
> 

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From peder at networkoblivion.com  Thu Apr 29 08:50:05 2010
From: peder at networkoblivion.com (Peder)
Date: Thu, 29 Apr 2010 10:50:05 -0500
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
References: <05d601cae718$36799710$a36cc530$@com>
	<l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
Message-ID: <07c301cae7b3$a327b090$e97711b0$@com>

Didn't like that at all.  Here is what I have in dialplan:

<action application="bridge"
data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>

 

 

Here is debug:

 

EXECUTE sofia/internal/1111 at 192.168.1.108
bridge([leg_timeout=4][lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/55
51212|[leg_timeout=6][lcr_carrier=tel,lcr_rate=0.03000]sofia/gateway/Tel/555
1212)

 

2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
channel type [lcr_carrier=tex|lcr_rate=0.01900]sofia

 

2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
outgoing channel of type [[lcr_carrier=tex|lcr_rate=0.01900]sofia] cause:
[CHAN_NOT_IMPLEMENTED]

 

2010-04-29 15:40:12.370181 [DEBUG] switch_ivr_originate.c:3299 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

 

2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
channel type [lcr_carrier=tel|lcr_rate=0.03000]sofia

 

2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
outgoing channel of type [[lcr_carrier=tel|lcr_rate=0.03000]sofia] cause:
[CHAN_NOT_IMPLEMENTED]

 

 

Since ${lcr_route_1} already contains variables in [], it appears that the
extra set in the beginning may be causing an issue.  Note that it is
chopping off the dialstring after sofia (should be sofia/gateway/Tex/5551212
and just shows as sofia) and the "leg_timeout" appears to disappear too.  If
I remove the leg timeouts, I do see it bridge "sofia/external/5551212".  Any
other ideas?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, April 28, 2010 7:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dial Timeout Issue

 

Is it failing because gateway1 is not responding at all? If so then I'd say
you'd need to use leg_timeout or leg_media_timeout on each leg as needed:

<action application="bridge"
data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>

Check these variables:
http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout

Test it out and let us know.
-MC

On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:

I'm having an issue where originate_timeout doesn't seem to be working as I
think it would.  I am using lcr to get the gateways.  I can bridge to either
gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
for failover and it worked fine.  I decided to change gateway1 to a fake IP
to test failover and it works, but it takes 30 seconds to timeout.  I then
found originate_timeout and set it to 4 seconds, but it doesn't timeout.  It
still takes 30 seconds.


Here is the snippet from my dialplan:

<action application="set" data="originate_timeout=4"/>
<action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>


Here is the debug that shows it is being set:

EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
EXECUTE sofia/internal/1111 at 192.168.1.108
bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)

Then I get this:
2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]


And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
I would guess.  Am I using the timeout wrong?  Or is there some other
setting I need 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

 

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From Prometheus001 at gmx.net  Thu Apr 29 09:05:59 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 29 Apr 2010 18:05:59 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>	<4BD9952E.604@gmx.net>	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
Message-ID: <4BD9AE67.2040600@gmx.net>

I have 3 of them, in Germany I think there are about 10 Mio. Plus
France, Austria,Switzerland etc. I would say some dozens of Millions.

Best regards
Peter

Brian West schrieb:
> OK how many of these devices are we talking about that you have?
>
> /b
>
> On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:
>
>> i think usually not, because with FS sometimes you get the patch
>> within minutes, which you'll never get somewhere else :-)
>>
>> On Thu, Apr 29, 2010 at 5:01 PM, Anthony
>> Minessale <anthony.minessale at gmail.com
>> <mailto:anthony.minessale at gmail.com>> wrote:
>>
>>     Does anyone ever report this issue to Fritzbox or is too hard
>>     to conceive that they have the bug in this situation?
>>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 29 09:13:10 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Apr 2010 11:13:10 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD9AE67.2040600@gmx.net>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>	<4BD9952E.604@gmx.net>	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
	<4BD9AE67.2040600@gmx.net>
Message-ID: <ADF49E76-C340-448A-B4F9-4B186E39660B@freeswitch.org>

But you have exactly three of them.  Its cheaper to go buy something thats not broken vs trying to work around this.

Just saying!

/b

On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:

> I have 3 of them, in Germany I think there are about 10 Mio. Plus
> France, Austria,Switzerland etc. I would say some dozens of Millions.
> 
> Best regards
> Peter



From mwhite at thesummit-grp.com  Thu Apr 29 07:50:51 2010
From: mwhite at thesummit-grp.com (Matt White)
Date: Thu, 29 Apr 2010 10:50:51 -0400
Subject: [Freeswitch-users] Build fails on Suse
Message-ID: <4BD96479020000F000017787@firewall.thesummit-grp.com>

I'm using the opensuse build service to create rpm's for Suse.    

                The build fails with the following error below.  I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions    

      
          I: Expression compares a char* pointer with a string literal.    
             Usually a strcmp() was intended by the programmer    
          E: freeswitch stringcompare strings/apr_snprintf.c:1261    

                Any thoughts?  I'm using the nightly snapshot.      

                -M    
  
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From mwhite at thesummit-grp.com  Thu Apr 29 08:13:21 2010
From: mwhite at thesummit-grp.com (Matt White)
Date: Thu, 29 Apr 2010 11:13:21 -0400
Subject: [Freeswitch-users] Build fails on Suse
Message-ID: <4BD969D1020000F000017790@firewall.thesummit-grp.com>


I'm using the opensuse build service to create rpm's for Suse. 

The build fails with the following error below.  I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions 


I: Expression compares a char* pointer with a string literal. 
   Usually a strcmp() was intended by the programmer 
E: freeswitch stringcompare strings/apr_snprintf.c:1261 

Any thoughts?  I'm using the nightly snapshot.   

-M 
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From stevendt at primrosebank.net  Thu Apr 29 11:15:58 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Thu, 29 Apr 2010 19:15:58 +0100
Subject: [Freeswitch-users] Picking up voicemail
Message-ID: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>

Hi,

I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good.

I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension. 


regards
Dave
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From rupa at rupa.com  Thu Apr 29 11:36:30 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 13:36:30 -0500
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <07c301cae7b3$a327b090$e97711b0$@com>
References: <05d601cae718$36799710$a36cc530$@com>
	<l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
	<07c301cae7b3$a327b090$e97711b0$@com>
Message-ID: <j2pb39273531004291136y8720a5aei1ae06f194af7fdc0@mail.gmail.com>

You won't be able to do it that way.  lcr is already using the [] and you
can't get two for each leg.  So, either use the same timeout for all legs
using {leg_timeout=5} or you can push me to get the new mod_lcr in which
lets you put arbitrary key/value pairs in their populated from your
custom_sql.  Unfortunately I'm travelling next week so I don't see doing
something sooner than that...

On Thu, Apr 29, 2010 at 10:50 AM, Peder <peder at networkoblivion.com> wrote:

>  Didn?t like that at all.  Here is what I have in dialplan:
>
> <action application="bridge"
> data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>
>
>
>
>
>
> Here is debug:
>
>
>
> EXECUTE sofia/internal/1111 at 192.168.1.108bridge([leg_timeout=4][lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[leg_timeout=6][lcr_carrier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
> channel type [lcr_carrier=tex|lcr_rate=0.01900]sofia
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
> outgoing channel of type [[lcr_carrier=tex|lcr_rate=0.01900]sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
>
>
> 2010-04-29 15:40:12.370181 [DEBUG] switch_ivr_originate.c:3299 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
> channel type [lcr_carrier=tel|lcr_rate=0.03000]sofia
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
> outgoing channel of type [[lcr_carrier=tel|lcr_rate=0.03000]sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
>
>
>
>
> Since ${lcr_route_1} already contains variables in [], it appears that the
> extra set in the beginning may be causing an issue.  Note that it is
> chopping off the dialstring after sofia (should be sofia/gateway/Tex/5551212
> and just shows as sofia) and the ?leg_timeout? appears to disappear too.  If
> I remove the leg timeouts, I do see it bridge ?sofia/external/5551212?.  Any
> other ideas?
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, April 28, 2010 7:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dial Timeout Issue
>
>
>
> Is it failing because gateway1 is not responding at all? If so then I'd say
> you'd need to use leg_timeout or leg_media_timeout on each leg as needed:
>
> <action application="bridge"
> data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>
>
> Check these variables:
> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout
>
> Test it out and let us know.
> -MC
>
> On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:
>
> I'm having an issue where originate_timeout doesn't seem to be working as I
> think it would.  I am using lcr to get the gateways.  I can bridge to
> either
> gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
> for failover and it worked fine.  I decided to change gateway1 to a fake IP
> to test failover and it works, but it takes 30 seconds to timeout.  I then
> found originate_timeout and set it to 4 seconds, but it doesn't timeout.
>  It
> still takes 30 seconds.
>
>
> Here is the snippet from my dialplan:
>
> <action application="set" data="originate_timeout=4"/>
> <action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>
>
>
> Here is the debug that shows it is being set:
>
> EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
> 2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
> sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
> EXECUTE sofia/internal/1111 at 192.168.1.108
>
> bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
> ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>
> Then I get this:
> 2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
> sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
>
>
> And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
> I would guess.  Am I using the timeout wrong?  Or is there some other
> setting I need 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Thu Apr 29 11:36:30 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 13:36:30 -0500
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <07c301cae7b3$a327b090$e97711b0$@com>
References: <05d601cae718$36799710$a36cc530$@com>
	<l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>
	<07c301cae7b3$a327b090$e97711b0$@com>
Message-ID: <j2pb39273531004291136y8720a5aei1ae06f194af7fdc0@mail.gmail.com>

You won't be able to do it that way.  lcr is already using the [] and you
can't get two for each leg.  So, either use the same timeout for all legs
using {leg_timeout=5} or you can push me to get the new mod_lcr in which
lets you put arbitrary key/value pairs in their populated from your
custom_sql.  Unfortunately I'm travelling next week so I don't see doing
something sooner than that...

On Thu, Apr 29, 2010 at 10:50 AM, Peder <peder at networkoblivion.com> wrote:

>  Didn?t like that at all.  Here is what I have in dialplan:
>
> <action application="bridge"
> data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>
>
>
>
>
>
> Here is debug:
>
>
>
> EXECUTE sofia/internal/1111 at 192.168.1.108bridge([leg_timeout=4][lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[leg_timeout=6][lcr_carrier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
> channel type [lcr_carrier=tex|lcr_rate=0.01900]sofia
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
> outgoing channel of type [[lcr_carrier=tex|lcr_rate=0.01900]sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
>
>
> 2010-04-29 15:40:12.370181 [DEBUG] switch_ivr_originate.c:3299 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
> channel type [lcr_carrier=tel|lcr_rate=0.03000]sofia
>
>
>
> 2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
> outgoing channel of type [[lcr_carrier=tel|lcr_rate=0.03000]sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
>
>
>
>
> Since ${lcr_route_1} already contains variables in [], it appears that the
> extra set in the beginning may be causing an issue.  Note that it is
> chopping off the dialstring after sofia (should be sofia/gateway/Tex/5551212
> and just shows as sofia) and the ?leg_timeout? appears to disappear too.  If
> I remove the leg timeouts, I do see it bridge ?sofia/external/5551212?.  Any
> other ideas?
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, April 28, 2010 7:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dial Timeout Issue
>
>
>
> Is it failing because gateway1 is not responding at all? If so then I'd say
> you'd need to use leg_timeout or leg_media_timeout on each leg as needed:
>
> <action application="bridge"
> data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>
>
> Check these variables:
> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout
>
> Test it out and let us know.
> -MC
>
> On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:
>
> I'm having an issue where originate_timeout doesn't seem to be working as I
> think it would.  I am using lcr to get the gateways.  I can bridge to
> either
> gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
> for failover and it worked fine.  I decided to change gateway1 to a fake IP
> to test failover and it works, but it takes 30 seconds to timeout.  I then
> found originate_timeout and set it to 4 seconds, but it doesn't timeout.
>  It
> still takes 30 seconds.
>
>
> Here is the snippet from my dialplan:
>
> <action application="set" data="originate_timeout=4"/>
> <action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>
>
>
> Here is the debug that shows it is being set:
>
> EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
> 2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
> sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
> EXECUTE sofia/internal/1111 at 192.168.1.108
>
> bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
> ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)
>
> Then I get this:
> 2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
> sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
>
>
> And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
> I would guess.  Am I using the timeout wrong?  Or is there some other
> setting I need 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From peder at networkoblivion.com  Thu Apr 29 11:53:11 2010
From: peder at networkoblivion.com (Peder)
Date: Thu, 29 Apr 2010 13:53:11 -0500
Subject: [Freeswitch-users] Dial Timeout Issue
In-Reply-To: <j2pb39273531004291136y8720a5aei1ae06f194af7fdc0@mail.gmail.com>
References: <05d601cae718$36799710$a36cc530$@com>	<l2p87f2f3b91004281700u103a2fd5ucf856348773be39e@mail.gmail.com>	<07c301cae7b3$a327b090$e97711b0$@com>
	<j2pb39273531004291136y8720a5aei1ae06f194af7fdc0@mail.gmail.com>
Message-ID: <09c301cae7cd$37144e30$a53cea90$@com>

That works, thanks.  Anybody have any idea why this doesn't accomplish the
same thing:

 

<action application="set" data="originate_timeout=4"/>

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Thursday, April 29, 2010 1:37 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Dial Timeout Issue

 

You won't be able to do it that way.  lcr is already using the [] and you
can't get two for each leg.  So, either use the same timeout for all legs
using {leg_timeout=5} or you can push me to get the new mod_lcr in which
lets you put arbitrary key/value pairs in their populated from your
custom_sql.  Unfortunately I'm travelling next week so I don't see doing
something sooner than that...

