August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Newbie - External Phone setup question
by Charles Bujold 26 Aug '11

26 Aug '11

Trying to get a phone located in another office registered to the Freeswitch server. I created an "external5090.xml" in the sip external folder as per the wiki and then set the phone to connect to 5090. But I cannot get the phone to register. When I look at wireshark I see a request from the phone to register and then I see an error "SIP:403 Forbidden" from the server to the Phone. I have created the extension in the Directory folder with the other internal extensions but I'm wondering if I'm missing something in the 1010extension.xml for it to work. (maybe the extension is located in the wrong folder, or some identification to say it is not in the local LAN?) I have checked the name and password and they are correct which seems to indicate the issue is at the server level. Any guidance on how to setup an external phone to Freeswitch would be appreciated. Thanks CJB

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Mod_Managed Mono is not following the system LD Path
by R H 26 Aug '11

26 Aug '11

Hey Everyone, I have recently been asked to set up a FreeSwitch server on a company development server and the development management wants to use Mono C# as our language for adding logic to the platform. Following the instructions at http://wiki.freeswitch.org/wiki/Mod_managed I have been able to successfully compile freeswitch on a Suse 11.4 box, with a custom installation of Mono 2.8 and I can *successfully execute Demo.csx*. But here is the problem, the only way I can get Demo.csx to run correctly without getting a NullReferenceException is to start the server from the freeswitch mod directory: EG. $> cd /usr/local/freeswitch/mod/ $> /usr/local/freeswitch/bin/freeswitch * [SUCCESS!, Demo.csx will run correctly]* $> cd / (or any other directory) $> /usr/local/freeswitch/bin/freeswitch [FAIL, The server starts, calls work, everything is fine EXCEPT Demo.csx now throws *NullReferenceExceptions when I hit it with a call that is routed to the script by the dial plan]* - I have tried changing the *dllmap* default to the mono installation, I have tried adding /usr/local/freeswitch/mod/*FreeSWITCH.Native.dll.config*with the following text inside: """ <configuration> <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/> </configuration> """ - I have tried adding /usr/local/freeswitch/mod/ to the system ld path (/etc/ld.so.conf). I also reloaded the ld cache (ldconfig) - I have tried copying FreeSWITCH.Managed.dll to the existing ld path (/usr/local/lib/FreeSWITCH.Managed.dll) - I even created a Console HelloWorld.exe that imports the Freeswitch libraries and ran it by simply executing: $> mono

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Making Freeswitch listen on 2 IP addresses
by Pete Kelly 26 Aug '11

26 Aug '11

Hi Is it possible to make freeswitch listen on more than one IP for requests within the same profile? I would like to assign a VIP to a pair of freeswitch boxes, for the purposes of failover if one of the boxes should fail or go down. The VIP can then be used by web applications which have to send XMLRPC requests to freeswitch. In addition I would like each of the boxes to listen on their "normal" eth0 IP too, so systems like opensips which support load balancing can evenly distribute load across the two boxes. My profile.xml file currently has this entry: <param name="sip-ip" value="$${local_ip_v4}"/> So I can't see how freeswitch can be told to listen on more than one IP for a profile. Is this possible?

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mod_nibblebill mp3
by Pete Kelly 26 Aug '11

26 Aug '11

Hi everyone I am looking at prepaid solutions at the moment, and mod_nibblebill seems pretty interesting. However the mp3 linked to from the wiki page is no longer available: http://wiki.freeswitch.org/wiki/Mod_nibblebill Does anyone have a copy at all? Also, how 'stable' is nibblebill, is it a well used module by people? I'll probably be starting off with large call volumes (rather than slow ramping up) so any ideas of usage statistics or advice on usage would be useful. Thanks Pete

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Re: [Freeswitch-users] H264 packetization with echo app
by Dan-Cristian Bogos 26 Aug '11

26 Aug '11

Hey Anthony, Many thanks for so fast reaction. It works like a charm with your new patches. Have a good one! DanB

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H264 packetization with echo app
by Dan-Cristian Bogos 25 Aug '11

