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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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Sangoma is hiring!
by Moises Silva 31 Aug '11

31 Aug '11

Hello everyone, I apologize in advance for the spam. Sangoma is looking for talented software engineers for development and QA. Roughly speaking we would love to have people with open source mentality and that knows his/her way around Linux and multithreaded C. Kernel level coding is an asset. They should be willing to relocate to work in our offices at Toronto, Canada. Full description in LinkedIn: http://ca.linkedin.com/jobs/jobs-Software-Engineer-VoIPTDMVideoNetworking-1… Interested candidates can contact me directly. Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy(a)sangoma.com

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OS X PPC Leopard error on "make"
by Peter Schrock 31 Aug '11

31 Aug '11

I am new to FreeSWITCH (sort of, tried it a while back and dropped it). I have a powerpc Apple running Leopard on it. I followed the directions by Mario G and managed to get all the way up to where he discusses error's with FLITE. I managed to go without the error message because I followed the instructions on how to fix it. However, I have come up with this error message and can't seem to make any sense of it. Help Please. Here is the error I get: make[8]: *** No rule to make target `tport/libtport.la', needed by ` libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Peter

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Condition based on custom sip header
by Alex Massover 31 Aug '11

31 Aug '11

Hi, I'm trying to implement dialplan condition based on custom SIP header ("Header1: value1"). As far as I understood there's no way to read custom SIP header which isn't X- header with sip_h_. Is that correct? What are the other options, please? Is lua's getHeader() suitable for this? -- Best Regards, Alex Massover This mail was sent via Mail-SeCure System.

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VoiceMail Issue - Call Disconnect
by Lloyd Aloysius 31 Aug '11

31 Aug '11

Hi All, I am having difficulty to find the cause of the problem. Voicemail deliver in partially. IVR -> Extension -> Voicemail While leaving the message (voicemail) the calls disconnect. This is happening randomly. <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> How to fix this issue or How to troubleshoot this problem or is there any parameter to fix this problem. Thanks Lloyd

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problem using mod_managed under linux
by covici＠ccs.covici.com 31 Aug '11

31 Aug '11

Hi. I am trying to use mod_managed on my linux box -- gentoo distribution -- and its driving me ... I wrote a very simple program for a test. The program just sets a dtmf callback and then streams a file and writes a log entry when it sees the dtmf. Under Windows 7 net framework 4, it works just fine. Under Linux, using mono versions 2.8.2 qand 2.10.4, I get the following exception which I will put in a pastebin. http://pastebin.freeswitch.org/17232 And the app is in this one http://pastebin.freeswitch.org/17233 I have tried using gmcs and dmcs thinking it may be a .net framework issue, and if I do it wrong, I can bring down fs itself, but the best I can get under Linux is the exception shown above. Thanks in advance for anything you can come up with on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Setup with Bandwidth.com
by Wayne 31 Aug '11

31 Aug '11

Does anyone have a setup with bandwidth.com? The example on the wiki has a username and password. I didn't receive a username and password form bandwidth.com. Just setup a sip peer and have an IP address. If I send calls to freeswitch from the peer I get Rejected by acl "domains". Falling back to Digest auth. Looks like they are sending it on port 5060. I'm sure the rejected by acl is the most asked question but I have been unable to find the answer. Thanks Wayne

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