On Thu, Apr 29, 2010 at 10:50 AM, Peder <peder at networkoblivion.com> wrote:

Didn't like that at all.  Here is what I have in dialplan:

<action application="bridge"
data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>

 

 

Here is debug:

 

EXECUTE sofia/internal/1111 at 192.168.1.108
bridge([leg_timeout=4][lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/55
51212|[leg_timeout=6][lcr_carrier=tel,lcr_rate=0.03000]sofia/gateway/Tel/555
1212)

 

2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
channel type [lcr_carrier=tex|lcr_rate=0.01900]sofia

 

2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
outgoing channel of type [[lcr_carrier=tex|lcr_rate=0.01900]sofia] cause:
[CHAN_NOT_IMPLEMENTED]

 

2010-04-29 15:40:12.370181 [DEBUG] switch_ivr_originate.c:3299 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

 

2010-04-29 15:40:12.370181 [ERR] switch_core_session.c:360 Could not locate
channel type [lcr_carrier=tel|lcr_rate=0.03000]sofia

 

2010-04-29 15:40:12.370181 [ERR] switch_ivr_originate.c:2480 Cannot create
outgoing channel of type [[lcr_carrier=tel|lcr_rate=0.03000]sofia] cause:
[CHAN_NOT_IMPLEMENTED]

 

 

Since ${lcr_route_1} already contains variables in [], it appears that the
extra set in the beginning may be causing an issue.  Note that it is
chopping off the dialstring after sofia (should be sofia/gateway/Tex/5551212
and just shows as sofia) and the "leg_timeout" appears to disappear too.  If
I remove the leg timeouts, I do see it bridge "sofia/external/5551212".  Any
other ideas?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, April 28, 2010 7:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dial Timeout Issue

 

Is it failing because gateway1 is not responding at all? If so then I'd say
you'd need to use leg_timeout or leg_media_timeout on each leg as needed:

<action application="bridge"
data="[leg_timeout=4]${lcr_route_1}|[leg_timeout=6]${lcr_route_2}"/>

Check these variables:
http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout

Test it out and let us know.
-MC

On Wed, Apr 28, 2010 at 2:17 PM, Peder <peder at networkoblivion.com> wrote:

I'm having an issue where originate_timeout doesn't seem to be working as I
think it would.  I am using lcr to get the gateways.  I can bridge to either
gateway1 or 2 just fine.  Then I set it up with bridge using both gateways
for failover and it worked fine.  I decided to change gateway1 to a fake IP
to test failover and it works, but it takes 30 seconds to timeout.  I then
found originate_timeout and set it to 4 seconds, but it doesn't timeout.  It
still takes 30 seconds.


Here is the snippet from my dialplan:

<action application="set" data="originate_timeout=4"/>
<action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/>


Here is the debug that shows it is being set:

EXECUTE sofia/internal/1111 at 192.168.1.108 set(originate_timeout=4)
2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818
sofia/internal/1111 at 192.168.1.108 SET [originate_timeout]=[4]
EXECUTE sofia/internal/1111 at 192.168.1.108
bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr
ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212)

Then I get this:
2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup
sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]


And it successfully goes out gateway2, but it is 30 seconds later, not 4 as
I would guess.  Am I using the timeout wrong?  Or is there some other
setting I need 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

 


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




-- 
-Rupa

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From msc at freeswitch.org  Thu Apr 29 12:41:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Apr 2010 12:41:52 -0700
Subject: [Freeswitch-users] Picking up voicemail
In-Reply-To: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>
Message-ID: <i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com>

On Thu, Apr 29, 2010 at 11:15 AM, Dave Stevenson
<stevendt at primrosebank.net>wrote:

>  Hi,
>
> I have FreeSWITCH running over an internal IP network, connected to the
> PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of
> phones are rung until the first one picks up or until the caller falls over
> to voice mail and can leave a message in the Group Mailbox (100). I can
> retrieve the messages from any local phone - so far, so good.
>
> I'd like to be able to call in through the same PSTN line and interrogate
> the group mailbox, or any mailbox come to that. Can someone point me in the
> right direction for how to set this up please ? i.e., once FreeSwitch has
> answered the call and the voicemail prompts have stated, can they be
> interrupted so that I can get to the voice mail extension.
>

I think by default you can dial 0 while listening to the voicemail outgoing
message to log into the voicemail, however I don't believe there's a default
way of turning the vm login into a gateway to the rest of the system.
Anyone?
-MC
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From pjintheusa at gmail.com  Thu Apr 29 12:58:22 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 29 Apr 2010 15:58:22 -0400
Subject: [Freeswitch-users] user_exists / user_registered?
Message-ID: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>

Hi there,

You can call the api command user_exists - to see it the user is defined in
the /directory.

Is there an api command that tells you whether a user is currently
registered? Like user_registered??



Thanks

Pj
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From pjintheusa at gmail.com  Thu Apr 29 12:58:39 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 29 Apr 2010 15:58:39 -0400
Subject: [Freeswitch-users] FS Cluster
In-Reply-To: <z2j191c3a031004281445z82ed6375vb7b8b2eb9483c1c3@mail.gmail.com>
References: <z2y367751821004281437o41339cc2wa578bb12786d40d@mail.gmail.com>
	<z2j191c3a031004281445z82ed6375vb7b8b2eb9483c1c3@mail.gmail.com>
Message-ID: <z2i367751821004291258gf787274bl1824894d351219e9@mail.gmail.com>

Perfect - thanks very much.


On Wed, Apr 28, 2010 at 5:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> choose a neutral domain name and set it on both boxes in vars.xml
>
>
> On Wed, Apr 28, 2010 at 4:37 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> I am sure this has been asked before but I can not find any reference to
>> the subject in the wiki.
>>
>> Basically - I have two FreeSWITCH servers, FS1 and FS2. Both sit behind
>> OpenSIPS for load balancing
>>
>> Phone A is registered on FS1. A call comes in on FS2 for phone A.
>>
>> How do I get that call across to FS1.
>>
>> So far I have the internal profile of FS1 and FS2 pointing to the same
>> ODBC database.
>>
>> What else do I have to do?
>>
>> If anyone can point me in the right direction I would be grateful.
>>
>> Thanks
>>
>> Pj
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 29 13:09:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Apr 2010 15:09:56 -0500
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
Message-ID: <u2l191c3a031004291309gb7a03723p9360c058159939c1@mail.gmail.com>

sofia_contact <user>@<domain>[/profile]

On Thu, Apr 29, 2010 at 2:58 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> You can call the api command user_exists - to see it the user is defined in
> the /directory.
>
> Is there an api command that tells you whether a user is currently
> registered? Like user_registered??
>
>
>
> Thanks
>
> Pj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rupa at rupa.com  Thu Apr 29 13:12:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 15:12:32 -0500
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
Message-ID: <y2wb39273531004291312v3ea47091x5aaccc7ffd2815ec@mail.gmail.com>

Don't know of an endpoint independent one, but if all you care about is sip:

sofia_contact

is what you want to use.

On Thu, Apr 29, 2010 at 2:58 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> You can call the api command user_exists - to see it the user is defined in
> the /directory.
>
> Is there an api command that tells you whether a user is currently
> registered? Like user_registered??
>
>
>
> Thanks
>
> Pj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From david.ponzone at gmail.com  Thu Apr 29 13:13:37 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 29 Apr 2010 22:13:37 +0200
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
Message-ID: <AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>

Phillip,

perhaos not the nicest way, but you could use sofia_contact.

sofia_contact <profile>/<user id>@<domain>

If it returns:
error/user_not_registered
then you know it is not registered.

if it is, the answer would be a string starting with:
sofia/<profile>/etc...

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 29/04/2010 ? 21:58, Phillip Jones a ?crit :

> Hi there,
>
> You can call the api command user_exists - to see it the user is  
> defined in the /directory.
>
> Is there an api command that tells you whether a user is currently  
> registered? Like user_registered??
>
>
>
> Thanks
>
> Pj
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Thu Apr 29 13:41:39 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 29 Apr 2010 16:41:39 -0400
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
	<AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>
Message-ID: <m2o367751821004291341x2682f217x17c71580407563cf@mail.gmail.com>

Thank you all!

On Thu, Apr 29, 2010 at 4:13 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Phillip,
>
> perhaos not the nicest way, but you could use sofia_contact.
>
> sofia_contact <profile>/<user id>@<domain>
>
> If it returns:
> error/user_not_registered
> then you know it is not registered.
>
> if it is, the answer would be a string starting with:
> sofia/<profile>/etc...
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 29/04/2010 ? 21:58, Phillip Jones a ?crit :
>
> Hi there,
>
> You can call the api command user_exists - to see it the user is defined in
> the /directory.
>
> Is there an api command that tells you whether a user is currently
> registered? Like user_registered??
>
>
>
> Thanks
>
> Pj
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Thu Apr 29 13:56:24 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 29 Apr 2010 16:56:24 -0400
Subject: [Freeswitch-users] mobile phone clients / fs cluster
Message-ID: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>

Hi there,

I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy.
Outbound (terminating) traffic goes directly from each FS box.

All my home office phones get calls no matter which box they are registered
with or which box the call comes in on - thanks Anthony.

However my SIP client on various iPhone/Androids etc only receive calls that
originate on the box on which they are registered.

Looking at the SIP trace - when the call comes in on the 'wrong' box, the
invites to these SIP clients do not even get a response. Presumably because
the 3G network has no idea who this new IP in the "from address" is, who
trying to contact them.

Question is, is there a way around this - our will I have start routing all
terminating traffic out through the proxy also.

Any insight appreciated.

Thanks!

Pj
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From jerry.richards at teotech.com  Thu Apr 29 15:31:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 29 Apr 2010 15:31:06 -0700
Subject: [Freeswitch-users] No PRI Call Until After Manual "load mod_openzap"
Message-ID: <5D4D561C47D74E1988B32D78CD6349B7@greyhawk.tonecommander.com>


I upgraded to FS snapshot as of 04-27-2010 and Wanpipe 3.5.11 and now, for
some reason, I cannot make an outbound call unless I manual execute "load
mod_openzap".  Anyone know why?  I posted the initialization, failed call
attempt, load mod_openzap, and successful call at
http://pastebin.freeswitch.org/12845.

Thanks And Best Regards,
Jerry



From brian at freeswitch.org  Thu Apr 29 15:46:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Apr 2010 17:46:06 -0500
Subject: [Freeswitch-users] No PRI Call Until After Manual "load
	mod_openzap"
In-Reply-To: <5D4D561C47D74E1988B32D78CD6349B7@greyhawk.tonecommander.com>
References: <5D4D561C47D74E1988B32D78CD6349B7@greyhawk.tonecommander.com>
Message-ID: <57A82550-5541-4DE5-A12C-D906A30A60C7@freeswitch.org>

I'm going to guess its not in modules.conf.xml to load on start?

/b

On Apr 29, 2010, at 5:31 PM, Jerry Richards wrote:

> 
> I upgraded to FS snapshot as of 04-27-2010 and Wanpipe 3.5.11 and now, for
> some reason, I cannot make an outbound call unless I manual execute "load
> mod_openzap".  Anyone know why?  I posted the initialization, failed call
> attempt, load mod_openzap, and successful call at
> http://pastebin.freeswitch.org/12845.

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From vfclists at googlemail.com  Thu Apr 29 18:20:52 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 02:20:52 +0100
Subject: [Freeswitch-users] How does Freeswitch CDR determine which gateway
	was used in failover?
Message-ID: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>

How does Freeswitch CDR determine which gateway was used in failover?

I am looking at the xml cdr and cdr_csv and they don't appear to show
which gateway was used in the bridge data.

Is there a separate uuid for each bridging attempt, that can be
obtained from the  CDR,  or the event socket as last resort?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From vfclists at googlemail.com  Thu Apr 29 18:26:05 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 02:26:05 +0100
Subject: [Freeswitch-users] Difference between effective-caller-id and
	outbound-caller-id
In-Reply-To: <u2u191c3a031004280706p1f6f1a7cp81ece637ece6a9d@mail.gmail.com>
References: <o2m7abab2411004280515wad8b1732t185e7b93ac2e37cd@mail.gmail.com>
	<u2u191c3a031004280706p1f6f1a7cp81ece637ece6a9d@mail.gmail.com>
Message-ID: <g2w7abab2411004291826tc02991a7j8a349e6e690a2565@mail.gmail.com>

I sort of worked it out but wanted to be sure. Effective_caller_id can
be useful if you want to be called back on your direct line.

On 28 April 2010 15:06, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> neither one mean anything?
> do you mean effective_caller_id_name effective_caller_id_number vs
> origination_caller_id_name origination_caller_id_number
> if so
> effective_caller_id_name/number are variables you set on an inbound channel
> so when that inbound channel calls the bridge app to connection to an
> outbound channel, it will pass the values in that variable instead of the
> actual caller_id fields that were passed by the caller.
> origination_caller_id_name/number are variables you set in a dial-string to
> control the caller-id
> {origination_caller_id_name=test,origination_caller_id_number=51212}sofia/internal/foo at bar.com
> This is?independent?of the inbound leg and is also usable when there is no
> inbound leg such as with the originate cli command.
>
>
> On Wed, Apr 28, 2010 at 7:15 AM, Frank Church <vfclists at googlemail.com>
> wrote:
>>
>> What are the differences between the two and when is one more
>> appropriate than the other?
>>
>> What are the use cases?
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From vfclists at googlemail.com  Thu Apr 29 18:56:21 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 02:56:21 +0100
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
Message-ID: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>

I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
are not working. The system is a windows system and I wonder if the
defaults for windows are different.

I have logs/xml_cdr  in addition to the  log/xml_cdr in the
c:\freeswitch directory but Freeswitch can't find them.

