September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Investigating build testing for FreeSWITCH
by Michael Collins 01 Sep '11

01 Sep '11

Hey all, On today's conference call someone pointed out that there is a utility called BuildBot that helps to automate testing of new commits to git, svn, etc. You can find it at buildbot.net. The catch: it's written in Python. (blech) I tried to build it on 3 different systems and I get 3 different undecipherable Python-ish errors. (Stuff like syntax errors during the build or syntax errors in twstd while it's running, dependency mismatches for python-virtualenv, etc.) If you have a system that can test this out, and you can stand to use Python this much (:D) then please try out the steps outlined here: http://buildbot.net/buildbot/docs/current/tutorial/firstrun.html Supposedly this thing will let us run multiple tests on multiple boxes and aggregate the results. I'll believe it when I see it. :) If you get this working please let me know. Also, if you can set up a test platform that would also be helpful. Ideally we would have multiple Linuxes, OS X, Windows, etc. and test on all these platforms. Let me know what you come up with. Thanks, Michael

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freeswitch-1.0.7 Conference set auto outcall bug?
by qiaoqiao7036 01 Sep '11

01 Sep '11

Hi, I have used the function of conference set auto outcall in freeswitch 1.0.6, it is very good, several days ago, I installed a new freeswitch with git, the version is 1.0.7, I wrote a very simple dialplan to test the conference auto outcall function, and the problem happened: This is my dialplan, I have tested it in my freeswitch 1.0.6 successfully: <extension name="Demonstrate conference_set_auto_outcall"> <condition field="destination_number" expression="^(12345)$"> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_flags=none"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${ caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${ caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=default"/> <action application="conference_set_auto_outcall" data="[call_timeout=15]user/1001@$${domain}"/> <action application="conference_set_auto_outcall" data="[call_timeout=15]user/1002@$${domain}"/> <action application="conference" data="$1@default"/> </condition> </extension> Problem 1: When I use a softphone to dial number 12345, in my old freeswitch I can hear the moh music when the phone 1001 and 1002 is ring, But in freeswitch 1.0.7 I can't hear anything, and I found the debug informations in freeswitch cli: 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK] 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:369 (sofia/internal/sip:1001@192.168.2.226:5060) State INIT going to sleep 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1001@192.168.2.226:5060) Running State Change CS_ROUTING 2011-08-31 18:59:14.520108 [DEBUG] switch_channel.c:1828 (sofia/internal/sip:1001@192.168.2.226:5060) Callstate Change DOWN -> RINGING 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK] 2011-08-31 18:59:14.520108 [DEBUG] sofia.c:5128 Channel sofia/internal/sip:1001@192.168.2.226:5060 entering state [calling][0] 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1001@192.168.2.226:5060) State ROUTING 2011-08-31 18:59:14.520108 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001@192.168.2.226:5060 SOFIA ROUTING 2011-08-31 18:59:14.520108 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001@192.168.2.226:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK] 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1001@192.168.2.226:5060) State ROUTING going to sleep 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1001@192.168.2.226:5060) Running State Change CS_CONSUME_MEDIA 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/sip:1001@192.168.2.226:5060) State CONSUME_MEDIA 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/sip:1001@192.168.2.226:5060) State CONSUME_MEDIA going to sleep 2011-08-31 18:59:14.540080 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000(a)192.168.2.242 [BREAK] 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000(a)192.168.2.242 [BREAK] 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000(a)192.168.2.242 [BREAK] 2011-08-31 18:59:14.559978 [DEBUG] sofia.c:5128 Channel sofia/internal/1000(a)192.168.2.242 entering state [ready][200] That mean the freeswitch is sending the moh music to the caller 1000, is that right? But I can't hear anything. Problem 2: when just one phone pickup, like phone 1001, and phone 1002 don't, (I use phone 1000 to call number 12345) I can't hear anything from phone 1001 when 1002 is ringing except phone 1002 pick up or when the phone 1002 is timeout to ring, it's very stranger!!! I tested this situation in freeswitch 1.0.6, when just only person pickup the phone the conference will work, 1000 will hear the phone 1001's voice, don't need the 1002 to pickup or timeout. By the way: I have tested this with the newest version: FreeSWITCH Version 1.0.head (git-6a5f6e5 2011-08-30 15-00-07 -0500) The problem is still exist. Thank you for advising me how to solve this problem. Regards! Dennis

