May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Codec Negotiation Help
by Spencer Thomason 08 May '13

08 May '13

Hello all, I'm having a bit of trouble getting codec negotiation working the way I'd like to. I'm trying to avoid transcoding unless its absolutely necessary for the call to be setup. I use two profiles with late negotiation enabled, an internal facing registered user endpoints and an external facing the PSTN via various ITSPs. For calls to users I've been setting codec_string=${ep_codec_string} in my dial strings and normal bridges work fine but I'm now reworking the hunt groups using enterprise originate and that doesn't seem to work correctly (the callee's preferences are always used). So I'd like to do the the "right" way now :-) The basic scenario is this: On the internal profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM On the external profile I have: CODECS IN G722,PCMU,G729 CODECS OUT G722,PCMU,G729 What I'm trying to accomplish is this: Call from PSTN, use caller's codecs as priority for the bridge, i.e. PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local Endpoint PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local Endpoint PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local Endpoint Call from local endpoint to PSTN, use callee's codecs as priority for the bridge, i.e. Local Endpoint Offers G722, PCMU, G729 GW Responds PCMU,G729 Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> PSTN Desired negotiation: Both legs use PCMU Local Endpoint Offers G722, PCMU, G729 GW Responds G729,PCMU Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Both legs use G729 Local Endpoint Offers GSM GW Responds PCMU, G729 Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH transcodes. Any help is greatly appreciated! Thanks, Spencer

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How to modify the way freeswitch handle SIP OPTIONS message
by Iwan Budi Kusnanto 08 May '13

08 May '13

Hi, I can see from console log that freeswitch always reply with SIP 200 when receive SIP OPTIONS message. Can i modify this behaviour? How to modify it? Thanks -- Iwan Budi Kusnanto

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Do Freeswitch conference support silent suppression?
by nordin mohamed 07 May '13

07 May '13

Currently I'm trying to create conference server using Freeswitch and hook up with the ROIP device (Vocality). The ROIP device is act as the voip extension for the Freeswitch, In order for the radio link to work, the conference must support the VAD/CNG so that the silence is not transmitted by the ROIP device to its radios.My problem is the ROIP will always transmit something on the radio. Am I correct to conclude that the Freeswitch not supporting silent suppression while in the conference mode.? thank you -nordin mohamed

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Request for review - FS-5388
by Cal Leeming [Simplicity Media Ltd] 07 May '13

07 May '13

Hello! Could a core dev spend a minute or two reviewing the following ticket please; http://jira.freeswitch.org/browse/FS-5388 In return, I have triaged the following ticket; http://jira.freeswitch.org/browse/FS-5168 I spent 10 minutes looking for other tickets I could help with, but all of them were beyond my skill level :/ Cal

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settings for a Trixbox v 2.6 connection to a Sip Trunking translate to gateway using Freeswitch 1.2.0
by Richard Cruz 07 May '13

07 May '13

Hi There! I have the following settings for a Trixbox v 2.6 connection to a Sip Trunking at tricom.net Host=XXX.X.132.37 defaultuser=********** Secret= *********** Fromuser=xxxxxxxxxx Type=friend insecure=invite,port Dtmfmode=rfc2833 Disallow=all Allow=g729 Canredirect=no Canreivite=no Context=from-trunk Qualify=yes and I want to translate to gateway using Freeswitch 1.2.0 -- Richard Cruz 678.394-6400

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No ring-back with 180 ringing and P-Early-Media tag
by Victor Chukalovskiy 07 May '13

07 May '13

Hello, Can FreeSWITCH be configured not to include P-Early-Media tag in it's 180 ringing? Is any of the boxes involved breaking the specs? Call scenario is that far-end SBC sends INVITE to FS with /P-Early-Media: supported/ tag. Freeswitch routes call and replies with 180 Ringing with /P-Early-Media: sendonly/ tag to the far-end SBC. Far-end does not generate ringback when it sees P-Early-Media: sendonly. Freswitch does not provide in-band ring-back either (cause it's just 180 Ringing without SDP). Last but not least, FreeSWITCH runs in bypass media mode. Thank you, Victor

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Can anyone out there look at this..
by Blake Priddy 07 May '13

07 May '13

I have a pcap capture from a phone that experiences the delay. Attached is the pcap where the user had a 7 sec delay. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Problem with pre-answer
by svb 07 May '13

07 May '13

Hello Freeswitch Users, I’ve got a problem with playing announcements in FreeSWITCH last stable Version 1.2.8. To be precise, when using ‘session::streamFile’ in a lua-script, I get the following error response: [ERR] sofia_glue.c:3015 No audio codec available The error only occurs when I set <action application="pre_answer"/> in the dialplan. On the other hand, if I do use <action application="answer"/> instead, a codeclist is generated and the announcement is played. The sip-profile has the following settings: <param name="global-codec-prefs" value="$${global_codec_prefs}"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="early-media-override" value="on-progress"/> <param name="inbound-codec-negotiation" value="generous"/> vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=G722,PCMU,PCMA,GSM"/> Furthermore the following variables are set int the dialplan: <action application="set" data="ignore_early_media=false"/> <action application="set" data="bypass_media=false"/> <action application="set" data="proxy_media=false"/> <action application="set" data="instant_ringback=false"/> <action application="set" data="execute_on_answer=sched_hangup +10800 alloted_timeout"/> <action application="set" data="call_timeout=120"/> <action application="set" data="uuid_bridge_continue_on_cancel=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="set_zombie_exec"/> Compairing the logfiles of an ‘answered’ with an ‘pre-answered’ call I found the following Variables are NOT set in the ‘pre-answered’ Version: Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] I’ve tried all the above mentioned combinations that I could think of, but without any luck. Anyone got a bright idea what actually is the matter and would kindly help me out? Regards, Sven

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urgent!!!! FS crashes on inbound calls
by Ashish gautam 07 May '13

07 May '13

Hi, I am running freeswitch on FreeBSD system with dahdi-libpri-freetdm stack. Here a serious issue is coming. Whenever I make an inbound call, FreeSWITCH just crashes with no errors in logs. I am attaching the part of freeswitch.log for reference. Please throw some light. Regards, Ashish

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ZRTP issue being hard to track
by Daniel Ivanov 07 May '13

07 May '13

I make currented the production fs a few weeks ago, because i've been unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup ). Trusted mitm works, but once i enable proxy-media the UAs ignore the zrtp-hash. One of them always says 'other side doesnt seem to support zrto'. I talked on irc to steevenielson to ask him any hints and he pointed the only difference between me and ostel.me setup is the rather old version he has. Could you elaborate with me on this?

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