Logs snippet
=========
2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/1811 at 192.168.1.133) State REPORTING
2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
posting to web server [http://192.168.1.20:8132/]
2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
url [http://192.168.1.20:8132/]
2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
web server, writing to file
2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
or directory]


-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From rupa at rupa.com  Thu Apr 29 20:09:57 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 29 Apr 2010 22:09:57 -0500
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
Message-ID: <y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>

set a var for each leg, this is how I do it in mod_lcr:

[gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1

On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com>wrote:

> How does Freeswitch CDR determine which gateway was used in failover?
>
> I am looking at the xml cdr and cdr_csv and they don't appear to show
> which gateway was used in the bridge data.
>
> Is there a separate uuid for each bridging attempt, that can be
> obtained from the  CDR,  or the event socket as last resort?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>



-- 
-Rupa
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From talk2ram at gmail.com  Thu Apr 29 22:11:28 2010
From: talk2ram at gmail.com (ram)
Date: Fri, 30 Apr 2010 10:41:28 +0530
Subject: [Freeswitch-users] mobile phone clients / fs cluster
In-Reply-To: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>
References: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>
Message-ID: <p2sb74751491004292211i4d66c1d2me49ffcfc18852692@mail.gmail.com>

Hi

why not its possible

Ram

On Fri, Apr 30, 2010 at 2:26 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy.
> Outbound (terminating) traffic goes directly from each FS box.
>
> All my home office phones get calls no matter which box they are registered
> with or which box the call comes in on - thanks Anthony.
>
> However my SIP client on various iPhone/Androids etc only receive calls
> that originate on the box on which they are registered.
>
> Looking at the SIP trace - when the call comes in on the 'wrong' box, the
> invites to these SIP clients do not even get a response. Presumably because
> the 3G network has no idea who this new IP in the "from address" is, who
> trying to contact them.
>
> Question is, is there a way around this - our will I have start routing all
> terminating traffic out through the proxy also.
>
> Any insight appreciated.
>
> Thanks!
>
> Pj
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vfclists at googlemail.com  Fri Apr 30 00:06:55 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 08:06:55 +0100
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
Message-ID: <u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>

I am not sure if you misunderstood my question.

The XML provided by mod_xml_cdr does not tell which gateway succeeded. When
I use a single gateway without failover I know the gateway specified in
lastapp was used. When I use failover the last_app string contains all the
gateways. I am looking for a value which tells which gateway was used.

<include>
>   <context name="external">
>       <extension name="external_outbound">
>         <condition field="destination_number" expression="(.*)">
>             <!-- <action application="set" data="proxy_media=true"/> -->
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="set" data="originate_timeout=2"/>
>             <action application="set" data="originate_retries=1"/>
>             <action application="set" data="progress_timeout=15"/>
>             <action application="set" data="call_timeout=35"/>
>             <action application="set" data="proxy_media=true"/>
>             <action application="bridge"
> data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1"
> />
>         </condition>
>       </extension>
>   </context>
> </include>
>



On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com> wrote:
> set a var for each leg, this is how I do it in mod_lcr:
> [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
>
> On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com>
> wrote:
>>
>> How does Freeswitch CDR determine which gateway was used in failover?
>>
>> I am looking at the xml cdr and cdr_csv and they don't appear to show
>> which gateway was used in the bridge data.
>>
>> Is there a separate uuid for each bridging attempt, that can be
>> obtained from the  CDR,  or the event socket as last resort?
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From david.ponzone at gmail.com  Fri Apr 30 00:32:31 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 30 Apr 2010 09:32:31 +0200
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
Message-ID: <A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>

Frank,

first of all, be sure you enabled writing leg B to CDR. If you don't,  
you won't see a gateway in there anytime soon.

In CSV CDR, you can change the template used by adding the field $ 
{sip_gateway_name}.
In XML CDR, you get this variable automatically, of course only in the  
leg B file.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/04/2010 ? 09:06, Frank Church a ?crit :

> I am not sure if you misunderstood my question.
>
> The XML provided by mod_xml_cdr does not tell which gateway  
> succeeded. When I use a single gateway without failover I know the  
> gateway specified in lastapp was used. When I use failover the  
> last_app string contains all the gateways. I am looking for a value  
> which tells which gateway was used.
>
> <include>
>   <context name="external">
>       <extension name="external_outbound">
>         <condition field="destination_number" expression="(.*)">
>             <!-- <action application="set" data="proxy_media=true"/>  
> -->
>             <action application="set"  
> data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="set" data="originate_timeout=2"/>
>             <action application="set" data="originate_retries=1"/>
>             <action application="set" data="progress_timeout=15"/>
>             <action application="set" data="call_timeout=35"/>
>             <action application="set" data="proxy_media=true"/>
>             <action application="bridge" data="sofia/gateway/ 
> provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1" />
>         </condition>
>       </extension>
>   </context>
> </include>
>
>
>
> On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com> wrote:
> > set a var for each leg, this is how I do it in mod_lcr:
> > [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
> >
> > On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com 
> >
> > wrote:
> >>
> >> How does Freeswitch CDR determine which gateway was used in  
> failover?
> >>
> >> I am looking at the xml cdr and cdr_csv and they don't appear to  
> show
> >> which gateway was used in the bridge data.
> >>
> >> Is there a separate uuid for each bridging attempt, that can be
> >> obtained from the  CDR,  or the event socket as last resort?
> >>
> >> --
> >> Frank Church
> >>
> >> =======================
> >> http://devblog.brahmancreations.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
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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  Fri Apr 30 00:55:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Apr 2010 00:55:05 -0700
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
	<A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
Message-ID: <q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>

It's late and I can't seem to locate it right now but Mathieu Rene mentioned
that there was a way to embed the b-leg cdr into the a leg cdr. Math, am I
losing it? :)

-MC

On Fri, Apr 30, 2010 at 12:32 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Frank,
>
> first of all, be sure you enabled writing leg B to CDR. If you don't, you
> won't see a gateway in there anytime soon.
>
> In CSV CDR, you can change the template used by adding the
> field ${sip_gateway_name}.
> In XML CDR, you get this variable automatically, of course only in the leg
> B file.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 30/04/2010 ? 09:06, Frank Church a ?crit :
>
> I am not sure if you misunderstood my question.
>
> The XML provided by mod_xml_cdr does not tell which gateway succeeded. When
> I use a single gateway without failover I know the gateway specified in
> lastapp was used. When I use failover the last_app string contains all the
> gateways. I am looking for a value which tells which gateway was used.
>
> <include>
>>   <context name="external">
>>       <extension name="external_outbound">
>>         <condition field="destination_number" expression="(.*)">
>>             <!-- <action application="set" data="proxy_media=true"/> -->
>>             <action application="set" data="hangup_after_bridge=true"/>
>>             <action application="set" data="continue_on_fail=true"/>
>>             <action application="set" data="originate_timeout=2"/>
>>             <action application="set" data="originate_retries=1"/>
>>             <action application="set" data="progress_timeout=15"/>
>>             <action application="set" data="call_timeout=35"/>
>>             <action application="set" data="proxy_media=true"/>
>>             <action application="bridge"
>> data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1"
>> />
>>         </condition>
>>       </extension>
>>   </context>
>> </include>
>>
>
>
>
> On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com> wrote:
> > set a var for each leg, this is how I do it in mod_lcr:
> > [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
> >
> > On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com>
> > wrote:
> >>
> >> How does Freeswitch CDR determine which gateway was used in failover?
> >>
> >> I am looking at the xml cdr and cdr_csv and they don't appear to show
> >> which gateway was used in the bridge data.
> >>
> >> Is there a separate uuid for each bridging attempt, that can be
> >> obtained from the  CDR,  or the event socket as last resort?
> >>
> >> --
> >> Frank Church
> >>
> >> =======================
> >> http://devblog.brahmancreations.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
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>
>
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From red.rain.seven at gmail.com  Fri Apr 30 01:26:14 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Fri, 30 Apr 2010 01:26:14 -0700
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <m2o367751821004291341x2682f217x17c71580407563cf@mail.gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
	<AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>
	<m2o367751821004291341x2682f217x17c71580407563cf@mail.gmail.com>
Message-ID: <t2o59ad9ca11004300126l97ebfa00i283948e6f642a57a@mail.gmail.com>

Does it also tells you if a user is busy?
If not, what should I do to find out if a user is on phone already so I can
send the second call to some place else?

Thanks,

Henry

On Thu, Apr 29, 2010 at 1:41 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Thank you all!
>
>
> On Thu, Apr 29, 2010 at 4:13 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Phillip,
>>
>> perhaos not the nicest way, but you could use sofia_contact.
>>
>> sofia_contact <profile>/<user id>@<domain>
>>
>> If it returns:
>> error/user_not_registered
>> then you know it is not registered.
>>
>> if it is, the answer would be a string starting with:
>> sofia/<profile>/etc...
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 29/04/2010 ? 21:58, Phillip Jones a ?crit :
>>
>> Hi there,
>>
>> You can call the api command user_exists - to see it the user is defined
>> in the /directory.
>>
>> Is there an api command that tells you whether a user is currently
>> registered? Like user_registered??
>>
>>
>>
>> Thanks
>>
>> Pj
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From david.ponzone at gmail.com  Fri Apr 30 01:36:00 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 30 Apr 2010 10:36:00 +0200
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <t2o59ad9ca11004300126l97ebfa00i283948e6f642a57a@mail.gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
	<AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>
	<m2o367751821004291341x2682f217x17c71580407563cf@mail.gmail.com>
	<t2o59ad9ca11004300126l97ebfa00i283948e6f642a57a@mail.gmail.com>
Message-ID: <8BBDD745-D92C-48F7-A5C2-3ACF0BEFA390@gmail.com>

A way, quite tricky, is mod_limit.

Another way, afaik, is to query the internal DB.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 30/04/2010 ? 10:26, Henry Huang a ?crit :

> Does it also tells you if a user is busy?
> If not, what should I do to find out if a user is on phone already  
> so I can send the second call to some place else?
>
> Thanks,
>
> Henry
>
> On Thu, Apr 29, 2010 at 1:41 PM, Phillip Jones  
> <pjintheusa at gmail.com> wrote:
> Thank you all!
>
>
> On Thu, Apr 29, 2010 at 4:13 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Phillip,
>
> perhaos not the nicest way, but you could use sofia_contact.
>
> sofia_contact <profile>/<user id>@<domain>
>
> If it returns:
> error/user_not_registered
> then you know it is not registered.
>
> if it is, the answer would be a string starting with:
> sofia/<profile>/etc...
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 29/04/2010 ? 21:58, Phillip Jones a ?crit :
>
>> Hi there,
>>
>> You can call the api command user_exists - to see it the user is  
>> defined in the /directory.
>>
>> Is there an api command that tells you whether a user is currently  
>> registered? Like user_registered??
>>
>>
>>
>> Thanks
>>
>> Pj
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mgg at giagnocavo.net  Fri Apr 30 02:09:59 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 30 Apr 2010 05:09:59 -0400
Subject: [Freeswitch-users] How does Freeswitch CDR determine
	which	gateway was used in failover?
In-Reply-To: <q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
	<A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
	<q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67035A6B0E94@mse17be1.mse17.exchange.ms>

There is/was a feature that did this: copy_xml_cdr, which sticks the b-leg CDR into the a-leg XML as a channel var. In practice I've found it sometimes just doesn't work, so don't use it if it's still there. There's also failed_xml_cdr_prefix which will write the failed b-legs into the main CDR. But I'm seeing a bug there too, where sometimes the b-leg CDR XML gets truncated. I have to do more research and I'll open a jira on it.

If you don't need the full b-leg CDR, it'll make your life a lot easier to just track the few pieces of info you need right on the a-leg by setting variables before bridging.

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, April 30, 2010 1:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How does Freeswitch CDR determine which gateway was used in failover?

It's late and I can't seem to locate it right now but Mathieu Rene mentioned that there was a way to embed the b-leg cdr into the a leg cdr. Math, am I losing it? :)

-MC
On Fri, Apr 30, 2010 at 12:32 AM, David Ponzone <david.ponzone at gmail.com<mailto:david.ponzone at gmail.com>> wrote:
Frank,

first of all, be sure you enabled writing leg B to CDR. If you don't, you won't see a gateway in there anytime soon.

In CSV CDR, you can change the template used by adding the field ${sip_gateway_name}.
In XML CDR, you get this variable automatically, of course only in the leg B file.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<http://www.ipeva.fr>  -   www.ipeva-studio.com<http://www.ipeva-studio.com>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 30/04/2010 ? 09:06, Frank Church a ?crit :


I am not sure if you misunderstood my question.

The XML provided by mod_xml_cdr does not tell which gateway succeeded. When I use a single gateway without failover I know the gateway specified in lastapp was used. When I use failover the last_app string contains all the gateways. I am looking for a value which tells which gateway was used.
<include>
  <context name="external">
      <extension name="external_outbound">
        <condition field="destination_number" expression="(.*)">
            <!-- <action application="set" data="proxy_media=true"/> -->
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="originate_timeout=2"/>
            <action application="set" data="originate_retries=1"/>
            <action application="set" data="progress_timeout=15"/>
            <action application="set" data="call_timeout=35"/>
            <action application="set" data="proxy_media=true"/>
            <action application="bridge" data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1" />
        </condition>
      </extension>
  </context>
</include>



On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
> set a var for each leg, this is how I do it in mod_lcr:
> [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
>
> On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com<mailto:vfclists at googlemail.com>>
> wrote:
>>
>> How does Freeswitch CDR determine which gateway was used in failover?
>>
>> I am looking at the xml cdr and cdr_csv and they don't appear to show
>> which gateway was used in the bridge data.
>>
>> Is there a separate uuid for each bridging attempt, that can be
>> obtained from the  CDR,  or the event socket as last resort?
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>> _______________________________________________
>> 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
>
>
>
> --
> -Rupa
>
> _______________________________________________
> 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
>
>



--
Frank Church

=======================
http://devblog.brahmancreations.com

_______________________________________________
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
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From vfclists at googlemail.com  Fri Apr 30 02:30:51 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 10:30:51 +0100
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
	<A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
	<q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
Message-ID: <m2m7abab2411004300230u825835echef0552b96b976039@mail.gmail.com>

Could that by any chance be referring to this thread -
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg09429.html?

