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sip trace
by Andrew 28 Jan '15

28 Jan '15

Can FreeSWITCH export a sip trace like XML CDR? Andrew

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inband dtmf
by huseyin kalyoncu 28 Jan '15

28 Jan '15

hello first i want to thank the developers and contributors of this amazing product. we have been using freeswitch for almost 4 years without a major problem. i have a question regarding inband dtmf. we have receiving calls from telco using dtmf rfc2833. most of outgoing calls also dtmf rfc2833. but we have a new outgoing profile which is behind a firewall. we did not make a successful call with transport UDP. so we set the transport to TCP and now we have successful calls but the only problem is with dtmf. when we set the dtmf to rfc2833 for this profile, we saw that dtmf packets do not arrive correctly to outgoing destination. when we dig up the problem we realized that there is always a time skew on dtmf packets. for this reason we tried to set dtmf inband for this particular outgoing profile. to accomplish this we used start_dtmf_generate just before the bridge action. but this time no dtmf package arrive at destination. is this dtmf conversion (from rfc2833 to inband) even possible? what should be the correct configuration to achieve this? thanks huseyin

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Connecting to FS from a mobile app
by Oleg Stolyar 28 Jan '15

28 Jan '15

Guys, Has anyone done that? I currently have WebRTC clients connecting to FS from Chrome and/or Firefox but I now need to do it from my mobile apps (Android and iOS). Any recommendation on how to do it? I use JsSip in the browsers. Any way to make a JS library like that to work from an app? If not, any other suggestions? WebRTC vs. SIP? WebRTC with SIP vs WebRTC with Verto? I apologize if this topic is not about Freeswitch's own functionality but I thought people would b interested and wanted to pick this group's brain.

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Limit on b-leg
by Артур Краев 27 Jan '15

27 Jan '15

Hello everyone! Could you help please how-to realize limits on b-leg? I already saw a wiki page, and way to outbound limit on gateways but, there's some problems to use this solution in my cases. Case 1: [provider sip] -> [fs] -> [user 1] In this case, I'm need to set call limit for user 1 (it can handle no more than concurrent 1 call, connected device is not managed by me) . If I will set it on a-leg in dialplan, the limit will bound to a-leg's UUID and stay here even if b-leg (which is user 1) will be destroyed, for example if user 1 transferred (of att-xferred) call to another user. Case 2: [provider sip] -> [fs] -> [user 1], [user 2], [user 3] (enterprise dial) In this case, I'm need to set own call limit for each user's channel, only for matching b-leg, not for all. This cases, may be realized with loopback channel (which is not elegant, because produces addition problems with cdrs...) but, what will be if one of this users is fax, and we're need to make T.38 pass-trough between a-leg and b-leg? At this moment there's no way to pass-trough t.38 with loopback channels. Ok, after all of this, I'm tried to some sort of googling and found this issue in FS Jira: https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-1792/FS-1792.h… I was tried to use uuid_limit for setting limit on b-leg, but: - I cannot use uuid-limit on a-leg because, there's no uuid for b-leg before bridge called. I can generate uuid by api function, and set for b-leg as channel variable in bridge, but now I also cannot run uuid_limit on b-leg because there's no b-leg channel before bridge called. - If I'm write a script which handles channel init event, checking some variable, and after this, executes uuid_limit on it, there's some other problem - events working in async way, so user's phone will ring before uuid_limit action will be executed (after that user will have +1 missed call, which is not must be here). So the last way to implement this cases is to write module for FS (C or .Net) which will handle execute_on_init, variable, and executes action when channel initializing. But, maybe there any other way to implement this functionality. Or may be execute_on_init shall be added to FS core? -- With best regards, Arthur Kraev

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[CRIT] mod_event_socket.c:2620 Socket Error!
by Ali Pey 27 Jan '15

27 Jan '15

Hello, What should I be looking at if I get this error time to time? [CRIT] mod_event_socket.c:2620 Socket Error! It seems to be traffic related and happens around 370 channels utilized. Sometimes, it kills the FS server. I have the ulimit settings for the freeswitch process as followings: ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 240 ulimit -l unlimited Any suggestions? Thanks, Ali Pey

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Buy the FreeSWITCH Dev's Dinner!
by Ken Rice 27 Jan '15

27 Jan '15

Hey Guys, The core FreeSWITCH Team is all together for the Annual Engineering and Planning Meeting! Nows your chance to buy them dinner in appreciation for all their hard work! Visit https://freeswitch.org/ and hit the donate button and let them know this is for dinner! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Error whith xml_curl
by Antony Aguirre Morales 27 Jan '15

27 Jan '15

I set my freeswitch with the directory database, now by registering my extensions are connected correctly, but when making a call of the extension extention 1 to 2 me throws the following error "01/27/2015 10: 22: 35.158536 [ERR] mod_dptools.c: 3977 No dial-string available, Please check your user directory." and directly sends the voice mailbox of the extension. Thanks!

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Bug Marshalls Needed!
by Brian West 27 Jan '15

27 Jan '15

FreeSWITCHers, We're currently in need of bug marshalls, The job is basically to triage JIRA's make sure the data is correct, try to replicate the issue if the description is clear enough or ask for more info about the issue. If you're interested in helping out please email krice(a)freeswitch.org or myself. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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lua + aes|md5
by Денис 27 Jan '15

27 Jan '15

Hello. I need module md5 or aes for checksums, but modules from luarocks not work. Who can make luarocks version for freeswitch with 5.2 version? Or system RPM with 5.2 My OS is CentOS 6.

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how to transfer all conference members to an extension on stop-talking event.
by Aqs Younas 27 Jan '15

27 Jan '15

Hi, list. I need to transfer all conference members to some other extension whenever there is any stop-talking event. But don't know how can i do so? I am using python script as inbound connection to listening to events. Whenever there is stop-talking event I need to transfer all conference members to another dialplan extension. Your help would be much appreciated. Thanks.

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