February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

little problem with gateway registration
by Helmut Kuper 11 Feb '09

11 Feb '09

Hello, today I connected FS to my SIP-DDI SoftSwitch. It registered successfully, but not as expected at the register-proxy (sip2.ewetel.net) but at the proxy (proxy2.ewetel.net) My gateway config is this: <include> <gateway name="IPPBXSIPDDI">  <param name="username" value="ippbx"/>  <param name="realm" value="sip2.ewetel.net"/>  <param name="from-domain" value="sip2.ewetel.net"/>  <param name="password" value="XXXXX"/>  <param name="proxy" value="proxy2.ewetel.net"/>  <param name="register-proxy" value="sip2.ewetel.net"/>  <param name="expire-seconds" value="60"/>  <param name="retry-seconds" value="30"/> </gateway> </include> The corresponding register request generated by FS is this: REGISTER sip:proxy2.ewetel.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 85.16.246.22;branch=z9hG4bK2FNpmSQHpZtye Via: SIP/2.0/UDP 85.16.246.6:5070;received=85.16.246.6;rport=5070;branch=z9hG4bK2FNpmSQHpZtye Max-Forwards: 69 From: <sip:ippbx@sip2.ewetel.net;transport=udp>;tag=QrFtyDUrpg94N To: <sip:ippbx@sip2.ewetel.net;transport=udp> Call-ID: 8bd8f32c-f759-11dd-bb38-778d64a5265f CSeq: 111011668 REGISTER Contact: <sip:gw+IPPBXSIPDDI@85.16.246.6:5070;transport=udp> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11698M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Authorization: Digest username="ippbx", realm="sip2.ewetel.net", nonce="4991527400000e269de527c4c699b9466bbf1e015cd0532c", cnonce="HE1PznH9Eiyb4QAUT+bjMA", algorithm=MD5, uri="sip:proxy2.ewetel.net;transport=udp", response="b5af17310919ec6b8f82acd4c89ffb9e", qop=auth, nc=00000001 Content-Length: 0 Shouldn't it register to register-proxy when it is given? regards Helmut

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Dynamic Dialplan
by Doug Blacksone 11 Feb '09

11 Feb '09

Hi, Right now, I am working on getting freeswitch configured for our call-center with more than 1000 agents. There are several areas where we need the dialplan to be configurable based on some user detail in the database. Therefore, the dialplan needs to be some-what dynamic based-on inputs from the database. I would like to know from other implementation as to the most scalable way of doing high performance dynamic dialplan that is super scalable. There are three ways I can think of: 1. Static dialplan using customized freeswitch mod to access postgres for data pros: best performance cons: harder to program 2. Static dialplan using lua to access postgres for data pros: easy to program, maybe-performance is better than curl cons: need to search through all the extensions to find one dialplan, performance is slower than the first one. 3. curl-based dialplan using Java Servlet and HTTP pros: easy to program, freeswitch only gets one extension and no extension search cons: performance is slow than the other two Is this a correct analysis? If from a pure performance's perspective, how much performance can a customized mod gains in comparison to lua? For a production system that needs to be highly scalable, what do you recommend? Thank you very much for any input to our critical design decision. Doug

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dialplan question
by Jonathan Creasy 11 Feb '09

11 Feb '09

I'm trying to setup an extension to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX@contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX@contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios?

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High CPU load after starting
by Public Dump 10 Feb '09

10 Feb '09

After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards

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dialplan question
by Jonathan Creasy 10 Feb '09

10 Feb '09

I'm trying to setup a dialstring to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX@contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX@contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios?

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FreeSWITCH uclibc segfault
by Kristian Kielhofner 10 Feb '09

10 Feb '09

I don't think this is worth filing a bug for (yet)... FS rev 11655 segfaults with AstLinux (uClibc). Backtrace: http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt I'm sorry it doesn't have all the symbols... Everything except FS is stripped. Configure options: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-frees… FreeSWITCH compiles cleanly. Are there any known issues with uclibc (couldn't find anything on Jira) or did I do something stupid? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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DTMF not being recognised
by Nik Middleton 10 Feb '09

10 Feb '09

Hi Guys, I have an IVR that's working fine on internal extensions, but when a call is via my sip GW, they're not being trapped. I have tried the following in the gw profile <param name="dtmf-type" value="rfc2833"/> <param name="rfc2833-pt" value="101"/> <param name="pass-rfc2833" value="false"/ I should add that this sip provider works fine with asterisk. Anyone any ideas? Regards

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Recording play end of sound file again
by Blake France 10 Feb '09

10 Feb '09

Whenever I try to record and IVR or Voicemail Greeting, it will record and playback, but playback does something like this. "Please leave a message" ... "Message" It plays the end of the sound file AGAIN after playing the sound file. I've tried leaving extra time before and after speaking, but it still does this. Anyone ran into this issue?

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Can I force the A-leg codec in FS?
by Helmut Kuper 10 Feb '09

10 Feb '09

Hello, in my reallife setup of FS all internal extensions use G.722 as preferred codec. Unfortunately when there is an outgoing TDM call I found that FS starts transcoding instead of forcing G.711 for A leg. So is there a way to force the A codec? regards helmut

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javascript to get the status
by Gopalakrishnan A.N 10 Feb '09

10 Feb '09

Hi, I am trying to execute the following script, its working fine for call origination, but cant able to get the status for dialed numbers, able to get only the last dialed number not for both the numbers. The script as follows, Javascript var array = [2]; array[0]="39841799874"; array[1]="39894929942"; for(var i=0;i<array.length;i++) { session = new Session(); session.originate(session, "{ignore_early_media=true}sofia/default/"+array[i]+"@172.20.191.228"); if (session.ready()) { //session.setVariable("session.uuid", "Session OUTPUT: " + session.uuid); //session.setVariable("notice", "Disconnect cause: " + session.cause); //session.setVariable("info", "Disconnect cause: " + session.causecode); console_log("notice", "Session UUID: " + session.uuid + "\n"); console_log("info", "Disconnect cause: " + session.cause + "\n"); console_log("notice", "Disconnect Code: " + session.causecode + "\n"); } } if (session.answer()) { session.execute("sleep","5000000"); console_log("notice", "Session UUID: " + session.uuid + "\n"); console_log("info", "Disconnect cause: " + session.cause + "\n"); console_log("notice", "Disconnect Code: " + session.causecode + "\n"); } The steps as follows, 1. enter into the array - working fine 2. take each number to the originate session via for loop - working fine 3. the call has to be answered by the session - working fine 4. the call status need to be printed in console for each call - printing only for the last dialed call Any suggestion would help us. -- Thank you with regards, Gopal,

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