On 30 April 2010 08:55, Michael Collins <msc at freeswitch.org> wrote:

> It's late and I can't seem to locate it right now but Mathieu Rene
> mentioned that there was a way to embed the b-leg cdr into the a leg cdr.
> Math, am I losing it? :)
>
> -MC
>
>
> On Fri, Apr 30, 2010 at 12:32 AM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Frank,
>>
>> first of all, be sure you enabled writing leg B to CDR. If you don't, you
>> won't see a gateway in there anytime soon.
>>
>> In CSV CDR, you can change the template used by adding the
>> field ${sip_gateway_name}.
>> In XML CDR, you get this variable automatically, of course only in the leg
>> B file.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 30/04/2010 ? 09:06, Frank Church a ?crit :
>>
>> I am not sure if you misunderstood my question.
>>
>> The XML provided by mod_xml_cdr does not tell which gateway succeeded.
>> When I use a single gateway without failover I know the gateway specified in
>> lastapp was used. When I use failover the last_app string contains all the
>> gateways. I am looking for a value which tells which gateway was used.
>>
>> <include>
>>>   <context name="external">
>>>       <extension name="external_outbound">
>>>         <condition field="destination_number" expression="(.*)">
>>>             <!-- <action application="set" data="proxy_media=true"/> -->
>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>             <action application="set" data="continue_on_fail=true"/>
>>>             <action application="set" data="originate_timeout=2"/>
>>>             <action application="set" data="originate_retries=1"/>
>>>             <action application="set" data="progress_timeout=15"/>
>>>             <action application="set" data="call_timeout=35"/>
>>>             <action application="set" data="proxy_media=true"/>
>>>             <action application="bridge"
>>> data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1"
>>> />
>>>         </condition>
>>>       </extension>
>>>   </context>
>>> </include>
>>>
>>
>>
>>
>> On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com> wrote:
>> > set a var for each leg, this is how I do it in mod_lcr:
>> > [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
>> >
>> > On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com>
>> > wrote:
>> >>
>> >> How does Freeswitch CDR determine which gateway was used in failover?
>> >>
>> >> I am looking at the xml cdr and cdr_csv and they don't appear to show
>> >> which gateway was used in the bridge data.
>> >>
>> >> Is there a separate uuid for each bridging attempt, that can be
>> >> obtained from the  CDR,  or the event socket as last resort?
>> >>
>> >> --
>> >> Frank Church
>> >>
>> >> =======================
>> >> http://devblog.brahmancreations.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
>> >
>> >
>> >
>> > --
>> > -Rupa
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From codecomplete at free.fr  Fri Apr 30 03:27:26 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 30 Apr 2010 03:27:26 -0700 (PDT)
Subject: [Freeswitch-users] Good book on (VoIP) telephony?
In-Reply-To: <1272412604620-4971798.post@n2.nabble.com>
References: <1272412604620-4971798.post@n2.nabble.com>
Message-ID: <1272623246083-4984890.post@n2.nabble.com>


Thanks everyone for the links.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Good-book-on-VoIP-telephony-tp4971798p4984890.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Prometheus001 at gmx.net  Fri Apr 30 04:23:21 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 30 Apr 2010 13:23:21 +0200
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
Message-ID: <4BDABDA9.5020707@gmx.net>

I just updated, with the same result:


Anthony Minessale schrieb:
> do you have lastest git HEAD ?
> can you update and try again?
>
>

Here's the log:

2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP:
v=0
o=root 929923105 929923106 IN IP4 192.168.178.125
s=call
c=IN IP4 192.168.178.125
t=0 0
m=audio 60566 RTP/AVP 18 8 0 99 3 101
a=rtpmap:18 g729/8000
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel
sofia/internal/sip:211 at 192.168.178.125:2048 entering state [ready][200]
2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare
[g729:18:8000:20]/[G729:18:8000:20]
2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec
sofia/internal/sip:211 at 192.168.178.125:2048 G729/8000 20 ms 160 samples
2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send
payload to 101
2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP
[sofia/internal/sip:211 at 192.168.178.125:2048] 192.168.178.220 port 12046
-> 192.168.178.125 port 60566 codec: 18 ms: 20
2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer
[soft] 160 bytes per 20ms
2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
payload to 101
2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
receive payload to 101
2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal
sofia/internal/200 at my.domain [BREAK]
2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel
[sofia/internal/sip:211 at 192.168.178.125:2048] has been answered
2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP
[sofia/internal/200 at my.domain] 192.168.178.220 port 12006 ->
192.168.178.50 port 12770 codec: 18 ms: 20
2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer
[soft] 160 bytes per 20ms
2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
payload to 101
2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
receive payload to 101
2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure
RTP SEND
2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure
RTP RECV
2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP
sofia/internal/200 at my.domain:
v=0
o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220
s=FreeSWITCH
c=IN IP4 192.168.178.220
t=0 0
m=audio 12006 RTP/SAVP 18 101
a=rtpmap:18 g729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK

2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal
sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel
sofia/internal/200 at my.domain entering state [completed][200]
2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal
sofia/internal/200 at my.domain [BREAK]
2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel
[sofia/internal/200 at my.domain] has been answered
2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate
Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate
Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal
sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal
sofia/internal/200 at my.domain [BREAK]
2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182
(sofia/internal/sip:211 at 192.168.178.125:2048) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send
signal sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:211 at 192.168.178.125:2048) Running State Change
CS_EXCHANGE_MEDIA
2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:211 at 192.168.178.125:2048) State EXCHANGE_MEDIA
2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA
2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching
BUG to sofia/internal/sip:211 at 192.168.178.125:2048
2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port
confirmed.
2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port
confirmed.
2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE -
0x904e070 0x8fb39c0
2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE -
0x8fcd7e8 0x8e448a0
2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching
BUG to sofia/internal/200 at my.domain
2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE
FAILED - 0x90990a0 (nil)
2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729
decoder error!
2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478
sofia/internal/200 at my.domain ending bridge by request from read function



From helmut.kuper at ewetel.de  Fri Apr 30 06:24:01 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 30 Apr 2010 15:24:01 +0200
Subject: [Freeswitch-users] Checking whether both RTP-streams are up or not
Message-ID: <4BDAD9F1.8050805@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I'm looking for an (easy) way to detect whether both RTP legs to caller
resp. callee are streaming. Just for debugging reasons. Because
sometimes there is only one RTP stream working, e.g. callee can hear
caller but not vice versa.

Is there already a solution or method in FS to debug this?

best regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFL2tnw4tZeNddg3dwRAjbaAKCEjNysCFYKkSoyo/PrE0r50xHPUQCcCoT1
ybWe6dovDrsayyFnlwKWFeI=
=lcFu
-----END PGP SIGNATURE-----


From vfclists at googlemail.com  Fri Apr 30 06:41:58 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 30 Apr 2010 14:41:58 +0100
Subject: [Freeswitch-users] How does Freeswitch CDR determine which
	gateway was used in failover?
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67035A6B0E94@mse17be1.mse17.exchange.ms>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
	<A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
	<q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B0E94@mse17be1.mse17.exchange.ms>
Message-ID: <x2y7abab2411004300641xfcf4527di1bab4f2f2f329ee0@mail.gmail.com>

On 30 April 2010 10:09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> There is/was a feature that did this: copy_xml_cdr, which sticks the b-leg
> CDR into the a-leg XML as a channel var. In practice I?ve found it sometimes
> just doesn?t work, so don?t use it if it?s still there. There?s also
> failed_xml_cdr_prefix which will write the failed b-legs into the main CDR.
> But I?m seeing a bug there too, where sometimes the b-leg CDR XML gets
> truncated. I have to do more research and I?ll open a jira on it.
>
>
>
> If you don?t need the full b-leg CDR, it?ll make your life a lot easier to
> just track the few pieces of info you need right on the a-leg by setting
> variables before bridging.
>

Could you show me an example of the above?

I am right the CDR into a database immediately and now it looks like I have
to save the b-leg and wait for the a-leg to come through before I can match
them. I would love the a-leg cdr to show which b-leg succeeded though.

Is this the  problem you are describing
http://jira.freeswitch.org/browse/FSCORE-565 ?

Is the copy_xml_cdr the stuff implemented by xml_copy_cdr in the thread?
>

-Michael
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, April 30, 2010 1:55 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] How does Freeswitch CDR determine which
> gateway was used in failover?
>
>
>
> It's late and I can't seem to locate it right now but Mathieu Rene
> mentioned that there was a way to embed the b-leg cdr into the a leg cdr.
> Math, am I losing it? :)
>
> -MC
>
> On Fri, Apr 30, 2010 at 12:32 AM, David Ponzone <david.ponzone at gmail.com>
> wrote:
>
> Frank,
>
>
>
> first of all, be sure you enabled writing leg B to CDR. If you don't, you
> won't see a gateway in there anytime soon.
>
>
>
> In CSV CDR, you can change the template used by adding the
> field ${sip_gateway_name}.
>
> In XML CDR, you get this variable automatically, of course only in the leg
> B file.
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
>
>
> Le 30/04/2010 ? 09:06, Frank Church a ?crit :
>
>
>
> I am not sure if you misunderstood my question.
>
> The XML provided by mod_xml_cdr does not tell which gateway succeeded. When
> I use a single gateway without failover I know the gateway specified in
> lastapp was used. When I use failover the last_app string contains all the
> gateways. I am looking for a value which tells which gateway was used.
>
> <include>
>   <context name="external">
>       <extension name="external_outbound">
>         <condition field="destination_number" expression="(.*)">
>             <!-- <action application="set" data="proxy_media=true"/> -->
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="set" data="originate_timeout=2"/>
>             <action application="set" data="originate_retries=1"/>
>             <action application="set" data="progress_timeout=15"/>
>             <action application="set" data="call_timeout=35"/>
>             <action application="set" data="proxy_media=true"/>
>             <action application="bridge"
> data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1"
> />
>         </condition>
>       </extension>
>   </context>
> </include>
>
>
>
>
> On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com> wrote:
> > set a var for each leg, this is how I do it in mod_lcr:
> > [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
> >
> > On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com>
> > wrote:
> >>
> >> How does Freeswitch CDR determine which gateway was used in failover?
> >>
> >> I am looking at the xml cdr and cdr_csv and they don't appear to show
> >> which gateway was used in the bridge data.
> >>
> >> Is there a separate uuid for each bridging attempt, that can be
> >> obtained from the  CDR,  or the event socket as last resort?
> >>
> >> --
> >> Frank Church
> >>
> >> =======================
> >> http://devblog.brahmancreations.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
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From david.ponzone at gmail.com  Fri Apr 30 06:55:04 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 30 Apr 2010 15:55:04 +0200
Subject: [Freeswitch-users] Checking whether both RTP-streams are up or
	not
In-Reply-To: <4BDAD9F1.8050805@ewetel.de>
References: <4BDAD9F1.8050805@ewetel.de>
Message-ID: <48D4CC4E-1832-4126-B52D-82B3F7C546CA@gmail.com>

Well, there is a way, it's called thark.

Ok it's not included in FS, but it's really powerful :)


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 30/04/2010 ? 15:24, Helmut Kuper a ?crit :

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I'm looking for an (easy) way to detect whether both RTP legs to  
> caller
> resp. callee are streaming. Just for debugging reasons. Because
> sometimes there is only one RTP stream working, e.g. callee can hear
> caller but not vice versa.
>
> Is there already a solution or method in FS to debug this?
>
> best regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFL2tnw4tZeNddg3dwRAjbaAKCEjNysCFYKkSoyo/PrE0r50xHPUQCcCoT1
> ybWe6dovDrsayyFnlwKWFeI=
> =lcFu
> -----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

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From stevendt at primrosebank.net  Fri Apr 30 07:02:55 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 30 Apr 2010 15:02:55 +0100
Subject: [Freeswitch-users] Picking up voicemail
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>
	<i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com>
Message-ID: <EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>

Hi Mike,

thanks a lot for the reply. I would still like to be able to get into the rest of the system, i.e., other mailboxes than the default, so if someone can shed any more light on this, that would be great.

In the meantime though, you are absolutely right on the default behaviour! Hitting "0" when the outgoing message is being played does get you to the default mailbox. I was not aware of this and there seems to be another problem which has muddied the waters for me. When I test various bit and pieces coming in from the PSTN line, I have been using my mobile phone (cell phone to you guys), the fact that my cell phone does not behave the same way as a land line seems to have been the problem.

It seems that hitting a key on the cell phone is picked up by FreeSWITCH to drop you into the voice mail system, but entering the "#" to enter the mailbox password isn't.

However, when I do the same thing from a land-line (actually, a VOIP phone from the office) the mailbox password is correctly recognised. The problem seems to be recognising the "#" entered through a cell phone. Are you aware of any issues with cell phone tones being recognised by FreeSWITCH ?

regards
Dave

  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, April 29, 2010 8:41 PM
  Subject: Re: [Freeswitch-users] Picking up voicemail





  On Thu, Apr 29, 2010 at 11:15 AM, Dave Stevenson <stevendt at primrosebank.net> wrote:

    Hi,

    I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good.

    I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension. 

  I think by default you can dial 0 while listening to the voicemail outgoing message to log into the voicemail, however I don't believe there's a default way of turning the vm login into a gateway to the rest of the system. Anyone?
  -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
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From brian at freeswitch.org  Fri Apr 30 07:05:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Apr 2010 09:05:51 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BDABDA9.5020707@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
	<4BDABDA9.5020707@gmx.net>
Message-ID: <E8427DA6-35E1-4CA1-BDD9-38861BCC78E8@freeswitch.org>

Can you refer your support with the commercial G729 to consulting at freeswitch.org.

I'll try to reproduce this issue today and see if I have the same results.

/b

On Apr 30, 2010, at 6:23 AM, Peter P GMX wrote:

> I just updated, with the same result:
> 
> 
> Anthony Minessale schrieb:
>> do you have lastest git HEAD ?
>> can you update and try again?
>> 
>> 
> 
> Here's the log:



From infos at madovsky.org  Thu Apr 29 19:25:43 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 29 Apr 2010 22:25:43 -0400
Subject: [Freeswitch-users] fs_cli
Message-ID: <7B847F47C4BE42019B25DE706841F10B@MOBILEE1705>

Hi all,

when I run from bash as "root"
I get this error:
/usr/local/freeswitch/bin/fs_cli -x "any_command" 
[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket Connection Error]


Freeswitch is running without problems.
I think I forgot to something....
svn17134M

Thanks

Franck
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From helmut.kuper at ewetel.de  Fri Apr 30 07:08:09 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 30 Apr 2010 16:08:09 +0200
Subject: [Freeswitch-users] Checking whether both RTP-streams are up or
 not
In-Reply-To: <48D4CC4E-1832-4126-B52D-82B3F7C546CA@gmail.com>
References: <4BDAD9F1.8050805@ewetel.de>
	<48D4CC4E-1832-4126-B52D-82B3F7C546CA@gmail.com>
Message-ID: <4BDAE449.4000308@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi David,

well, thx for your hint.