25 Aug '11

Hey Guys, I have recently started playing with the echo application for video calls and found some possible issue (asking here if this is the desired behavior). Scenario: * Call gets connected with echo application, echoing audio just fine. * On reinvite the client sends the following SDP(please note the line with a=fmtp:102 packetization-mode=1): {{{ v=0. o=dan.test 0 1 IN IP4 10.10.10.106. s=-. c=IN IP4 10.10.10.106. t=0 0. m=audio 48576 RTP/AVP 9 96 97 0 8 98 100 5 6 15 101. a=rtpmap:9 G722/8000. a=rtpmap:96 speex/32000. a=rtpmap:97 speex/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:100 speex/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:6 DVI4/16000. a=rtpmap:15 G728/8000. a=rtpmap:101 telephone-event/8000. a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level. m=video 55774 RTP/AVP 102 99. a=rtpmap:102 H264/90000. a=fmtp:102 packetization-mode=1. a=imageattr:102 send [x=[0-640],y=[0-480]] recv *. a=rtpmap:99 H264/90000. a=imageattr:99 send [x=[0-640],y=[0-480]] recv *. a=nortpproxy:yes. }}} * FreeSWITCH accepts it with 200 OK, returning the following SDP: {{{ v=0. o=Webi 1314226434 1314226435 IN IP4 10.10.10.106. s=Webi. c=IN IP4 127.0.0.1. t=0 0. m=audio 32256 RTP/AVP 9 101. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. m=video 18750 RTP/AVP 102. a=rtpmap:102 H264/90000. }}} Issue as seen by me: FreeSWITCH remaps the 102 from an offer with packetization=1 to a reply with packetization information missing, therefore defaulting to 0 (default out of RFC). Question: * Is this intentional or just some wrong negotiation of packetization parameter? Is FreeSWITCH not supporting packetization=1? Based on the negotiated stream the client should send now 102 with packetization=0 (as preferred by FreeSWITCH in signaling) and FreeSWITCH should return to the client 99 since 102 with packetization=1 does not look like being supported. My FreeSWITCH version is the latest git compiled yesterday: {{{ /opt/freeswitch/bin/freeswitch -version FreeSWITCH version: 1.0.head (git-8f15bc7 2011-08-23 14-40-44 -0500) }}} Thx in advance for any tip! DanB

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about fax status
by Madovsky 25 Aug '11

25 Aug '11

Sometimes fax fails with status like Received a DCN while waiting for a DIS what does it mean ? Thanks Franck

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Re: [Freeswitch-users] Forcing CN offer
by Alex Massover 25 Aug '11

25 Aug '11

Hi, Sorry, the RFC says that a= is not mandatory, m= is enough. I was just confused why FS in some scenario puts it and in other doesn’t, despite using only G711@8000. From: Alex Massover Sent: יום ה 25 אוגוסט 2011 12:26 To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Forcing CN offer Hello, I’m working on the following scenarios: scenario 1: A --> FS --> B scenario 2: C --> FS --> A Where A supports CN (a=rtpmap:13 CN/8000) and requires CN negotiation, and B and C do not support and aren’t able to negotiate it. I successfully implemented scenario 1, CN is negotiated between A party and FS, and not negotiated between FS and B. But scenario 2 doesn’t work for me! When C doesn’t offer CN in INVITE towards FS, FS also doesn’t offer CN in SDP in INVITE towards A. And nothing helps, tried all the combination of VAD options. Gateway A is in the same internal sip profile in both scenarios. This is the SDP in INVITE from C to FS: Media Description, name and address (m): audio 30224 RTP/AVP 0 8 18 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): ptime:20 Media Attribute (a): sendrecv And this is the SDP from FS towards A: Media Description, name and address (m): audio 23564 RTP/AVP 0 8 101 13 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 FS even puts 13 in m= attribute, but doesn’t add a= attribute for rtpmap:13. Is that a bug? -- Best Regards Alex Massover. This mail was sent via Mail-SeCure System.

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Flex Client Receives Periodic Interrupted Audio
by afshin afzali 25 Aug '11

25 Aug '11

Hi Guys, As a taste of RTMP end point, I'm experiencing periodic interrupted audio. Everything is as defaults in client / server sides except that "auth-calls = false" and client does not login / logout, It just makes call. I don't know if it can justified by buffer-len / chunksize in server side or NetStream.bufferTime in client side. appreciate all comment. -- afshin

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Paging with Polycom Phones
by Duvid Rottenberg 25 Aug '11

25 Aug '11

Has anyone successfully implemented paging (auto-answer) with a polycom phone? I am using the Conferencing and Intercom sample which sets the sip_auto_answer variable to true, however on my polycom phone the result is that the phone rings once and hangs up right away (the phone is sending a BYE message). I tried adding an Alert-Info header, however it seems that the Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I couldn't get freeswitch to send the header in this format. Has anyone else been able to either get polycom phones to work with sip_auto_answer or to get freeswitch to send an Alert-Info header in the polycom format? Thanks, Duvid Rottenberg

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