May 2007 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Cluecon 2007 is almost here don't forget to register. June 26th - 28th
by Brian West 01 Jun '07

01 Jun '07

FreeSWITCHers, If you haven't please take a moment and look at http:// www.cluecon.com, This year is on track to be even better than our previous two years. If you can't make it please tell all your friends and co-workers that might be interested. Meet the people that make Open Source VoIP possible. Surj Patel and I will be working the conference together this year so please show up.. interact.. have fun. Cluecon Telephony Developer Conference 2007 (June 26th - 28th) will be held in: Best Western Inn of Chicago 162 E Ohio Street Chicago, Illinois 60611 Phone: (312) 787-3100 Toll Free: (800) 557-2378 US/Canada Call 877 742-CLUE for more information about the conference or to just chat with me. Thanks, Brian West PS: <subliminal-message>YOU MUST GO!!!</subliminal-message>

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Re: [Freeswitch-dev] SIP Timeout waiting for 200 ACK
by Mike Murdock 30 May '07

30 May '07

I am dealing with the same problem... Brian, there is something wrong here. The effect is my calls get terminated every two minutes +hen my sipura re-registers. I have a trace with full debug captured if you want to see. Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Brian West [mailto:brian.west@mac.com] Sent: Tuesday, May 29, 2007 10:14 AM Eastern Standard Time To: freeswitch-dev(a)lists.freeswitch.org Subject: Re: [Freeswitch-dev] SIP Timeout waiting for 200 ACK I think that since we are a B2BUA tech_pvt->profile->url is the right way. The other way is correct if you're a proxy. Remember FreeSWITCH NOT a proxy. /b On May 29, 2007, at 8:58 AM, freeswitch(a)dalethatcher.com wrote: > I'm seeing an issue during a bridge where the '200' from freeswitch is > not being ACK'd and Freeswitch is correctly dropping the connection. > However I belive that the Contact header in the 200 packet is causing > the problem. Am I missing a config parameter? > > If I back out the revision 5208 change to mod_sofia.c from the trunk > source then the problem goes away. (SIPTAG_CONTACT_STR(tech_pvt- > >to_uri) > back to SIPTAG_CONTACT_STR(tech_pvt->profile->url)). > > My test setup is (Freeswitch is behind a NAT): > > External SIP -> Freeswitch -> Gizmo > > The freeswitch config for the extension is: > > <extension name="test"> > <condition field="destination_number" expression="^test$"> > <action application="bridge" data="sofia/gizmo/ > 17470355893(a)proxy01.sipphone.com"/> > </condition> > </extension> > > I've put a full copy of the log here: > > http://pastebin.freeswitch.org/3008 > > My internet facing profile is: > > <profile name="internet"> > <settings> > <param name="debug" value="1"/> > <param name="rfc2833-pt" value="101"/> > <param name="sip-port" value="5060"/> > <param name="dialplan" value="XML"/> > <param name="dtmf-duration" value="100"/> > <param name="codec-prefs" value="$${global_codec_prefs}"/> > <param name="codec-ms" value="20"/> > <param name="use-rtp-timer" value="true"/> > <param name="rtp-timer-name" value="soft"/> > <param name="rtp-ip" value="$${bind_server_ip}"/> > <param name="sip-ip" value="$${bind_server_ip}"/> > <param name="accept-blind-reg" value="true"/> > <param name="nonce-ttl" value="60"/> > <param name="ext-rtp-ip" value="80.189.98.58"/> > <param name="ext-sip-ip" value="80.189.98.58"/> > <param name="sip-domain" value="sip.dalethatcher.com"/> > </settings> > </profile> > > With the current trunk the SIP packet that does not get a response is: > > send 1182 bytes to udp/[193.111.201.32]:5060 at 13:05:15.136189: > > ---------------------------------------------------------------------- > -- > SIP/2.0 200 OK > Via: SIP/2.0/UDP 193.111.201.32;branch=z9hG4bK614f.db2cc426.0 > Via: SIP/2.0/UDP 193.111.200.182;branch=z9hG4bK614f.9dbef354.0 > Via: SIP/2.0/UDP > 194.145.189.10:3670;branch=z9hG4bK00E0F556495005A857E00000230F > Record-Route: <sip: > 193.111.201.32;lr=on;ftag=00E0F556495005A857E00000222A> > Record-Route: <sip: > 193.111.200.182;lr=on;ftag=00E0F556495005A857E00000222A> > From: <sip: > 02085555129(a)194.145.189.10>;tag=00E0F556495005A857E00000222A > To: <sip:442070968842@193.111.200.182>;tag=r5ae93vm8aK1p > Call-ID: 00E0F556495005A857E000001549(a)194.145.189.10 > CSeq: 35816 INVITE > Contact: <sip:442070968842@193.111.200.182:5060> > User-Agent: FreeSWITCH(mod_sofia) > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: 100rel, precondition > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 290 > > v=0 > o=FreeSWITCH 6489556414011969222 9057524047013517960 IN IP4 > 192.168.93.5 > s=FreeSWITCH > c=IN IP4 80.189.98.58 > t=0 0 > a=sendrecv > m=audio 10002 RTP/AVP 0 96 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > m=image 0 UDPTL 9 > > ---------------------------------------------------------------------- > -- > > Backing out the change modifies the Contact header to: > > Contact: <sip:mod_sofia@80.189.98.58:5060> > > Which gets an ACK packet and doesn't drop the call. > > thanks, > > - Dale > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