I', looking for a way to have this information in freeswitch's log so I
would be able to track such issues even when they occured some days ago ...



On 30.04.2010 15:55, David Ponzone wrote:
> Well, there is a way, it's called thark.
> 
> Ok it's not included in FS, but it's really powerful :)
> 
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1v4gIKxZNp11AWSIp+G2HKY=
=0QIh
-----END PGP SIGNATURE-----


From david.ponzone at gmail.com  Fri Apr 30 07:11:36 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 30 Apr 2010 16:11:36 +0200
Subject: [Freeswitch-users] Picking up voicemail
In-Reply-To: <EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>
	<i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com>
	<EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
Message-ID: <EEA1A3DC-2DD3-4855-8082-B52528A831D9@gmail.com>

Dave,

first of all, I think you should tell us which DTMF method you  
configured on the SPA.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/04/2010 ? 16:02, Dave Stevenson a ?crit :

> Hi Mike,
>
> thanks a lot for the reply. I would still like to be able to get  
> into the rest of the system, i.e., other mailboxes than the default,  
> so if someone can shed any more light on this, that would be great.
>
> In the meantime though, you are absolutely right on the default  
> behaviour! Hitting "0" when the outgoing message is being played  
> does get you to the default mailbox. I was not aware of this and  
> there seems to be another problem which has muddied the waters for  
> me. When I test various bit and pieces coming in from the PSTN line,  
> I have been using my mobile phone (cell phone to you guys), the fact  
> that my cell phone does not behave the same way as a land line seems  
> to have been the problem.
>
> It seems that hitting a key on the cell phone is picked up by  
> FreeSWITCH to drop you into the voice mail system, but entering the  
> "#" to enter the mailbox password isn't.
>
> However, when I do the same thing from a land-line (actually, a VOIP  
> phone from the office) the mailbox password is correctly recognised.  
> The problem seems to be recognising the "#" entered through a cell  
> phone. Are you aware of any issues with cell phone tones being  
> recognised by FreeSWITCH ?
>
> regards
> Dave
>
> ----- Original Message -----
> From: Michael Collins
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, April 29, 2010 8:41 PM
> Subject: Re: [Freeswitch-users] Picking up voicemail
>
>
>
> On Thu, Apr 29, 2010 at 11:15 AM, Dave Stevenson <stevendt at primrosebank.net 
> > wrote:
> Hi,
>
> I have FreeSWITCH running over an internal IP network, connected to  
> the PSTN line with an SPA-3102. When an outside (PSTN) call comes  
> in, a Group of phones are rung until the first one picks up or until  
> the caller falls over to voice mail and can leave a message in the  
> Group Mailbox (100). I can retrieve the messages from any local  
> phone - so far, so good.
>
> I'd like to be able to call in through the same PSTN line and  
> interrogate the group mailbox, or any mailbox come to that. Can  
> someone point me in the right direction for how to set this up  
> please ? i.e., once FreeSwitch has answered the call and the  
> voicemail prompts have stated, can they be interrupted so that I can  
> get to the voice mail extension.
>
> I think by default you can dial 0 while listening to the voicemail  
> outgoing message to log into the voicemail, however I don't believe  
> there's a default way of turning the vm login into a gateway to the  
> rest of the system. Anyone?
> -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 helmut.kuper at ewetel.de  Fri Apr 30 07:31:11 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 30 Apr 2010 16:31:11 +0200
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <7B847F47C4BE42019B25DE706841F10B@MOBILEE1705>
References: <7B847F47C4BE42019B25DE706841F10B@MOBILEE1705>
Message-ID: <4BDAE9AF.3020602@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

by default fs_cli connects to 127.0.0.1 (localhost).

I guess you hit either the access list configured in acl.conf.xml and
event_socket.conf.xml or you just have to add the -H parameter (for
host) to fs_cli.

regards
helmut
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Version: GnuPG v1.4.7 (MingW32)

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=mx8U
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From stevendt at primrosebank.net  Fri Apr 30 07:39:34 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 30 Apr 2010 15:39:34 +0100
Subject: [Freeswitch-users] Picking up voicemail
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com><i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com><EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
	<EEA1A3DC-2DD3-4855-8082-B52528A831D9@gmail.com>
Message-ID: <D19337C46911437BB308AC6BCFBEB556@bp1.ad.bp.com>

Hi David,

thanks for the follow up - the SPA-3102 has "DTMF Tx" configuration options under Voice for Line 1 and PSTN Line.

These are currently both set to the Default (Auto), the available options are :-
DTMF Tx Method
    Inband
    AVT
    INFO
    Inband+INFO
    AVT+INFO

I could change these as you might recommend, but I'm not sure why this would make a difference when the SPA seems to respond OK to phones (at least the one that I've tried) other than my cell phone correctly. Would the DTMF setting not affect everything equally ?

regards
Dave



  ----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, April 30, 2010 3:11 PM
  Subject: Re: [Freeswitch-users] Picking up voicemail


  Dave,


  first of all, I think you should tell us which DTMF method you configured on the SPA.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 30/04/2010 ? 16:02, Dave Stevenson a ?crit :


    Hi Mike,

    thanks a lot for the reply. I would still like to be able to get into the rest of the system, i.e., other mailboxes than the default, so if someone can shed any more light on this, that would be great.

    In the meantime though, you are absolutely right on the default behaviour! Hitting "0" when the outgoing message is being played does get you to the default mailbox. I was not aware of this and there seems to be another problem which has muddied the waters for me. When I test various bit and pieces coming in from the PSTN line, I have been using my mobile phone (cell phone to you guys), the fact that my cell phone does not behave the same way as a land line seems to have been the problem.

    It seems that hitting a key on the cell phone is picked up by FreeSWITCH to drop you into the voice mail system, but entering the "#" to enter the mailbox password isn't.

    However, when I do the same thing from a land-line (actually, a VOIP phone from the office) the mailbox password is correctly recognised. The problem seems to be recognising the "#" entered through a cell phone. Are you aware of any issues with cell phone tones being recognised by FreeSWITCH ?

    regards
    Dave

      ----- Original Message -----
      From: Michael Collins
      To: freeswitch-users at lists.freeswitch.org
      Sent: Thursday, April 29, 2010 8:41 PM
      Subject: Re: [Freeswitch-users] Picking up voicemail





      On Thu, Apr 29, 2010 at 11:15 AM, Dave Stevenson <stevendt at primrosebank.net> wrote:

        Hi,

        I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good.

        I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension.

      I think by default you can dial 0 while listening to the voicemail outgoing message to log into the voicemail, however I don't believe there's a default way of turning the vm login into a gateway to the rest of the system. Anyone?
      -MC






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From brian at freeswitch.org  Fri Apr 30 07:43:08 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Apr 2010 09:43:08 -0500
Subject: [Freeswitch-users] Picking up voicemail
In-Reply-To: <D19337C46911437BB308AC6BCFBEB556@bp1.ad.bp.com>
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com><i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com><EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
	<EEA1A3DC-2DD3-4855-8082-B52528A831D9@gmail.com>
	<D19337C46911437BB308AC6BCFBEB556@bp1.ad.bp.com>
Message-ID: <0E6EE42B-7346-4150-87F5-992276EE11E6@freeswitch.org>

AVT
On Apr 30, 2010, at 9:39 AM, Dave Stevenson wrote:

> Hi David,
>  
> thanks for the follow up - the SPA-3102 has "DTMF Tx" configuration options under Voice for Line 1 and PSTN Line.
>  
> These are currently both set to the Default (Auto), the available options are :-
> DTMF Tx Method
>     Inband
>     AVT
>     INFO
>     Inband+INFO
>     AVT+INFO
>  
> I could change these as you might recommend, but I'm not sure why this would make a difference when the SPA seems to respond OK to phones (at least the one that I've tried) other than my cell phone correctly. Would the DTMF setting not affect everything equally ?
>  
> regards
> Dave

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From stevendt at primrosebank.net  Fri Apr 30 07:53:26 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 30 Apr 2010 15:53:26 +0100
Subject: [Freeswitch-users] Picking up voicemail
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com><i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com><EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com><EEA1A3DC-2DD3-4855-8082-B52528A831D9@gmail.com><D19337C46911437BB308AC6BCFBEB556@bp1.ad.bp.com>
	<0E6EE42B-7346-4150-87F5-992276EE11E6@freeswitch.org>
Message-ID: <1BD8956DAAE045D7AD44E2401A6496C4@bp1.ad.bp.com>

Hi Brian,

thanks for the pointer.

The SPA-3012 is pretty strong on configuration options, but the documentation is very light on what they actually mean. Just so that I know, what is AVT actually doing ?

regards 
Dave
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, April 30, 2010 3:43 PM
  Subject: Re: [Freeswitch-users] Picking up voicemail


  AVT

  On Apr 30, 2010, at 9:39 AM, Dave Stevenson wrote:


    Hi David,

    thanks for the follow up - the SPA-3102 has "DTMF Tx" configuration options under Voice for Line 1 and PSTN Line.

    These are currently both set to the Default (Auto), the available options are :-
    DTMF Tx Method
        Inband
        AVT
        INFO
        Inband+INFO
        AVT+INFO

    I could change these as you might recommend, but I'm not sure why this would make a difference when the SPA seems to respond OK to phones (at least the one that I've tried) other than my cell phone correctly. Would the DTMF setting not affect everything equally ?

    regards
    Dave




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From jerry.richards at teotech.com  Fri Apr 30 08:00:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 30 Apr 2010 08:00:48 -0700
Subject: [Freeswitch-users] No PRI Call Until After Manual
	"loadmod_openzap"
In-Reply-To: <57A82550-5541-4DE5-A12C-D906A30A60C7@freeswitch.org>
References: <5D4D561C47D74E1988B32D78CD6349B7@greyhawk.tonecommander.com>
	<57A82550-5541-4DE5-A12C-D906A30A60C7@freeswitch.org>
Message-ID: <333479949FF94115B17343A7FE5C1CAF@greyhawk.tonecommander.com>

Correct.  I used to use to following command to populate the
conf/autload_config/modules.conf.xml, but it appears that it has been
removed.  What would be the standard way of doing this now?
 
contrib/trixter/makemodconf.pl modules.conf >
/opt/server/conf/autoload_configs/modules.conf.xml
 
Thanks,
Jerry


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, April 29, 2010 3:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No PRI Call Until After Manual
"loadmod_openzap"


I'm going to guess its not in modules.conf.xml to load on start? 

/b

On Apr 29, 2010, at 5:31 PM, Jerry Richards wrote:



I upgraded to FS snapshot as of 04-27-2010 and Wanpipe 3.5.11 and now, for
some reason, I cannot make an outbound call unless I manual execute "load
mod_openzap".  Anyone know why?  I posted the initialization, failed call
attempt, load mod_openzap, and successful call at
http://pastebin.freeswitch.org/12845.



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From brian at freeswitch.org  Fri Apr 30 08:04:13 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Apr 2010 10:04:13 -0500
Subject: [Freeswitch-users] No PRI Call Until After Manual
	"loadmod_openzap"
In-Reply-To: <333479949FF94115B17343A7FE5C1CAF@greyhawk.tonecommander.com>
References: <5D4D561C47D74E1988B32D78CD6349B7@greyhawk.tonecommander.com>
	<57A82550-5541-4DE5-A12C-D906A30A60C7@freeswitch.org>
	<333479949FF94115B17343A7FE5C1CAF@greyhawk.tonecommander.com>
Message-ID: <A734EB4B-5C17-4E76-BD0B-2974BDBD0569@freeswitch.org>

Edit opt/server/conf/autoload_configs/modules.conf.xml by hand.

/b

On Apr 30, 2010, at 10:00 AM, Jerry Richards wrote:

> Correct.  I used to use to following command to populate the conf/autload_config/modules.conf.xml, but it appears that it has been removed.  What would be the standard way of doing this now?
>  
> contrib/trixter/makemodconf.pl modules.conf > /opt/server/conf/autoload_configs/modules.conf.xml
>  
> Thanks,
> Jerry



From peder at networkoblivion.com  Fri Apr 30 08:19:17 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 30 Apr 2010 10:19:17 -0500
Subject: [Freeswitch-users] Checking whether both RTP-streams are up or
	not
In-Reply-To: <4BDAE449.4000308@ewetel.de>
References: <4BDAD9F1.8050805@ewetel.de>	<48D4CC4E-1832-4126-B52D-82B3F7C546CA@gmail.com>
	<4BDAE449.4000308@ewetel.de>
Message-ID: <0bf401cae878$8055d9f0$81018dd0$@com>

The problem is that even though FS sends rtp, there is no way to know if the
phone on the far end receives it or not.  If they have a nat issue, the only
way to know if they receive rtp is to get a phone that can send per calls
stats, or have the user say "yes, audio worked both ways".

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Helmut
Kuper
Sent: Friday, April 30, 2010 9:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Checking whether both RTP-streams are up or
not

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi David,

well, thx for your hint.

I', looking for a way to have this information in freeswitch's log so I
would be able to track such issues even when they occured some days ago ...



On 30.04.2010 15:55, David Ponzone wrote:
> Well, there is a way, it's called thark.
> 
> Ok it's not included in FS, but it's really powerful :)
> 
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFL2uRJ4tZeNddg3dwRAmPzAJ90M5OaMymwvSUN2GxFlsmRRKEWNgCbBHqH
1v4gIKxZNp11AWSIp+G2HKY=
=0QIh
-----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  Fri Apr 30 08:32:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Apr 2010 10:32:37 -0500
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <4BDAE9AF.3020602@ewetel.de>
References: <7B847F47C4BE42019B25DE706841F10B@MOBILEE1705>
	<4BDAE9AF.3020602@ewetel.de>
Message-ID: <x2x191c3a031004300832u78ede026k9478dbf20926b237@mail.gmail.com>

or load mod_event_socket ?

