March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

One Way Bridge Channels
by Victor Toofic 26 Mar '10

26 Mar '10

Hi! Does anybody know if there is a way to "bridge" an answered channel to a channel that is in "early media" state? The scenario is like this: Channel A Channel B -------------------> FS -------------> (pre_answer, park) (answer) <---<---<---<---<--- Audio <---<---<--- When the channel A arrives I pre_answer and park it, so it is in "early media" state. After that I originate the channel B. When the channel B answers I just want to relay the audio coming from B to A, I don't care if A can't talk to B at that moment. Is there a way to achieve this? Some ideas? Thnks! - Regards, Víctor Toofic

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Installing DAHDI compiled with Oslec from RPMs in wiki
by Joseph L. Casale 26 Mar '10

26 Mar '10

Hey guys, I just added my notes to the wiki: http://wiki.freeswitch.org/wiki/Installing_DAHDI_compiled_with_Oslec_from_R… Maybe some of the more seasoned guys can have a quick look for the sake of accuracy. I have had this running for a couple days with good results. jlc

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Attention all FreeSWITCH and Openzap committors. ACTION REQUIRED
by Michael Jerris 26 Mar '10

26 Mar '10

We are in process of our migration from svn to git for our main source control. We will likely still have a read only svn mirror, but all future commits will go to git after the migration. In order to proceed with this process, I will need your full name, public key, email address, and svn username. If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo. For information on how to use git: http://git-scm.com/documentation , http://git-scm.com/course/svn.html For information on how to generate a public key if you do not already have one: http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html For general users, more information will follow on these changes. For now, nothing has changed. Around April 1, there will likely be a 1 time need to do a full checkout if you plan on continuing to use the svn mirror, or a fresh git checkout if you plan on switching to git. If you have any other questions about the process or changes, feel free to ask. Mike

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FreeSWITCH Pub Conf is bridged to the VUC conference
by Michael Collins 26 Mar '10

26 Mar '10

Feel free to call in to the FS public conference to listen in as the FS team and the VUC gang talk tech! SIP: 888(a)conference.freeswitch.org PSTN: +1-919-386-9900 VUC IRC channel: #vuc Come join us! -MC

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Sofia crashing when registering
by Spencer Thomason 26 Mar '10

26 Mar '10

Hello all, I use Bandtel as one of my trunking providers and Freeswitch will work flawlessly for 24-36 hours and then stop responding to SIP requests. I have monit setup to send an options ping once a minute and restart the process if it doesn't get a reply. This has been working ok but the Freeswitch process has to get restarted about once a day on average. Has anybody else had any issues like this? They are using fairly standard issue stuff, Freeswitch registers to SER 0.9.6 and they are using Sippy as a B2BUA on their end. The log shows a call ending, a period of no activity, and Freeswitch attempting to register repeatedly, the process then stops responding to requests and monit restarts Freeswitch. 2010-03-24 17:31:03.897756 [DEBUG] switch_core_state_machine.c:439 (sofia/default/13105106983(a)208.70.78.37 ) State DESTROY going to sleep 2010-03-24 17:33:16.353544 [NOTICE] sofia_reg.c:336 Registering bandtel 2010-03-24 17:37:58.452249 [NOTICE] sofia_reg.c:336 Registering bandtel 2010-03-24 17:42:40.588167 [NOTICE] sofia_reg.c:336 Registering bandtel 2010-03-24 17:47:22.529220 [NOTICE] sofia_reg.c:336 Registering bandtel 2010-03-24 17:53:15.899570 [INFO] switch_core.c:1734 Shutting down Here is a typical register request. ------------------------------------------------------------------------ 2010-03-24 21:12:59.955693 [NOTICE] sofia_reg.c:336 Registering bandtel send 772 bytes to udp/[66.237.65.67]:5060 at 04:12:59.956245: ------------------------------------------------------------------------ REGISTER sip:66.237.65.67 SIP/2.0 Via: SIP/2.0/UDP 208.70.78.36;rport;branch=z9hG4bKZcD7H9aN1D50j Max-Forwards: 70 From: <sip: 201009000001(a)66.237.65.67;transport=udp>;tag=3yZ6ZSZUFj3gQ To: <sip:201009000001@66.237.65.67;transport=udp> Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175 CSeq: 128622152 REGISTER Contact: <sip:gw+bandtel@208.70.78.36:5060;transport=udp;gw=bandtel> Expires: 390 User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, NOTIFY Supported: timer, precondition, path, replaces Authorization: Digest username="201009000001", realm="10.1.2.1", nonce="4baabf0e1e68a1fdfbde46913ca850da0cbecc8a", algorithm=MD5, uri="sip:66.237.65.67 ", response="3974689c5cb3eeb4f09e634716d2239e" Content-Length: 0 ------------------------------------------------------------------------ recv 505 bytes from udp/[66.237.65.67]:5060 at 04:12:59.980247: ------------------------------------------------------------------------ SIP/2.0 200 OK From: <sip: 201009000001(a)66.237.65.67;transport=UDP>;tag=3yZ6ZSZUFj3gQ To: <sip: 201009000001 @66.237.65.67;transport=UDP>;tag=2e41cc73d384c8f458f34dd410d7744b-b5e6 Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175 CSeq: 128622152 REGISTER Via: SIP/2.0/UDP 208.70.78.36;rport=5060;branch=z9hG4bKZcD7H9aN1D50j PortaBilling: currency:USD Server: Sip EXpress router (0.9.6 (i386/freebsd)) Contact: <sip:gw+bandtel@66.237.65.67:5060;gw=bandtel>;expires=385 Expires: 385 Content-Length: 0 ------------------------------------------------------------------------ 2010-03-24 21:12:59.979821 [DEBUG] sofia_reg.c:1478 Changing expire time to 385 by request of proxy sip:66.237.65.67 What is the best way to debug this? It seems strange that everything works for a while and then stops. Thanks, Spencer -- Spencer Thomason 5Nine Solutions LLC e. sales(a)5ninesolutions.com p. +1.888.271.7959 f. +1.310.510.6980

