September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with Freeswitch/Sangoma D100.
by Ricardo Martinez 24 Sep '11

24 Sep '11

Hello list. I have the next problem with my Freeswitch. I’m using a Sangoma D100 transcoding card, but ‘im having problems with G729. This is my scenario. GatewayA ---------> FreeSwitch+D100 -----------> GatewayB (10.0.0.220) (10.0.0.148) (10.0.0.222) Gateway A is calling through Freeswitch to Gateway B. The problem is with the “200 - OK” message that goes from FreeSwitch to Gateway A. The “200 - OK” coming from Gateway B has the parameter “a=fmtp:18 annexb=no” in the SDP, but Freeswitch is not attaching this parameter to the “Leg A” and this is causing a one-way audio in the call. (please see the attached sdp-problem.jpg file). The weird thing is when i unload the “mod_sangoma_codec” and load the “mod_g729”, this time the “200 OK” message from the Freeswitch is using the “a=fmtp:18 annexb=no” to the Leg A, and the call is successfully established without one way audio problem. I really don’t know what could be happening, I ask the Sangoma support but thay said that this is a Freeswitch bug. Can someone help me here? These are part of my configuration files. “default.xml” <extension name="to_prueba2"> <condition field="network_addr" expression="^10\.0\.0\.220$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="nolocal:absolute_codec_string=G729,G723"/> <action application="set" data="sip_invite_domain=10.0.0.222"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:18 G729/8000,a=fmtp:18 annexb=no"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}(a)10.0.0.222"/> <action application="answer"/> </condition> </extension> <extension name="to_prueba1"> <condition field="network_addr" expression="^10\.0\.0\.222$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=10.0.0.220"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}(a)10.0.0.220"/> <action application="answer"/> </condition> </extension> Part of the “interior.xml” file:  <param name="inbound-late-negotiation" value="true"/>  <param name="inbound-codec-negotiation" value="generous"/> Y finalmente en los codec preferentes tengo (vars.xml) <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G723,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,G723"/> Thanks in advance. Regards, Ricardo.-

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Web based reporting tools
by Joseph L. Casale 23 Sep '11

23 Sep '11

What are you guys using that can produce good reports for current and accumulated statistics on an FS box? Thanks! jlc

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Freeswitch Does Not Build
by Antonio Teixeira 23 Sep '11

23 Sep '11

At the time Date: Thu Sep 22 21:11:43 2011 +0200 Freeswitch fails with make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp' make[6]: Entering directory `/usr/local/src/freeswitch/libs/unimrcp' make[6]: *** No rule to make target `build/acmacros/swift.m4', needed by `Makefile.in'. Stop. make[6]: Leaving directory `/usr/local/src/freeswitch/libs/unimrcp' make[5]: *** [/usr/local/src/freeswitch/libs/unimrcp/platforms/libunimrcp-client/ libunimrcpclient.la] Error 2 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp' make[4]: *** [mod_unimrcp-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Or is just me ? Regards A/T

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Nibble Bill Issue
by Ahmed Bhaila 23 Sep '11

23 Sep '11

Hi I am trying to run nibble bill in our test environment and unable to see calls hanging up when the balance depletes to 0. Following is the xml from dialplan: <context name="public"> <extension name="Internal-XXX_Mobile"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="hangup_after_bridge=true" inline="true"/> <action application="set" data="continue_on_fail=true" inline="true"/> <action application="bridge" data="[nibble_account=1,nibble_rate=1.458]sofia/gateway/NEXTONE/1291892xxxxxxxxx"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="log" data="NOTICE Hanging up"/> <action application="hangup"/> </condition> </extension> </context> Now the balance for this account is already below zero and interestingly the A leg hangs up while B leg stays connected. This is what I am seeing on my console Transfer sofia/external/1291892xxxxxxxxx to XML[hangup@public] So the call does get transferred to the hangup extension but only manages to hangup the A leg. Thanks Ahmed

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via header
by Chad Vogel 23 Sep '11

23 Sep '11

Hello, does anyone know why fs isn't setting rport in the sip via header? is there a way to set this? Thanks Chad

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callerid name on a gateway
by Stanisław Pitucha 22 Sep '11

22 Sep '11

Hi all, I've got a gateway which I contact with: caller-id-in-from = false I've got the from-user set to the number (remote username), let's say 1234567. Unfortunately when I send the call, the invite has header: From: "" <1234567@....> The gateway doesn't like it very much (rejects the call, complaining about the from header) for some reason and I'd like to either fill in the from name with the username, or remove the from name completely. Is there some way to do this? -- KTHXBYE, Stanisław Pitucha

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Share PRI line?
by fssupport＠highergear.com 22 Sep '11

22 Sep '11

We have a client with a Mitel PBX with a PRI connected to it. We are deploying a freeswitch box that will do some autodialing tasks for them. Is there a way to share the PRI between the two systems? If nothing else, I'm pretty sure I can put a two line interface card in the Freeswitch server, plug the pri into it, then connected the other interface to the Mitel box, but that would make our box a point of failure and I'd like to avoid that if possible.

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Weekly FreeSwitch conference Call
by Ken Rice 22 Sep '11

22 Sep '11

Hey Guys, Michael and Brian have asked me to help out with the conference today since Michael is otherwise occupied today... Open Forum for the most part today, however, we will have some discussion of hotels for Cluecon 2012, and some more embedded talk. Plus more.... Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today! Just dial *9888 from your FreeSwitch installs that have the default dialplan or join as via the web at http://conference.freeswitch.org/ K

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Codec byte order
by Fabio Bigliardi 22 Sep '11

22 Sep '11

Hi all, I have an audio decoder registered to freeswitch server. It worked with a codec L16@24000h little endian. After software update to FreeSWITCH Version 1.0.head (git-25c725c 2011-06-30 18-30-24 -0500), the audio decoder ceased to work with this codec. It produces just noise. With the big endian version of the codec, it works again. The hardware on which freeswitch is running, is the same. How could you explain this behaviour? Thank you very much. Fabio Bigliardi

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last git and postgresql 9.1
by Madovsky 22 Sep '11

22 Sep '11

Hi Folks, only to say that I just upgraded to PG 9.1 and everything is working well Cheers Franck

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