On Fri, Apr 30, 2010 at 9:31 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi,
>
> by default fs_cli connects to 127.0.0.1 (localhost).
>
> I guess you hit either the access list configured in acl.conf.xml and
> event_socket.conf.xml or you just have to add the -H parameter (for
> host) to fs_cli.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFL2umv4tZeNddg3dwRAtf5AKCVVZ4N8pXZxsUCAvvj84plLfBybwCeKgWe
> 5AZhTFAUGLbLwf5JR9j0hhQ=
> =mx8U
> -----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/
Twitter: http://twitter.com/FreeSWITCH_wire

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

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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  Fri Apr 30 08:34:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Apr 2010 10:34:18 -0500
Subject: [Freeswitch-users] Checking whether both RTP-streams are up or
	not
In-Reply-To: <0bf401cae878$8055d9f0$81018dd0$@com>
References: <4BDAD9F1.8050805@ewetel.de>
	<48D4CC4E-1832-4126-B52D-82B3F7C546CA@gmail.com>
	<4BDAE449.4000308@ewetel.de> <0bf401cae878$8055d9f0$81018dd0$@com>
Message-ID: <t2t191c3a031004300834u91331c8bzf52fe8577b371862@mail.gmail.com>

see uuid_debug_media cli command
but you still probably also want wireshark, netcat or ngrep etc.


On Fri, Apr 30, 2010 at 10:19 AM, Peder <peder at networkoblivion.com> wrote:

> The problem is that even though FS sends rtp, there is no way to know if
> the
> phone on the far end receives it or not.  If they have a nat issue, the
> only
> way to know if they receive rtp is to get a phone that can send per calls
> stats, or have the user say "yes, audio worked both ways".
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Helmut
> Kuper
> Sent: Friday, April 30, 2010 9:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Checking whether both RTP-streams are up or
> not
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi David,
>
> well, thx for your hint.
>
> I', looking for a way to have this information in freeswitch's log so I
> would be able to track such issues even when they occured some days ago ...
>
>
>
> On 30.04.2010 15:55, David Ponzone wrote:
> > Well, there is a way, it's called thark.
> >
> > Ok it's not included in FS, but it's really powerful :)
> >
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFL2uRJ4tZeNddg3dwRAmPzAJ90M5OaMymwvSUN2GxFlsmRRKEWNgCbBHqH
> 1v4gIKxZNp11AWSIp+G2HKY=
> =0QIh
> -----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
>
>
> _______________________________________________
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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/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Apr 30 08:35:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Apr 2010 10:35:34 -0500
Subject: [Freeswitch-users] user_exists / user_registered?
In-Reply-To: <8BBDD745-D92C-48F7-A5C2-3ACF0BEFA390@gmail.com>
References: <p2z367751821004291258g5a0e1ddfqc72ff233835b93b3@mail.gmail.com>
	<AE8878FE-6945-4C40-86D5-F20B4ACD2859@gmail.com>
	<m2o367751821004291341x2682f217x17c71580407563cf@mail.gmail.com>
	<t2o59ad9ca11004300126l97ebfa00i283948e6f642a57a@mail.gmail.com>
	<8BBDD745-D92C-48F7-A5C2-3ACF0BEFA390@gmail.com>
Message-ID: <k2w191c3a031004300835n2abab6a2i58dd24f291987a2b@mail.gmail.com>

or provision the phone to only allow one call at a time and it will reject
it when someone is on the phone.


On Fri, Apr 30, 2010 at 3:36 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> A way, quite tricky, is mod_limit.
>
> Another way, afaik, is to query the internal DB.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 30/04/2010 ? 10:26, Henry Huang a ?crit :
>
> Does it also tells you if a user is busy?
> If not, what should I do to find out if a user is on phone already so I can
> send the second call to some place else?
>
> Thanks,
>
> Henry
>
> On Thu, Apr 29, 2010 at 1:41 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Thank you all!
>>
>>
>> On Thu, Apr 29, 2010 at 4:13 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> Phillip,
>>>
>>> perhaos not the nicest way, but you could use sofia_contact.
>>>
>>> sofia_contact <profile>/<user id>@<domain>
>>>
>>> If it returns:
>>> error/user_not_registered
>>> then you know it is not registered.
>>>
>>> if it is, the answer would be a string starting with:
>>> sofia/<profile>/etc...
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 29/04/2010 ? 21:58, Phillip Jones a ?crit :
>>>
>>> Hi there,
>>>
>>> You can call the api command user_exists - to see it the user is defined
>>> in the /directory.
>>>
>>> Is there an api command that tells you whether a user is currently
>>> registered? Like user_registered??
>>>
>>>
>>>
>>> Thanks
>>>
>>> Pj
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
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From anthony.minessale at gmail.com  Fri Apr 30 08:43:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Apr 2010 10:43:07 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BDABDA9.5020707@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
	<4BDABDA9.5020707@gmx.net>
Message-ID: <p2u191c3a031004300843qf4ef4b28ma406ad73ef54cd1c@mail.gmail.com>

log snippets are not useful.

as always, we need full log from beginning to end of the complete call
with the inline sip trace enabled.



On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I just updated, with the same result:
>
>
> Anthony Minessale schrieb:
> > do you have lastest git HEAD ?
> > can you update and try again?
> >
> >
>
> Here's the log:
>
> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP:
> v=0
> o=root 929923105 929923106 IN IP4 192.168.178.125
> s=call
> c=IN IP4 192.168.178.125
> t=0 0
> m=audio 60566 RTP/AVP 18 8 0 99 3 101
> a=rtpmap:18 g729/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:0 pcmu/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel
> sofia/internal/sip:211 at 192.168.178.125:2048 entering state [ready][200]
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare
> [g729:18:8000:20]/[G729:18:8000:20]
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec
> sofia/internal/sip:211 at 192.168.178.125:2048 G729/8000 20 ms 160 samples
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP
> [sofia/internal/sip:211 at 192.168.178.125:2048] 192.168.178.220 port 12046
> -> 192.168.178.125 port 60566 codec: 18 ms: 20
> 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer
> [soft] 160 bytes per 20ms
> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
> receive payload to 101
> 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel
> [sofia/internal/sip:211 at 192.168.178.125:2048] has been answered
> 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP
> [sofia/internal/200 at my.domain] 192.168.178.220 port 12006 ->
> 192.168.178.50 port 12770 codec: 18 ms: 20
> 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer
> [soft] 160 bytes per 20ms
> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
> receive payload to 101
> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure
> RTP SEND
> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure
> RTP RECV
> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP
> sofia/internal/200 at my.domain:
> v=0
> o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220
> s=FreeSWITCH
> c=IN IP4 192.168.178.220
> t=0 0
> m=audio 12006 RTP/SAVP 18 101
> a=rtpmap:18 g729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
>
> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal
> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel
> sofia/internal/200 at my.domain entering state [completed][200]
> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel
> [sofia/internal/200 at my.domain] has been answered
> 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate
> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
> 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate
> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
> 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182
> (sofia/internal/sip:211 at 192.168.178.125:2048) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:211 at 192.168.178.125:2048) Running State Change
> CS_EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:211 at 192.168.178.125:2048) State EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA
> 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/internal/sip:211 at 192.168.178.125:2048
> 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port
> confirmed.
> 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port
> confirmed.
> 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE -
> 0x904e070 0x8fb39c0
> 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE -
> 0x8fcd7e8 0x8e448a0
> 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/internal/200 at my.domain
> 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE
> FAILED - 0x90990a0 (nil)
> 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729
> decoder error!
> 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478
> sofia/internal/200 at my.domain ending bridge by request from read function
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mgg at giagnocavo.net  Fri Apr 30 10:06:43 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 30 Apr 2010 13:06:43 -0400
Subject: [Freeswitch-users] How does Freeswitch CDR determine
	which	gateway was used in failover?
In-Reply-To: <x2y7abab2411004300641xfcf4527di1bab4f2f2f329ee0@mail.gmail.com>
References: <m2l7abab2411004291820i3d6d1d96hd18b969f2b87347f@mail.gmail.com>
	<y2vb39273531004292009oc1928607n3aea75e838f8ef42@mail.gmail.com>
	<u2l7abab2411004300006k7a4f35cdw3a6fbe543c34580c@mail.gmail.com>
	<A2551E66-8C1F-41E2-803F-ABCC99E46EDC@gmail.com>
	<q2j87f2f3b91004300055j309c7ec6lb6b6defbc0505991@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67035A6B0E94@mse17be1.mse17.exchange.ms>
	<x2y7abab2411004300641xfcf4527di1bab4f2f2f329ee0@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67035A6B0F25@mse17be1.mse17.exchange.ms>

Oh, what I meant was setting a channel variable like "Now_calling_gateway=foo", then doing the bridge. I do not know of any simple way that drops all the b-leg GUIDs onto the a-leg XML. But, now that you mention it, this would be a smart thing to do. Then we can just scan a-leg XML, and wait for all b-legs to arrive.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank Church
Sent: Friday, April 30, 2010 7:42 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How does Freeswitch CDR determine which gateway was used in failover?


On 30 April 2010 10:09, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:
There is/was a feature that did this: copy_xml_cdr, which sticks the b-leg CDR into the a-leg XML as a channel var. In practice I've found it sometimes just doesn't work, so don't use it if it's still there. There's also failed_xml_cdr_prefix which will write the failed b-legs into the main CDR. But I'm seeing a bug there too, where sometimes the b-leg CDR XML gets truncated. I have to do more research and I'll open a jira on it.

If you don't need the full b-leg CDR, it'll make your life a lot easier to just track the few pieces of info you need right on the a-leg by setting variables before bridging.

Could you show me an example of the above?

I am right the CDR into a database immediately and now it looks like I have to save the b-leg and wait for the a-leg to come through before I can match them. I would love the a-leg cdr to show which b-leg succeeded though.

Is this the  problem you are describing http://jira.freeswitch.org/browse/FSCORE-565 ?
Is the copy_xml_cdr the stuff implemented by xml_copy_cdr in the thread?

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Friday, April 30, 2010 1:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] How does Freeswitch CDR determine which gateway was used in failover?

It's late and I can't seem to locate it right now but Mathieu Rene mentioned that there was a way to embed the b-leg cdr into the a leg cdr. Math, am I losing it? :)

-MC
On Fri, Apr 30, 2010 at 12:32 AM, David Ponzone <david.ponzone at gmail.com<mailto:david.ponzone at gmail.com>> wrote:
Frank,

first of all, be sure you enabled writing leg B to CDR. If you don't, you won't see a gateway in there anytime soon.

In CSV CDR, you can change the template used by adding the field ${sip_gateway_name}.
In XML CDR, you get this variable automatically, of course only in the leg B file.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<http://www.ipeva.fr>  -   www.ipeva-studio.com<http://www.ipeva-studio.com>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 30/04/2010 ? 09:06, Frank Church a ?crit :

I am not sure if you misunderstood my question.

The XML provided by mod_xml_cdr does not tell which gateway succeeded. When I use a single gateway without failover I know the gateway specified in lastapp was used. When I use failover the last_app string contains all the gateways. I am looking for a value which tells which gateway was used.
<include>
  <context name="external">
      <extension name="external_outbound">
        <condition field="destination_number" expression="(.*)">
            <!-- <action application="set" data="proxy_media=true"/> -->
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="originate_timeout=2"/>
            <action application="set" data="originate_retries=1"/>
            <action application="set" data="progress_timeout=15"/>
            <action application="set" data="call_timeout=35"/>
            <action application="set" data="proxy_media=true"/>
            <action application="bridge" data="sofia/gateway/provider1/$1|sofia/gateway/provider2/$1|sofia/gateway/provider3/$1" />
        </condition>
      </extension>
  </context>
</include>



On 30 April 2010 04:09, Rupa Schomaker <rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
> set a var for each leg, this is how I do it in mod_lcr:
> [gateway=foo]sofia/gateway/foo/$1|[gateway=bar]sofia/gateway/bar/$1
>
> On Thu, Apr 29, 2010 at 8:20 PM, Frank Church <vfclists at googlemail.com<mailto:vfclists at googlemail.com>>
> wrote:
>>
>> How does Freeswitch CDR determine which gateway was used in failover?
>>
>> I am looking at the xml cdr and cdr_csv and they don't appear to show
>> which gateway was used in the bridge data.
>>
>> Is there a separate uuid for each bridging attempt, that can be
>> obtained from the  CDR,  or the event socket as last resort?
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>> _______________________________________________
>> 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
>
>
>
> --
> -Rupa
>
> _______________________________________________
> 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
>
>



--
Frank Church

=======================
http://devblog.brahmancreations.com

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


_______________________________________________
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_______________________________________________
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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--
Frank Church

=======================
http://devblog.brahmancreations.com
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From tomb at cachecomm.com  Fri Apr 30 11:33:16 2010
From: tomb at cachecomm.com (Tom)
Date: Fri, 30 Apr 2010 12:33:16 -0600
Subject: [Freeswitch-users] libpri not working
Message-ID: <4BDB226C.8050608@cachecomm.com>

Voip gods,
having problem with libpri on incoming call it gives me this [WARNING] 
ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored)
then it wait a few seconds then i get a fast busy. If i turn off libpri 
i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no 
matching channel 0

Iam running
FreeSWITCH Version 1.0.head (svn-17188M)
wanpipe-3.5.10.14
dahdi-2.3.0
Sangoma B601de card


Tom Baldwin


From msc at freeswitch.org  Fri Apr 30 14:39:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Apr 2010 14:39:48 -0700
Subject: [Freeswitch-users] libpri not working
In-Reply-To: <4BDB226C.8050608@cachecomm.com>
References: <4BDB226C.8050608@cachecomm.com>
Message-ID: <l2v87f2f3b91004301439g1a1b099fg828d50351f975dbd@mail.gmail.com>

Haha, you are asking a TDM question of the "VoIP gods" :)

I recommend you join #openzap on irc.freenode.net and talk to moy from
Sangoma. He can tell you what the state of BRI is with their stack or if you
need to stick with libpri. (The stock OpenZAP ISDN stack doesn't do PRI.)

