September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

482 Request merged, in serial forking
by Humberto Quintana 04 Sep '09

04 Sep '09

Hello, I'm a new Freeswitch user. After some reading I put Freeswitch (Version 1.0.4) to work as Session Border Controller. I have only one problem that I dont know how to solve it ( or which parameter to set) and I'd appreciate if someone could give me a clue about this. Kamailio is sitting behind FS and it selects the route or routes in case of failure (serial forking) . Freeswitch is configured to use directly the Request-URI sent by Kamailio. So, when the 1st route fails, Kamailio receives the Reply from FS and sends back the ACK to end the transaction. More than 1 second later, a new INVITE from Kamailio with the next route is tried (With the To-header's tag is empty. Same Callid, From and Cseq header but different VIA-header's branch parameter) and FS is answering back 482 Merged Request. It happens the same for the 3rd route. It seems that the transaction is still 'alive' in FS even if the ACK was received ? Thanks, Humberto ===1st route=== U 2009/09/03 17:20:36.069147 kamailio -> freeswitch INVITE sip:5145555555@gw1 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 U 2009/09/03 17:20:36.169147 freeswitch -> gw1 INVITE sip:5145555555@gw1 SIP/2.0. Call-ID: 2 U 2009/09/03 17:20:36.170158 gw1 -> freeswitch SIP/2.0 100 Trying. Call-ID: 2 U 2009/09/03 17:20:36.190457 gw1 -> freeswitch SIP/2.0 503 Service Unavailable. Call-ID: 2 U 2009/09/03 17:20:36.193296 freeswitch -> gw1 ACK sip:5142776756@gw1 SIP/2.0. Call-ID: 2 U 2009/09/03 17:20:36.227492 freeswitch -> kamailio SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 U 2009/09/03 17:20:36.228122 kamailio -> freeswitch ACK sip:5145555555@gw1 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 ===2nd route=== U 2009/09/03 17:20:37.596885 kamailio -> freeswitch INVITE sip:15145555555@gw2:5061 SIP/2.0 Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 U 2009/09/03 17:20:37.597590 freeswitch -> kamailio SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 U 2009/09/03 17:20:37.598163 kamailio -> freeswitch ACK sip:15145555555@gw2:5061 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 ===3rd route=== U 2009/09/03 17:20:37.642098 kamailio -> freeswitch INVITE sip:5145555555@gw3 SIP/2.0 Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 U 2009/09/03 17:20:37.642634 freeswitch -> kamailio SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 U 2009/09/03 17:20:37.643139 kamailio -> freeswitch ACK sip:5145555555@gw3 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404

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Does FreeSWITCH wiki have update notify?
by Mitul Limbani 04 Sep '09

04 Sep '09

Hello, It may sound a bit stupid but still wanna ask out here, if there is any way to replicate FreeSWITCH wiki mirror for local reference or mainataining local copy instead of Reading online which costs a lot in developing countries like that of ours and Asia/Africa in general. We tried httrack but that brings in everything back here as static HTML so we can't really search coz every search the static content takes us back online on wiki. Any suggestions in this space would be really helpful, in past I have asked if one can mirror FS wiki but so far it won't work unless FS server allows rsync request as how it works with mirroring php.net with all comments etc. I look forward for more suggestions on the same, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com

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Monitoring FreeSWITCH Events via Telnet socket (Net::Telnet)
by Tina Martinez 04 Sep '09

04 Sep '09

Hello, I have a question, but IÃ¢â¬â¢m not certain whether this is a FreeSWITCH issue, or something specific to Perl. I have setup a Perl application (Ã¢â¬ÅlistenerÃ¢â¬Â) that monitors the events of my FreeSWITCH server via a Telnet socket. So far, the application seems to work very nicely, except that the listener does not consistently capture all of the events that are streaming through the socket. I have been able to work around most of the issues, but one of the more significant pain-points is when a new member is getting added to a conference (Event-Name: CUSTOM, Action: add-member). The log file that I generate from the Telnet socket shows the event details, so I know that the data is coming across the pipe, but I donÃ¢â¬â¢t consistently see the details in my event listener code. Also, the fact that I sometimes do see all of the details confuses me (I don't see a pattern to give any clues towards the cause). BTW - I do have the <action application="verbose_events" data="true"/> in the dialplan. So, I wonder, is there something else that I can do in FreeSWITCH to increase the reliability of capturing the event details? or is there something I should be doing in Perl to somehow buffer the data (i.e., why would I see the details in the socket log file, but not in the data stream within the code?). I'm not a socket wizard by any stretch, so I'm hoping that it might be a simple issue related to the Net::Telnet implementation. Any suggestions would be greatly appreciated. Thank you in advance, - Tina

