January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Porting
by Blake Priddy 29 Jan '15

29 Jan '15

Hi all! I was wondering what you have experienced with flowroute and their porting? I love their service but it seems when I try to port a number over to them from someone who wants to get away from their current provider. Flowroute doesn't service the area.. Is there a provider I can port a number to then have flowroute get the number from them? Like a proxy somewhat. I have schools that are wanting to switch to freeswitch and I always have to do remote call forwarding on their line. So they pay flowroute and still their current provider for the RCF feature. Just trying to get advice on what you all have had experiences with and gather any ideas if I can. Thanks! -- *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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APR issue when configuring
by Grant Bagdasarian 29 Jan '15

29 Jan '15

Hello, I'm getting the following error during configure. checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. configure: error: ./configure.gnu failed for libs/apr-util I'm running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. Is there a way to disable apr? Or is it required by FS? Regards, Grant

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Cannot contact registered endpoint behind NAT since latest make current
by Emrah 28 Jan '15

28 Jan '15

Hi all, Since my make current of this week, I cannot bridge to registered endpoints that are seen by my FS as devices behind NAT. I tried bridging both with user/aurora605@domain as well as using the full sofia_contact value. Here is the output I get: Cannot locate registered user aurora605@192.168.2.191;received=1.2.3.4:34146;fs_nat=yes;fs_path=sip@3Aaurora605%401.2.3.4%3A34146 this is the output of sofia_contact sofia/internal/sip:aurora605@192.168.2.191;received=1.2.3.4:1029;fs_nat=yes;fs_path=sip%3Aaurora605%401.2.3.4%3A1029 What are your suggestions to fix this issue? Best, Emrah

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Audio levels between webRTC and Softphone very different
by Thomas Auge 28 Jan '15

28 Jan '15

We're seeing an issue where the audio level from our webRTC app when connecting to softphones (and some hardware devices) via SIP though freeswitch arrives much higher at the other end than it should be. The other way around the levels from the phone to webRTC are way too low. Does anyone have an idea what could be causing this?

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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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