- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Bind to multiple ports with Sofia
by Emrah 07 May '13

07 May '13

Hi guys, I need to bind to multiple ports with Sofia. If I start one profile per port, I cannot get a consistent registration table working. A domain can only be aliased to one profile if my understanding is correct. If you have a working set up, I'd appreciate some advice. Thanks, Emrah

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FreeSWITCH Weekly News and Notes
by Michael Collins 06 May '13

06 May '13

Happy first May Monday to all. We are please to let everyone know that the new FreeSWITCH book<http://www.packtpub.com/freeswitch-1-2/book>is nearing completion. Each day the Packt editors have been sending us revised and re-revised chapters to polish off. If we keep up our current pace then it is quite likely the book will be available in early June. We hope you enjoy it. On last week's call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_01>we enjoyed a nice presentation from Omar over at OrecX <http://www.orecx.com> who showed us some of the useful features of their call recording solutions. They have an open source version as well as commercial versions. If you are looking for a call recording solution then we recommend that you review what OrecX has to offer. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_08> we continue in our series of presentations on FreeSWITCH-compatible software applications. We look forward to having Dan Bogos from CGRateS<http://www.cgrates.org/>join us to talk about their project. Click here <http://cgrates.readthedocs.org/en/latest/introduction.html> to get a quick look at what CGRateS does, and then make plans to join our call on Wednesday. Regarding ClueCon 2013 we would like to remind everyone that we are still accepting talk proposals. If you have a talk idea please let us know. In the meantime feel free to get registered <http://cluecon.com/register/> and book your hotel room at the Hyatt. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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WiFi IP VOIP SIP phones
by Sean Devoy 06 May '13

06 May '13

HI, Can anyone recommend a good quality WiFi (wireless IP) VOIP Phone that works with FS? Looking for < $200 and top quality functionality. Thanks, Sean

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Early Media indication from B in CHANNEL_STATE event
by David Wilkie 06 May '13

06 May '13

Hi, Is there a way to get an early media indication from a B-leg in the CHANNEL_STATE event. According to http://wiki.freeswitch.org/wiki/Event_list#CHANNEL_STATE there doesn't seem to be a variable which indicates early media from B. We need to use this information to figure out whether to join B to A or generate early media and join on answer. If there is no way to get this info can we query it using Event Socket?

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