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[Freeswitch-dev] bay area person needed to get pri docs
by Trixter aka Bret McDanel 30 May '07

30 May '07

There are some documents that are desired for completion of the pri stack, they are located in San Mateo. I was hoping that someone on this list is local and can collect them so they can be used to make the PRI stuff more robust. I know where they are but ultimately they need to get to mikej. Can anyone local see about retreiving them? They are being donated by someone I have been in contact with, but he wanted someone to physically get them. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

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[Freeswitch-dev] SIP Timeout waiting for 200 ACK
by freeswitch＠dalethatcher.com 29 May '07

29 May '07

I'm seeing an issue during a bridge where the '200' from freeswitch is not being ACK'd and Freeswitch is correctly dropping the connection. However I belive that the Contact header in the 200 packet is causing the problem. Am I missing a config parameter? If I back out the revision 5208 change to mod_sofia.c from the trunk source then the problem goes away. (SIPTAG_CONTACT_STR(tech_pvt->to_uri) back to SIPTAG_CONTACT_STR(tech_pvt->profile->url)). My test setup is (Freeswitch is behind a NAT): External SIP -> Freeswitch -> Gizmo The freeswitch config for the extension is: <extension name="test"> <condition field="destination_number" expression="^test$"> <action application="bridge" data="sofia/gizmo/17470355893(a)proxy01.sipphone.com"/> </condition> </extension> I've put a full copy of the log here: http://pastebin.freeswitch.org/3008 My internet facing profile is: <profile name="internet"> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/> <param name="accept-blind-reg" value="true"/> <param name="nonce-ttl" value="60"/> <param name="ext-rtp-ip" value="80.189.98.58"/> <param name="ext-sip-ip" value="80.189.98.58"/> <param name="sip-domain" value="sip.dalethatcher.com"/> </settings> </profile> With the current trunk the SIP packet that does not get a response is: send 1182 bytes to udp/[193.111.201.32]:5060 at 13:05:15.136189: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 193.111.201.32;branch=z9hG4bK614f.db2cc426.0 Via: SIP/2.0/UDP 193.111.200.182;branch=z9hG4bK614f.9dbef354.0 Via: SIP/2.0/UDP 194.145.189.10:3670;branch=z9hG4bK00E0F556495005A857E00000230F Record-Route: <sip:193.111.201.32;lr=on;ftag=00E0F556495005A857E00000222A> Record-Route: <sip:193.111.200.182;lr=on;ftag=00E0F556495005A857E00000222A> From: <sip:02085555129@194.145.189.10>;tag=00E0F556495005A857E00000222A To: <sip:442070968842@193.111.200.182>;tag=r5ae93vm8aK1p Call-ID: 00E0F556495005A857E000001549(a)194.145.189.10 CSeq: 35816 INVITE Contact: <sip:442070968842@193.111.200.182:5060> User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 290 v=0 o=FreeSWITCH 6489556414011969222 9057524047013517960 IN IP4 192.168.93.5 s=FreeSWITCH c=IN IP4 80.189.98.58 t=0 0 a=sendrecv m=audio 10002 RTP/AVP 0 96 13 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=image 0 UDPTL 9 ------------------------------------------------------------------------ Backing out the change modifies the Contact header to: Contact: <sip:mod_sofia@80.189.98.58:5060> Which gets an ACK packet and doesn't drop the call. thanks, - Dale

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[Freeswitch-dev] Google talk and other issues
by SLim 25 May '07

25 May '07

Hello freeswitch people! first of all, congratulations for your work and ongoing effort! I have a few questions regarding the features of freeswitch and it would be great if you had some time to answer them... here they are : * is freeswitch able to retrieve the presence status of any Google Talk client (prior to making a routing decision) ? * is it possible to receive an incoming Google Talk call (voice + chat) on freeswitch and bridge it to another Google Talk client that is not registered on freeswitch ? * is it possible to initiate an outgoing Google talk call (chat text only or voice +chat text) from freeswitch ? * is it possible to parse an incoming Google talk chat text within freeswitch (prior to making a routing decision for instance) ? * is there anything planned with regards to VXML and CCXML within freeswitch ? * I have been told the guys from www.i6net.com are making a version of their VXIasterisk product for freeswitch. Can you confirm it ? * is there anything planned with regards to queueing within freeswitch ? * is there anything planned with regards to call center agent positions within freeswitch ? * is it possible to add custom data to your CDR ? Thanks for your help and answers! Best regards.