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Proxy authorization
by Dmitry Bely 04 Sep '09

04 Sep '09

Hi, My SIP provider's gateway requires authorization with correct realm specified. So I configured gateway with "realm" parameter: <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="realm" value="$${default_provider_from_domain}"/> <param name="proxy" value="$${default_provider}"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/>  <param name="context" value="public"/> </gateway> (default_provider_register = true) Unfortunately even after that there is no "Authorization:" header in the REGISTER message: REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF Max-Forwards: 70 From: <sip:1111111@Domain;transport=udp>;tag=Nrc6Z9yrNBS3H To: <sip:1111111@Domain;transport=udp> Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4 CSeq: 119885384 REGISTER Contact: <sip:gw+1.2.3.4@5.6.7.8:5080;transport=udp> Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 and the provider responds with 403 User '1111111' is disabled (as there is no correct authorization realm there). How to force Authorization header? Am I missing something? - Dmitry Bely

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SIP provider directory?
by Lon Baker 04 Sep '09

04 Sep '09

Does anyone know of a SIP provider or network directory? A list of all the public service provider or networks? Gizmo, Google Voice, etc? Or Vitelity, iCall? Lon Baker Kickass Pixels - +1-415-894-0184 - http://kickasspixels.com http://twitter.com/kickasspixels http://www.linkedin.com/in/lonbaker --

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REMINDER: Weekly call is now happening. Join us!
by Michael Collins 04 Sep '09

04 Sep '09

Hello all, We are now on line and welcoming callers. Here's the agenda so far: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04 Come join the conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> 1-213-799-1400 -MC

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UK English wav files
by Brian Stafford 04 Sep '09

04 Sep '09

Hi all anyone know where I can find UK English recordings for the FS prompts (assuming there are any)? (I've googled to no avail). Alternatively is there a list of the text used so we can record our own? Regards Brian

2 1

skypiax error
by Jingwei Yang 04 Sep '09

04 Sep '09

Hi Folks, I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine. After having followed the big help doc from the wiki page ( http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITC…) I hit an error when running multi.sh (under freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username). *Couldn't open RGB_DB '/usr/share/X11/rgb' error opening security policy file /usr/lib64/xserver/SecurityPolicy * This error seems not stopping xvfb from getting started. Then I started FS and loaded mod_skypiax. However, when I initiated a test call (originate skypiax/ANY/userAAA &echo), I saw a bunch of ALSA lib errors popping up: *ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi* Again, this error doesn't prohibit the call from reaching me. It's just too annoying and it keeps popping up after a while. Does anyone know how to get rid of those errors? I found a similar post here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html. Phil, if you happen to see my question, could you please reply and let me know what the cause was and what you have done to solve it? Thanks! By the way, I started xvfb and FS using root. Regards, -Jingwei

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Passing Variables in FS
by Ahmed Munir 04 Sep '09

04 Sep '09

Hi, I'm newbie in FS. As far as I know for setting up custom variables in FS we use this syntex in dialplan XML i.e. <action application="set" data="ABC='value'"/> But when I call this variable using eval application i.e. <action application="eval" data="${ABC}"/> the value I get from variable ABC is undefined means no values are passed to the variable. So kindly do let me know how I can pass values in variables in FS. -- Regards, Ahmed Munir

2 1

FS on 5400zl Procurve module
by Luis F Urrea 04 Sep '09

04 Sep '09

Has anyone out there has the opportunity to get hands on a 5400zl series Procurve? FS on the Intel based module That would be a sweet application!! As far as I understand applications need to undergo some testing before they can be run on the module. anyone can comment??

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