-MC

On Fri, Apr 30, 2010 at 11:33 AM, Tom <tomb at cachecomm.com> wrote:

> Voip gods,
> having problem with libpri on incoming call it gives me this [WARNING]
> ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored)
> then it wait a few seconds then i get a fast busy. If i turn off libpri
> i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no
> matching channel 0
>
> Iam running
> FreeSWITCH Version 1.0.head (svn-17188M)
> wanpipe-3.5.10.14
> dahdi-2.3.0
> Sangoma B601de card
>
>
> Tom Baldwin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From grsingh750 at gmail.com  Fri Apr 30 14:45:50 2010
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 1 May 2010 03:15:50 +0530
Subject: [Freeswitch-users] Lag on extension behind spa3102
Message-ID: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>

Hi,
I am running FreeSWITCH Version 1.0.trunk (17134).
I just got my spa3102 and have configured my phone as extension 1000.
There's no PSTN or external VoIP yet. I am getting a lag whenever I dial out
from the phone. For eg. when I dial the voicemail(4000) I hear "please enter
your id....." not the part before that. There's a 10 sec lag when I call the
demo IVR, there's also a lag when I dial other extensions(softphones).
There's no lag whatsoever while dialing from softphones.

Please recommend where I should be looking to resolve this before I move on
to more adventurous stuff :)

Thanks
gs
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From msc at freeswitch.org  Fri Apr 30 14:58:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Apr 2010 14:58:27 -0700
Subject: [Freeswitch-users] Lag on extension behind spa3102
In-Reply-To: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
References: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
Message-ID: <j2i87f2f3b91004301458s5da269ccj8071afa6f68dd84@mail.gmail.com>

Get a pcap of all the traffic to/from the spa3102 and open it in wireshark.
Look for clues in the SIP dialog. If you are having trouble analyzing then
export the SIP trace from wireshark and drop it into
pastebin.freeswitch.organd link to it in the thread so others can have
a look.

-MC

On Fri, Apr 30, 2010 at 2:45 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
> I am running FreeSWITCH Version 1.0.trunk (17134).
> I just got my spa3102 and have configured my phone as extension 1000.
> There's no PSTN or external VoIP yet. I am getting a lag whenever I dial out
> from the phone. For eg. when I dial the voicemail(4000) I hear "please enter
> your id....." not the part before that. There's a 10 sec lag when I call the
> demo IVR, there's also a lag when I dial other extensions(softphones).
> There's no lag whatsoever while dialing from softphones.
>
> Please recommend where I should be looking to resolve this before I move on
> to more adventurous stuff :)
>
> Thanks
> gs
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Fri Apr 30 15:26:32 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 30 Apr 2010 18:26:32 -0400
Subject: [Freeswitch-users] mobile phone clients / fs cluster
In-Reply-To: <p2sb74751491004292211i4d66c1d2me49ffcfc18852692@mail.gmail.com>
References: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>
	<p2sb74751491004292211i4d66c1d2me49ffcfc18852692@mail.gmail.com>
Message-ID: <y2m367751821004301526r17a57e70o45ec9ba275539095@mail.gmail.com>

Ram,

I think thats what I am asking? I am not sure how the 3G network works.
Whether there are restrictions on how servers can communicate to clients
etc.

Perhaps I am just way of base also. I don't know. TGIF.

Pj


On Fri, Apr 30, 2010 at 1:11 AM, ram <talk2ram at gmail.com> wrote:

> Hi
>
> why not its possible
>
> Ram
>
> On Fri, Apr 30, 2010 at 2:26 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy.
>> Outbound (terminating) traffic goes directly from each FS box.
>>
>> All my home office phones get calls no matter which box they are
>> registered with or which box the call comes in on - thanks Anthony.
>>
>> However my SIP client on various iPhone/Androids etc only receive calls
>> that originate on the box on which they are registered.
>>
>> Looking at the SIP trace - when the call comes in on the 'wrong' box, the
>> invites to these SIP clients do not even get a response. Presumably because
>> the 3G network has no idea who this new IP in the "from address" is, who
>> trying to contact them.
>>
>> Question is, is there a way around this - our will I have start routing
>> all terminating traffic out through the proxy also.
>>
>> Any insight appreciated.
>>
>> Thanks!
>>
>> Pj
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Fri Apr 30 16:26:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 30 Apr 2010 23:26:48 +0000
Subject: [Freeswitch-users] Lag on extension behind spa3102
In-Reply-To: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
References: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E99EB32B@prato.activenetwerx.local>

>Please recommend where I should be looking to resolve this before I move on to more adventurous stuff :)

First of all, I will pass my condolences on with that purchase.
That device is a major steaming pile of shit:)

But, on to your problem:) It has a dial pattern internally for
Vertical Service Activation Codes and for distinction on what
to pass outward.

Yours is waiting as a result of the dial pattern along with
the time out specified.

http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5179

Disable the act codes and tweak your dp on it...

HTH,
jlc


From grsingh750 at gmail.com  Fri Apr 30 17:01:25 2010
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 1 May 2010 05:31:25 +0530
Subject: [Freeswitch-users] Lag on extension behind spa3102
In-Reply-To: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
References: <w2x7b463d951004301445jf287916fw3c8db8853e1579d5@mail.gmail.com>
Message-ID: <g2v7b463d951004301701v5f4f8b62k458ad7936e014cf9@mail.gmail.com>

Solved! The default dialplan was configured for N. America.
I am still trying to write the write dial plan for India.
Thanks to R-Guy on irc.

Thanks
gs

On Sat, May 1, 2010 at 3:15 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
> I am running FreeSWITCH Version 1.0.trunk (17134).
> I just got my spa3102 and have configured my phone as extension 1000.
> There's no PSTN or external VoIP yet. I am getting a lag whenever I dial out
> from the phone. For eg. when I dial the voicemail(4000) I hear "please enter
> your id....." not the part before that. There's a 10 sec lag when I call the
> demo IVR, there's also a lag when I dial other extensions(softphones).
> There's no lag whatsoever while dialing from softphones.
>
> Please recommend where I should be looking to resolve this before I move on
> to more adventurous stuff :)
>
> Thanks
> gs
>
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From freeswitch.org at todandlorna.com  Fri Apr 30 17:02:44 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Fri, 30 Apr 2010 18:02:44 -0600
Subject: [Freeswitch-users] mobile phone clients / fs cluster
In-Reply-To: <y2m367751821004301526r17a57e70o45ec9ba275539095@mail.gmail.com>
References: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>	<p2sb74751491004292211i4d66c1d2me49ffcfc18852692@mail.gmail.com>
	<y2m367751821004301526r17a57e70o45ec9ba275539095@mail.gmail.com>
Message-ID: <4BDB6FA4.10606@todandlorna.com>

Phil (Or do you prefer Phillip?),

You will need to probably draw this out a bit.  What is the path by 
which the home office phones connect to the FS boxes?  What path do the 
cell phones take?  What routes do you have for the return data on each 
box?  Are any NATs/Firewalls involved?

I think this is firmly a networking question.  The proxy might come into 
play as well here.  The 3G networks are just like connecting wirelessly 
to the internet.  Depending on your provider, your phone gets an 
internet addressable address, or just an internal network address which 
is NATed before going to the internet.  That NAT can be tested, if you 
have the right tools on your phone and what you're connecting to, but 
that will change from provider to provider and maybe even day to day, 
location to location.

That should be enough to start thinking about the problem and where it 
might lay.

Cheers,

Tod Hansmann

On 4/30/2010 4:26 PM, Phillip Jones wrote:
> Ram,
>
> I think thats what I am asking? I am not sure how the 3G network 
> works. Whether there are restrictions on how servers can communicate 
> to clients etc.
>
> Perhaps I am just way of base also. I don't know. TGIF.
>
> Pj
>
>
> On Fri, Apr 30, 2010 at 1:11 AM, ram <talk2ram at gmail.com 
> <mailto:talk2ram at gmail.com>> wrote:
>
>     Hi
>     why not its possible
>     Ram
>
>     On Fri, Apr 30, 2010 at 2:26 AM, Phillip Jones
>     <pjintheusa at gmail.com <mailto:pjintheusa at gmail.com>> wrote:
>
>         Hi there,
>
>         I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS
>         proxy. Outbound (terminating) traffic goes directly from each
>         FS box.
>
>         All my home office phones get calls no matter which box they
>         are registered with or which box the call comes in on - thanks
>         Anthony.
>
>         However my SIP client on various iPhone/Androids etc only
>         receive calls that originate on the box on which they are
>         registered.
>
>         Looking at the SIP trace - when the call comes in on the
>         'wrong' box, the invites to these SIP clients do not even get
>         a response. Presumably because the 3G network has no idea who
>         this new IP in the "from address" is, who trying to contact them.
>
>         Question is, is there a way around this - our will I have
>         start routing all terminating traffic out through the proxy also.
>
>         Any insight appreciated.
>
>         Thanks!
>
>         Pj
>
>
>         _______________________________________________
>         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 <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 anthony.minessale at gmail.com  Fri Apr 30 17:36:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Apr 2010 19:36:00 -0500
Subject: [Freeswitch-users] libpri not working
In-Reply-To: <4BDB226C.8050608@cachecomm.com>
References: <4BDB226C.8050608@cachecomm.com>
Message-ID: <h2k191c3a031004301736o85d7d614ra8e735eeffe1e79c@mail.gmail.com>

if you are using ozmod_isdn you are not using libpri
you need ozmod_libpri compiled against libpri it's the difference between
pri_spans and libpri_spans



On Fri, Apr 30, 2010 at 1:33 PM, Tom <tomb at cachecomm.com> wrote:

> Voip gods,
> having problem with libpri on incoming call it gives me this [WARNING]
> ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored)
> then it wait a few seconds then i get a fast busy. If i turn off libpri
> i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no
> matching channel 0
>
> Iam running
> FreeSWITCH Version 1.0.head (svn-17188M)
> wanpipe-3.5.10.14
> dahdi-2.3.0
> Sangoma B601de card
>
>
> Tom Baldwin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From troy at tlainvestments.com  Fri Apr 30 18:21:49 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 30 Apr 2010 18:21:49 -0700
Subject: [Freeswitch-users] Picking up voicemail
In-Reply-To: <EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
References: <D1272E93BDF64501BC9A699449E659F5@bp1.ad.bp.com>
	<i2i87f2f3b91004291241h517b24d6hd3c7085f8b583fe0@mail.gmail.com>
	<EA8BCF607C704E82920147E7E192487F@bp1.ad.bp.com>
Message-ID: <3FD20123-7D37-49D2-BF00-50D8FE7D7556@tlainvestments.com>

In voicemail.conf.xml, there are vmain-key and vmain-extension params.  Set the vmail-extension to something in your dial plan that drops you into voicemail main, and you're good to go.

In voicemail.conf.xml:

      <param name="vmain-key" value="9"/>
      <param name="vmain-extension" value="vmain"/>

In default.xml:

    <extension name="vmain">
      <condition field="destination_number" expression="^vmain$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name}"/>  
      </condition>
    </extension>

Then, pressing 9 while in the voicemail menu would prompt you to enter the extension you want to check, then password, etc...

-Troy


On Apr 30, 2010, at 7:02 AM, Dave Stevenson wrote:

> Hi Mike,
>  
> thanks a lot for the reply. I would still like to be able to get into the rest of the system, i.e., other mailboxes than the default, so if someone can shed any more light on this, that would be great.
>  
> In the meantime though, you are absolutely right on the default behaviour! Hitting "0" when the outgoing message is being played does get you to the default mailbox. I was not aware of this and there seems to be another problem which has muddied the waters for me. When I test various bit and pieces coming in from the PSTN line, I have been using my mobile phone (cell phone to you guys), the fact that my cell phone does not behave the same way as a land line seems to have been the problem.
>  
> It seems that hitting a key on the cell phone is picked up by FreeSWITCH to drop you into the voice mail system, but entering the "#" to enter the mailbox password isn't.
>  
> However, when I do the same thing from a land-line (actually, a VOIP phone from the office) the mailbox password is correctly recognised. The problem seems to be recognising the "#" entered through a cell phone. Are you aware of any issues with cell phone tones being recognised by FreeSWITCH ?
>  
> regards
> Dave
>  
> ----- Original Message -----
> From: Michael Collins
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, April 29, 2010 8:41 PM
> Subject: Re: [Freeswitch-users] Picking up voicemail
> 
> 
> 
> On Thu, Apr 29, 2010 at 11:15 AM, Dave Stevenson <stevendt at primrosebank.net> wrote:
> Hi,
>  
> I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good.
>  
> I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension.
> 
> I think by default you can dial 0 while listening to the voicemail outgoing message to log into the voicemail, however I don't believe there's a default way of turning the vm login into a gateway to the rest of the system. Anyone?
> -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 grsingh750 at gmail.com  Fri Apr 30 19:20:59 2010
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 1 May 2010 07:50:59 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 347
In-Reply-To: <mailman.2546.1272641691.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.2546.1272641691.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <z2g7b463d951004301920k5fc7f24aif66b0f997a71db6@mail.gmail.com>

>
> From:?"Dave Stevenson" <stevendt at primrosebank.net>
> To:?<freeswitch-users at lists.freeswitch.org>
> Date:?Fri, 30 Apr 2010 15:53:26 +0100
> Subject:?Re: [Freeswitch-users] Picking up voicemail
> Hi?Brian,
>
> thanks for the pointer.
>
> The SPA-3012 is pretty strong on configuration options, but the documentation is very light on what they actually mean. Just so that I know, what is AVT actually doing ?
>
> regards
> Dave
DTMF Tx Method               Select the method to transmit DTMF
signals to the far end:
                                           InBand, AVT, INFO, Auto,
InBand+INFO, or AVT+INFO.
                                           InBand sends DTMF using the
audio path. AVT sends
                                           DTMF as eypents. INFO uses
the SIP INFO method. Auto
                                           uses InBand or AVT based on
the outcome of codec negotiation.
                                           The default is Auto.

