February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Happy B-Day BKW!
by Ken Rice 18 Feb '09

18 Feb '09

Ok Tomorrow is BKW's b-day... Lets as a community see if we cant pull together and get him something nice off his Amazon wish list... Visit his wish list here... http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0 If you want to chip in contact me on IRC (I'm SwK) or send something via the donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put "For BKW" in the comment or drop me an off list email... If you want to send something via paypal direct to brian his paypal ID is brian(a)freeswitch.org Come on guys he works day and night on FS to help bring us arguably the best OpenSource Telephony platform out there!! Happy B'Day Brian!! Ken

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Re: [Freeswitch-users] Sending media streams to a media gateway
by fax＠virgintechnologies.com 18 Feb '09

18 Feb '09

I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge <action application="set" data="bypass_media=true/> <action application="bridge" data="sofia/internal/someuser@somehost"/> that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, <fax(a)virgintechnologies.com> wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale@hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888@conference.freeswitch.org http://iax:guest@conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888@conference.freeswitch.org pstn:213-799-1400

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Sending media streams to a media gateway
by fax＠virgintechnologies.com 18 Feb '09

18 Feb '09

I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin

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Debian rules
by Dan 17 Feb '09

17 Feb '09

Hi guys, I noticed that the debian build is missing lines for shout.conf.xml and does not install mod_flite (if its built) . This can be fixed by adding the following lines to debian/freeswitch.install opt/freeswitch/mod/mod_flite* opt/freeswitch/conf/autoload_configs/shout.conf.xml and the following lines to debian/freeswitch.conffiles /opt/freeswitch/conf/autoload_configs/shout.conf.xml thanks.

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Big delays in playing audio files
by Nik Middleton 17 Feb '09

17 Feb '09

Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards

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xml_cdr Call Flow for attended transfer
by Luis F Urrea 17 Feb '09

17 Feb '09

Hi all, I am trying to understand xml_cdr for an attended (consultative) transfer, I was thinking that the A-leg that initially originated the call would remain untouched but I see that it's global tags get replaced. I have a test call that goes as follows: 201 originates a call and talks to 203 -----> A-leg(1) and B-leg(1) 203 puts 201 on hold and calls 202 (attended) ------> A-leg(2) and B-leg(2) 203 transfers the call 201 and 202 are talking ------> A-leg(1) w/ B-leg(2) ??? Here are the relevant captures: A-leg(1) http://pastebin.freeswitch.org/7253 B-leg(1) http://pastebin.freeswitch.org/7254 A-leg(2) http://pastebin.freeswitch.org/7252 B-leg(2) http://pastebin.freeswitch.org/7255 I was expecting A-leg(1) to have <variables> corresponding to 201 which is the original A-leg but it seems that on the transfer, it reverts and 202 appears as the A-leg and 201 as the B-leg. Can someone shed some light on how that transfer gets logged in terms of A-leg and B-leg? TIA

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Re: [Freeswitch-users] Realm Value
by Ali Al-Rubaie 17 Feb '09

17 Feb '09

I have to use a specific softphone, HelpCaster, but it can not pass the authentication stage. However it can authenticate with OpenSips server! What I had noticed is that it uses static realm with OpenSips therefore I'm trying to do the same. recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ------------------------------------------------------------------------ REGISTER sip:209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: sip:1001@209.82.10.235 Contact: sip:1001@209.82.10.250:1059 Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775306 REGISTER Content-Length: 0 Expires: 3600 ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: <sip:1001@209.82.10.235>;tag=7yam2F01ZH3vH Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775306 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ------------------------------------------------------------------------ REGISTER sip:209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: sip:1001@209.82.10.235 Contact: sip:1001@209.82.10.250:1059 Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775307 REGISTER Content-Length: 0 Expires: 3600 Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: <sip:1001@209.82.10.235>;tag=873c4aH5vtSFD Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775307 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ Thanks! Message: 2 Date: Tue, 17 Feb 2009 11:39:58 -0600 From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users(a)lists.freeswitch.org Message-ID: <2ADB8FA0-3E87-4C99-9A6F-F09C6E0F5DB8(a)freeswitch.org> Content-Type: text/plain; charset="us-ascii" I'm trying to get a clear picture of what you're trying to accomplish. Why would you need/want to set a static realm? Anyway can you collect sip traces? /b On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote: > Thanks Brian, > > Actually we're using freeswitch ver 1.0.2. > > > Regards,

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Options for configure script
by Pablo Hernan Saro 17 Feb '09

17 Feb '09

Hi there, Anyone knows how to configure FS in order to have log folder in /var/log/freeswitch ? I did ./configure --prefix=/opt/freeswitch and after install I got logs, pid file and cdrs under /opt/freeswitch/log Thanks in advance. Pablo

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Re: [Freeswitch-users] Realm Value
by Ali Al-Rubaie 17 Feb '09

17 Feb '09

Thanks Brian, Actually we're using freeswitch ver 1.0.2. Regards, Message: 5 Date: Thu, 12 Feb 2009 15:48:00 -0600 From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] Realm value To: freeswitch-users(a)lists.freeswitch.org Message-ID: <F7EFCA6F-FA3F-4CC1-8CEC-4AA56EDFF75D(a)freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > <param name="challenge-realm" value="150"/> > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! >

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SIPX/FS Auto attendant
by Chris Jones 17 Feb '09

17 Feb '09

I'm having a problem that calls to the auto-attendant won't transfer. I know this has been a problem in the past but thought that it was fixed. Whenever I enter an extension (or press a key to transfer me to one), the call just hangs up. I ran Freeswitch on the console, but all I see happening is this: 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116] has been answered 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() Hangup sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 ( sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116) Ended 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116 [CS_HANGUP] Does anyone have an idea as to how to fix this? The call is coming from a SIP Trunk on vitelity.

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