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Thomson ST2030 IP Phones
by Dave Stevenson 26 Mar '10

26 Mar '10

Hi, is anyone using Thomson ST2030 phones with FreeSwitch ? I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite a bit of noise (hiss) on the line and some breakup of the voice. The phones are all on a private LAN behind a router - all external calls are over the PSTN - no external IP calls are made (I don't use a VOIP provider). I have loaded the latest Thomson SIP firmware (2.69). There are a couple of options on the phone that I've tried such as Silence Suppression, Acoustic Echo Cancellation and Packet Loss Compensation with no improvement. Most of the other options are beyond me I'm afraid, so I'd be really interested to hear from anyone who has good (or even acceptable !) results from the ST2030 - particularly if they could share their phone configuration regards Dave

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Returning the ip address of a registered user
by Kristian Kielhofner 26 Mar '10

26 Mar '10

Hello everyone, What's the easiest/lightest way to return just the IP address and port of a registered user? So far I can see using function_sofia_contact and parsing that (not preferred because it won't detect NAT) OR executing the SQL lookups from FS and returning the IP and port as known to FreeSWITCH. Any ideas? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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ClueCon MMX - Call For Speakers!
by Michael Collins 26 Mar '10

26 Mar '10

Hello everyone! We are gearing up for ClueCon 2010 in August later this year. We are making the necessary arrangements for the conference facilities and rooming. Things are beginning to fall into place. Now we need to hear from you. We would like to put out a call for speakers for this year's event. Please contact us if you or your organization would like to give a presentation at ClueCon this year. We want to get the speakers scheduled as early as possible. Keep in mind that those organizations which sponsor ClueCon will be given the highest priority when it comes to scheduling. Please contact Brian West to discuss sponsorship opportunities for this year's event. We also would like to hear from the conference attendees: what would you like to see this year? Please give us your input. ClueCon is, of course, "By Developers, For Developers." However, developers come in all shapes and sizes and we would like have something for everyone. Please tell us what would make ClueCon MMX the best conference of the year! Stay tuned for more announcements. We look forward to hearing from you and seeing everyone this August in Chicago. -ClueCon team

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How To: Sip Logging and Freeswitch
by OwenW 26 Mar '10

26 Mar '10

This is a quick guide as to one approach to logging the sip messages between a Sip phone and freeswitch. There are other methods (e.g. wireshark) - feel free to reply if you have other alternatives. It involves putting a java TCP Proxy (grinder) in front of freeswitch and reconfiguring your SipPhone to send SIP to the proxy. The software (grinder) is Java based so will require an JDK. Once you've downloaded Grinder create a file called startProxy5061.cmd and put the following into it: call setGrinderEnv.cmd java -cp %CLASSPATH% net.grinder.TCPProxy -console -localhost 192.168.30.10 -localport 5061 -remotehost 192.168.30.10 -remoteport 5060 > freeswitch_sip.log The proxy forwards all traffic it receives on 5061 to 5060 and logs to freeswitch_sip.log. You'll need to change the IPs above. Create another file called setGrinderEnv.cmd and put the following in to it: set GRINDERPATH=C:\dev\apps\grinder-3.3 set GRINDERPROPERTIES=C:\dev\apps\grinder-3.3\grinder.properties set CLASSPATH=%GRINDERPATH%\lib\grinder.jar;%CLASSPATH% set JAVA_HOME=c:\Program Files\java\jdk1.6.0_11 PATH=%JAVA_HOME%\bin;%PATH% You'll need to change the paths above. The soft phone will need to be configured to use port 5061 instead of 5060 as the proxy. Owen -- View this message in context: http://n2.nabble.com/How-To-Sip-Logging-and-Freeswitch-tp4802944p4802944.ht… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to answer USER_BUSY on NO_USER_RESPONSE?
by Peter P GMX 26 Mar '10

26 Mar '10

I have the folowing scenario: A Snom phone has activated DND (do not disturb) When the phone is called, it returns "480 Do Not Disturb" and then "404 Not Found". Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN). The PSTN then (wrongly) announces "this number is not listed". My question: How can we overcome this and announce back a different message to PSTN, but only on NO_USER_RESPONSE. I tried to continue on bridge and then this after the bridge command the following: <condition field="variable_originate_disposition" expression="NO_USER_RESPONSE" inline="true"> <action application="hangup" data="USER_BUSY"/> </condition> But the installed Freeswith does not support the "inline" yet and we will not be able to update Freeswitch soon. So my question: Is there any other way to achieve this? Best regards Peter

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