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[Freeswitch-dev] pastebin
by Brian West 24 May '07

24 May '07

Can someone that is good with php provide me with a patched version of pastebin that has captcha support .. I have had to do some drastic things to stop spammers. I'm not that great with php but someone out there can whip up a patched version ? /b

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[Freeswitch-dev] freeswitch use with .NET
by Josh Lloyd 19 May '07

19 May '07

Hi, I saw a few open ended threads regarding using FreeSwitch. I have downloaded the c# assembly and have it compiled successfully and referenced from a test project. There are so many classes, properties, methods, etc. I really am not sure where to begin and it does not see anyone else has been answered yet. Is there any documentation or perhaps a sample program available using the .net api. Many thanks! Josh Lloyd

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[Freeswitch-dev] It must be Monday
by lconroy 16 May '07

16 May '07

Hi Folks, just signed up, and I can't enter a bug/featurette report into Jira as requested - it crashed when I searched for the module name. Sigh... Does anyone know what where mod_enum is under the JIRA tree? I had though it would be under applications, but this doesn't list it. (Yup - Jira crashed when I did a quick search on mod_enum :). There's an obscure issue with ENUM parsing of valid Resource Records, but until I can find where to create the issue (and propose a fix), I'm stuck. help! all the best, Lawrence

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Re: [Freeswitch-dev] Unintended call termination
by Anthony Minessale 14 May '07

14 May '07

Line 781 shows the other end of the call sending a bye. For whatever reason it is that end of the call who is doing it. What looks more interesting is all those registers I think that must have at least something to do with your issue. Can you collect the problem description and the traces and post them to a bug report in jira? Thanks, Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Ivan C Myrvold <ivan(a)myrvold.org> To: freeswitch-dev(a)lists.freeswitch.org Sent: Sunday, May 13, 2007 5:15:53 AM Subject: Re: [Freeswitch-dev] Unintended call termination Here is the trace, following your advice. Hope you can find out what is wrong. I also have a full Wireshark trace file of the whole conversation if you want that.Looks like it is on line 810 the unintended termination is happening. http://pastebin.freeswitch.org/750 Ivan On 11. mai. 2007, at 18:28, Anthony Minessale wrote: Also make sure you run freeswitch like this when reproducing: NUA_DEBUG=9 TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

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[Freeswitch-dev] Attention all FreeSWITCH users BETA release this week.
by Anthony Minessale 14 May '07

14 May '07

Hi, I would like to announce that FreeSWITCH will be entering into a BETA status within the week so we can produce a series of release candidates which will ultimately produce a formal release by the end of the summer or sooner if possible. Our software is growing rapidly and we've come a long way from our modest initial public release in January of 2006. I hope everyone enjoys the opportunity to participate in the development process which is one of the best benefits of open source software in my opinion. Not only do we have a few finishing touches to put on the code, we also have to institute a version policy, make sure the WIKI is accurate and, of course, find all the bugs so we can focus our energy on forward development and stay away from nasty unresolved bugs. In order to make the debugging process successful we ask that everyone use our jira tracker for all bug reports http://jira.freeswitch.org, feature requests, or feature contributions. We are glad to help but it's beginning to be more than we can handle in real-time so we really need to document issues on the tracker so we won't forget! We also ask anyone who receives help to please pay it forward and document it on the WIKI http://wiki.freeswitch.org. There are plenty of logistical and clerical necessities to make FreeSWITCH a success so anyone who is interested in helping out by being a bug marshal or managing the svn for one of the modules or any way you think you can help, please mail the dev list http://lists.freeswitch.org or visit us on IRC. Please also feel free to start a new page on the wiki with your irc nick and your paypal address or wishlist urls so when you help someone they can show their gratitude properly. Thank you all for participating and making FreeSWITCH a fun project! Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Ivan C Myrvold <ivan(a)myrvold.org> To: freeswitch-dev(a)lists.freeswitch.org Sent: Sunday, May 13, 2007 5:15:53 AM Subject: Re: [Freeswitch-dev] Unintended call termination Here is the trace, following your advice. Hope you can find out what is wrong. I also have a full Wireshark trace file of the whole conversation if you want that.Looks like it is on line 810 the unintended termination is happening. http://pastebin.freeswitch.org/750 Ivan On 11. mai. 2007, at 18:28, Anthony Minessale wrote: Also make sure you run freeswitch like this when reproducing: NUA_DEBUG=9 TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php

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