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Reorder tone immediately on Cisco 7960 Line 1 upon off-hook
by A E [Gmail] 01 Sep '11

01 Sep '11

Hi Guys, I'm pretty sure this is a Cisco firmware problem (v 8.9) problem but just wanted to check if anyone else has seen this. With Line 1 on the Cisco 7960 registered to FS, when the line buttons is pressed or the "Speaker" button is pressed, it immediately gets a re-order tone instead of a dial-tone. Watching the console on FS, I see an Invite going to "%14" which obv doesn't exist. INVITE sip:%14@192.168.3.80 SIP/2.0 Via: SIP/2.0/UDP 69.201.146.xxx:5062;branch=z9hG4bK121399ed From: "user1" <sip:user1@192.168.3.80 >;tag=003094c3e08b02705d7d5000-560be9da To: <sip:%14@192.168.3.80> Call-ID: 003094c3-e08b000a-43fa29c5-5d1840e9(a)69.201.146.xxx Max-Forwards: 70 Date: Thu, 01 Sep 2011 16:39:30 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: <sip:user1@69.201.146.xxx:5062;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 278 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 12375 0 IN IP4 69.201.146.xxx s=SIP Call t=0 0 m=audio 19982 RTP/AVP 0 8 18 101 c=IN IP4 69.201.146.xxx a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv When this same account is configured on any other line, pressing the line button brings up a dial-tone. Anyone have any idea why it does that or how to make it not do it? Cheers aeg

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Yealink t28p setup - please help
by ocset 01 Sep '11

01 Sep '11

Hi All I am new to FreeSwitch and this is my first post on this forum/mailing-list. I have a default install of FS on Ubuntu 10.10 and have two Yealink T28P phones. I am trying to understand the best way to config these phones for FS. What I am fighting with at the moment is that if I assign the same extension to both phones, only one phone will ring when the shared extension is called. The phones have the option to assign 6 SIP accounts/extensions and this is my current test config. Phone 1 ext 1001 ext 1010 ext 1011 Phone 2 ext 1002 ext 1010 ext 1011 My thinking behind this is that I want to assign a POST line to the 1010 extension (using an SPA3102) and a VOIP line to the 1011 extension. That way, when an external call comes in on one of those lines, all parties have the option to pick up the call. What I have found however is that only one of the phones rings and is able to answer the 1010 and 1011 extensions. Is this expected behaviour? Should I be doing this another way? I am very new to VOIP, Freeswitch etc so my terminology may be wrong - hopefully you understand my explanation. Thanks in advance

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/usr/src/freeswitch.git/libs/ldns/configure: No such file or directory
by David Wafula 01 Sep '11

01 Sep '11

Hi all, I am updating my freeswitch installation ..after months. Am unable to get past this...have never encountered it before: /bin/bash: /usr/src/freeswitch.git/libs/ldns/configure: No such file or directory make[5]: *** [/usr/src/freeswitch.git/libs/ldns/Makefile] Error 127 make[4]: *** [install] Error 1 make[3]: *** [mod_enum-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 Environment: Ubuntu 10.04 i686, running in Linode.com Regards. -- David Wafula

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Start FreeSWITCH as a Windows service with -nonat option
by Aleš Zelenik 01 Sep '11

01 Sep '11

Hello all, I am having trouble starting fs as a windows service with -nonat option If I run from command prompt "FreeSwitchConsole -nonat" it works ok, but I am left with open window in logged-in session. Command to run as a service is "...path...\FreeSwitchConsole.exe -service FreeSWITCH", which can be also verified with "sc qc FreeSWITCH" If I want to add an argument -nonat with command: sc config FreeSWITCH binPath= "C:\Program Files (x86)\FreeSWITCH \FreeSwitchConsole.exe -nonat -service FreeSWITCH" service won't start anymore also, fs cannot be installed as a service with optional arguments, eg FreeSwitchConsole -install -nonat Anyone managed to run fs like this? Thanks, -- Ales Zelenik,

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