Shamelessly pasted from
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf.
I was reading this manual when I saw your post. You should check it
out if you haven't already.

gs


From tomb at cachecomm.com  Fri Apr 30 19:29:18 2010
From: tomb at cachecomm.com (Tom Baldwin)
Date: Fri, 30 Apr 2010 20:29:18 -0600
Subject: [Freeswitch-users] libpri not working [solved]
In-Reply-To: <4BDB226C.8050608@cachecomm.com>
References: <4BDB226C.8050608@cachecomm.com>
Message-ID: <1272680958.2090.26.camel@tomb-laptop>

I was trying to show that i tried both libpri and openzap's pri stack. 
But not at the same time, sorry about the confusion. I found my problem
now it works great thanks for your time. I have a fractional PRI that
start on channel 16 to 23 . Openzap had set them starting at 1:1 to 1:8
so the call was trying to go out on channels one through eight on the
PRI . 

Fri, 2010-04-30 at 12:33 -0600, Tom wrote:
> Voip gods,
> having problem with libpri on incoming call it gives me this [WARNING] 
> ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored)
> then it wait a few seconds then i get a fast busy. If i turn off libpri 
> i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no 
> matching channel 0
> 
> Iam running
> FreeSWITCH Version 1.0.head (svn-17188M)
> wanpipe-3.5.10.14
> dahdi-2.3.0
> Sangoma B601de card
> 
> 
> Tom Baldwin
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From telteclistas at gmail.com  Fri Apr 30 20:21:17 2010
From: telteclistas at gmail.com (leonardo alves)
Date: Fri, 30 Apr 2010 23:21:17 -0400
Subject: [Freeswitch-users] Error with media ptime
In-Reply-To: <v2q191c3a031004280743vf64121f6u4351356a1ac5659e@mail.gmail.com>
References: <s2naa768f581004272027m15a62473y9defe9764f48ac18@mail.gmail.com>
	<p2haa768f581004272119qe1b44579yb07c2d0d63e895ed@mail.gmail.com>
	<v2q191c3a031004280743vf64121f6u4351356a1ac5659e@mail.gmail.com>
Message-ID: <m2vaa768f581004302021gc0f966f3zdb279ca860232208@mail.gmail.com>

Actually,

I tougth it was working but it is not.
If when I answer the call and stay in silence everything works ok. But if I
say something or there is any noise in the room shows in the freeswitch
console the [WARNING] mod_sofia.c:999 and the call starts to get choppy.

The command that I am running in the freeswitch console is this:
bgapi originate
{ignore_early_media=true,originate_timeout=60}sofia/gateway/nettophone/18197713136
&playback(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-sample_submenu.wav)

Reading the debug of the call what I could understand is while the ptime of
the call is in 20ms everything works good, but when it changes to 30ms the
call starts to get choppy. And it gets changed after the warning.

 On the file vars.xml I also tried this configuration:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i"/>
But after the warning it still changes the ptime to 30 and the call starts
to get choppy.

Does anyone knows what else I could do ? Thanks for any ideas.

Here is the debug of the call:
2010-04-30 21:40:09.656257 [DEBUG] mod_sofia.c:140
sofia/external/18197713136 SOFIA ROUTING
2010-04-30 21:40:09.656257 [DEBUG] switch_ivr_originate.c:66
(sofia/external/18197713136) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-04-30 21:40:09.656257 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/18197713136 [BREAK]
2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:341
(sofia/external/18197713136) State ROUTING going to sleep
2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:314
(sofia/external/18197713136) Running State Change CS_CONSUME_MEDIA
2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:360
(sofia/external/18197713136) State CONSUME_MEDIA
2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:360
(sofia/external/18197713136) State CONSUME_MEDIA going to sleep
send 1085 bytes to udp/[66.33.157.119]:5060 at 02:40:09.657682:
   ------------------------------------------------------------------------
   INVITE sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>SIP/2.0
   Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKBB4KpNNHQUFyS
   Max-Forwards: 70
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>>
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222468 INVITE
   Contact: <sip:gw+nettophone at 10.1.1.1:5080;transport=udp;gw=nettophone>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 270
   X-FS-Support: update_display
   Remote-Party-ID:
<sip:0000000000 at 66.33.157.119<sip%3A0000000000 at 66.33.157.119>
>;party=calling;screen=yes;privacy=off
   v=0
   o=FreeSWITCH 1272653081 1272653082 IN IP4 10.1.1.1
   s=FreeSWITCH
   c=IN IP4 10.1.1.1
   t=0 0
   m=audio 28528 RTP/AVP 0 8 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
2010-04-30 21:40:09.656257 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [calling][0]
recv 411 bytes from udp/[66.33.157.119]:5060 at 02:40:09.703164:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.1.1:5080
;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222468 INVITE
   Contact: <sip:66.33.157.119:5060>
   Server: Net2Phone Carrier
   Content-Length: 0
   ------------------------------------------------------------------------
recv 657 bytes from udp/[66.33.157.119]:5060 at 02:40:11.008062:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.1.1.1:5080
;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222468 INVITE
   Contact: <sip:66.33.157.119:5060>
   Server: Net2Phone Carrier
   Content-Length: 203
   Content-Type: application/sdp
   v=0
   o=44952 713620800 713620800 IN IP4 169.132.188.43
   s=SIP Call
   c=IN IP4 169.132.188.43
   t=0 0
   m=audio 22696 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-11
   ------------------------------------------------------------------------
2010-04-30 21:40:11.008120 [INFO] sofia.c:662 Update Callee ID to
"18197713136" <18197713136>
2010-04-30 21:40:11.008120 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [proceeding][183]
2010-04-30 21:40:11.008120 [DEBUG] sofia.c:4183 Remote SDP:
v=0
o=44952 713620800 713620800 IN IP4 169.132.188.43
s=SIP Call
c=IN IP4 169.132.188.43
t=0 0
m=audio 22696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:3674 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:2372 Set Codec
sofia/external/18197713136 PCMU/8000 20 ms 160 samples
2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:3607 Set 2833 dtmf send
payload to 101
2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:2612 AUDIO RTP
[sofia/external/18197713136] 10.1.1.1 port 28528 -> 169.132.188.43 port
22696 codec: 0 ms: 20
2010-04-30 21:40:11.008120 [DEBUG] switch_rtp.c:1346 Starting timer [soft]
160 bytes per 20ms
2010-04-30 21:40:11.012070 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
payload to 101
2010-04-30 21:40:11.012070 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive
payload to 101
2010-04-30 21:40:11.012070 [NOTICE] sofia_glue.c:3227 Pre-Answer
sofia/external/18197713136!
recv 681 bytes from udp/[66.33.157.119]:5060 at 02:40:18.227385:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.1.1:5080
;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Allow: ACK,BYE,CANCEL,INVITE,OPTIONS
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222468 INVITE
   Contact: <sip:66.33.157.119:5060>
   Server: Net2Phone Carrier
   Content-Length: 203
   Content-Type: application/sdp
   v=0
   o=44952 713620800 713620800 IN IP4 169.132.188.43
   s=SIP Call
   c=IN IP4 169.132.188.43
   t=0 0
   m=audio 22696 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-11
   ------------------------------------------------------------------------
2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [completing][200]
2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4180 Duplicate SDP
v=0
o=44952 713620800 713620800 IN IP4 169.132.188.43
s=SIP Call
c=IN IP4 169.132.188.43
t=0 0
m=audio 22696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
send 429 bytes to udp/[66.33.157.119]:5060 at 02:40:18.228604:
   ------------------------------------------------------------------------
   ACK sip:66.33.157.119:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKcmXcrg6mm45gN
   Max-Forwards: 70
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222468 ACK
   Contact: <sip:gw+nettophone at 10.1.1.1:5080;transport=udp;gw=nettophone>
   Content-Length: 0
   ------------------------------------------------------------------------
2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [ready][200]
2010-04-30 21:40:18.227023 [NOTICE] sofia.c:4696 Channel
[sofia/external/18197713136] has been answered
2010-04-30 21:40:18.228967 [DEBUG] switch_ivr_originate.c:3210 Originate
Resulted in Success: [sofia/external/18197713136]
2010-04-30 21:40:18.228967 [DEBUG] mod_commands.c:2870
(sofia/external/18197713136) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2010-04-30 21:40:18.228967 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/18197713136 [BREAK]
2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:314
(sofia/external/18197713136) Running State Change CS_EXECUTE
2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:348
(sofia/external/18197713136) State EXECUTE
2010-04-30 21:40:18.228967 [DEBUG] mod_sofia.c:233
sofia/external/18197713136 SOFIA EXECUTE
2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:157
sofia/external/18197713136 Standard EXECUTE
EXECUTE sofia/external/18197713136
playback(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-sample_submenu.wav)
2010-04-30 21:40:18.229973 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-04-30 21:40:18.287128 [DEBUG] switch_rtp.c:2446 Correct ip/port
confirmed.
2010-04-30 21:40:18.486993 [WARNING] mod_sofia.c:999 We were told to use
ptime 20 but what they meant to say was 30
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who
knows what will happen..
2010-04-30 21:40:18.486993 [DEBUG] sofia_glue.c:2301 Changing Codec from
PCMU at 20ms to PCMU at 30ms
2010-04-30 21:40:18.486993 [DEBUG] switch_rtp.c:1232 RE-Starting timer
[soft] 240 bytes per 30000ms
2010-04-30 21:40:18.486993 [DEBUG] sofia_glue.c:2372 Set Codec
sofia/external/18197713136 PCMU/8000 30 ms 240 samples
send 1081 bytes to udp/[66.33.157.119]:5060 at 02:40:18.487972:
   ------------------------------------------------------------------------
   INVITE sip:66.33.157.119:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKDXp5SBQrHDv3g
   Max-Forwards: 70
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222469 INVITE
   Contact: <sip:gw+nettophone at 10.1.1.1:5080;transport=udp;gw=nettophone>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-Broken-PTIME: Adv=20;Sent=30
   X-FS-Support: update_display
   Remote-Party-ID:
<sip:0000000000 at 66.33.157.119<sip%3A0000000000 at 66.33.157.119>
>;party=calling;screen=yes;privacy=off
   v=0
   o=FreeSWITCH 1272653081 1272653083 IN IP4 10.1.1.1
   s=FreeSWITCH
   c=IN IP4 10.1.1.1
   t=0 0
   m=audio 28528 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:30
   ------------------------------------------------------------------------
2010-04-30 21:40:18.486993 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [calling][0]
2010-04-30 21:40:18.517050 [DEBUG] switch_core_io.c:896 Engaging Write
Buffer at 480 bytes to accommodate 320->480
recv 681 bytes from udp/[66.33.157.119]:5060 at 02:40:18.534493:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.1.1:5080
;branch=z9hG4bKDXp5SBQrHDv3g;received=10.1.1.1;rport=5080
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Allow: ACK,BYE,CANCEL,INVITE,OPTIONS
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222469 INVITE
   Contact: <sip:66.33.157.119:5060>
   Server: Net2Phone Carrier
   Content-Length: 203
   Content-Type: application/sdp
   v=0
   o=44952 713620800 713620800 IN IP4 169.132.188.43
   s=SIP Call
   c=IN IP4 169.132.188.43
   t=0 0
   m=audio 22696 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-11
   ------------------------------------------------------------------------
send 429 bytes to udp/[66.33.157.119]:5060 at 02:40:18.534747:
   ------------------------------------------------------------------------
   ACK sip:66.33.157.119:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKe6FyU67Uepjpc
   Max-Forwards: 70
   From: "" <sip:cust_USERNAME at 66.33.157.119<sip%3Acust_USERNAME at 66.33.157.119>
;transport=udp>;tag=BBvj27gt2ZFQB
   To: <sip:18197713136 at 66.33.157.119 <sip%3A18197713136 at 66.33.157.119>
>;tag=ccid-713620800-1-574
   Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8
   CSeq: 130222469 ACK
   Contact: <sip:gw+nettophone at 10.1.1.1:5080;transport=udp;gw=nettophone>
   Content-Length: 0
   ------------------------------------------------------------------------
2010-04-30 21:40:18.534975 [DEBUG] sofia.c:4172 Channel
sofia/external/18197713136 entering state [ready][200]
2010-04-30 21:40:18.534975 [DEBUG] sofia.c:4180 Duplicate SDP
v=0
o=44952 713620800 713620800 IN IP4 169.132.188.43
s=SIP Call
c=IN IP4 169.132.188.43
t=0 0
m=audio 22696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
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From babak.freeswitch at gmail.com  Fri Apr 30 22:53:45 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 1 May 2010 10:23:45 +0430
Subject: [Freeswitch-users] run error after building in vs 2008!
Message-ID: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>

Hi
freeswitch is built without any problems but when running it gives the error
about "entry point inet_ntop could not be located in dynamic link library
WS2_32.dll"
I'm using xp sp3.
Thanks
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From mark.maly at molcs.org  Fri Apr 30 19:07:00 2010
From: mark.maly at molcs.org (Mark Maly)
Date: Fri, 30 Apr 2010 21:07:00 -0500
Subject: [Freeswitch-users] Aastra and SCA
Message-ID: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>

Hi,

 

I've tried to patiently figure this out by reading the wiki and this list.
Unfortunately, I've been unable to get it right.  

 

I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I'm
experiencing problems similar to the Cisco thread from last month - outgoing
calls implement SCA well.  Incoming calls ring all lines and appearances
work, but when one phone is answered, the line appearances are removed from
the remaining phones.  I am not attempting to use any DNS.

 

My configuration has all three phones plus FS on a local LAN.  Nothing too
fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
enabled globally for the phones.

 

Trying to update/replace an old phone at my church. 

 

Any help would be greatly appreciated.

 

Thanks, Mark

Mark.maly at molcs.org

 

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