March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8
by Thomas Løcke 31 May '16

31 May '16

Hi all, Using 1.6.2 this works from fs_cli and dialplan: originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. Has something changed between 1.6.2 and 1.6.6 that may be the cause of this, and if so, what can I do to fix it? :o) Thomas

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FreeSWITCH using outbound proxy but to bypass media
by Muhammad Naseer Bhatti 28 May '16

28 May '16

I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It’s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it’s not working. Is this even doable? Thanks. -- Muhammad Naseer Bhatti

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Fax....How I hate you.
by Blake Priddy 24 May '16

24 May '16

Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing.. -- *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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Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

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API break vs uuid_break
by Marek Slivanský - Livispace s.r.o. 10 May '16

10 May '16

Hi there, sorry if I'm not choosing the right way for asking for something, please point me out, this is my first time using the mailing list. I have a question regarding API command "break" has changed to "uuid_break". I really tried to but I've not found any information about that. We are using "bgapi break <uuid>" to stop playing some sound to a call (leg), e.g. moh, and to continue with next dialplan item. We are using v1.2.stable. Probably since last upgrade to the latest commit in this branch (prior we were using some 09/2012 commit), sometimes (rarely) the command "bgapi break <uuid>" gets ignored by FreeSwitch and sound remains playing forever (local_stream://moh). Could anyone please shed some light on this change (and perhaps add some info why break has been deprecated and consequences of continuing to use it, into wiki)? Thanks in advance for any answer, Marek

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module question
by Chris Mandra 13 Apr '16

13 Apr '16

Hey guys - I've been working on a ladspa module and I thought that after I make a change to it and recompile I can just run reload and hear my changes, but it's not working, so I tried unload and reload and that's not allowing me to hear my changes either. Restarting freeSWITCH does allow me to hear my changes. Any ideas why this would be happening. I'm running 1.6 I know about the how the modules shutdown function is supposed to work ( https://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Reloading_You…"N.B. your module's Shutdown function must accomplish a complete reversion of any setup that happens in it's Load function for a reload to work properly.") and this used to work without issue, so I'm wondering what might be going on here. Any suggestions appreciated, thanks, chris

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Mastering FreeSWITCH Book
by Anthony Minessale 09 Apr '16

09 Apr '16

Hi, We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH" Would anyone be interested in working on the authoring of that book and get their name in print? We need help with creating the content as well as formatting the text into the word templates they require etc. Please contact consulting(a)freeswitch.org if you are interested. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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extended FS as ZRTP client
by Assaf Dahary 07 Apr '16

07 Apr '16

Hi, I would like to use FS (multiple) as a ZRTP client register on a the main FS. I have already managed to setup a Gateway with user/pass on the FS client and register it on the main FS for regular incoming/outgoing calls (without ZRTP). To enable ZRTP calls I setup the FS client/main as follow: None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled). FS Client (ver 1.4) : ZRTP enabled globally in VARS and in dialplan. Media proxy disabled on both internal and external profiles and in dialplan. FS Main (ver 1.4): Media proxy enabled - including late negotiation. I forced the FS client and the CSipSimple to use only PCMU codec to avoid transcoding. The problem is that on a call from the SIP phone via the FS client there are always CRC errors on the ZRTP log. Only if the FS main is set to disable media proxy then there are no CRC errors - but then it becomes a MITM with incompatible SASs. >From reading other posts about FS and ZRTP CRC errors I assume that it happens because the FS client is not creating a zrtp hash in the invite SDP. So my question is how to make the FS client to generate the zrtp hash in the invite SDP to act as real ZRTP enabled client? I've already tried to set the FS client internal/external/dialplan with several zrtp configs with no success. I would appreciate any tip to resolve this issue. Regards Assaf The client FS

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Is MySQL OK to use?
by Alex Lake 06 Apr '16

06 Apr '16

I find these "Error in my_thread_global_end()" messages somewhat annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice (other than "don't use MySQL") for how to install it better? Might it be something to do with thread-safe libraries (I'm using libmyodbc_r.so at the moment)

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480 Temporarily Unavailable[MANDATORY_IE_MISSING]
by Rutu Patel 06 Apr '16

06 Apr '16

Hi, We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server. Call flow is as below: User => opensips => FS => PBX, but call not sending out from FS and drop by FS. Here I am attaching both sip logs and freeswitch logs. Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>

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