- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Spontaneous reboots of Polycom and Linksys phones
by Wellie Chao 26 Mar '10

26 Mar '10

I am using Freeswitch and I have some phones that spontaneously reboot (often in the middle of a call). I did some searching around on the web, but didn't find any information that really applied to the problem I'm experiencing. I have a few dozen Polycom phones and a handful of Linksys phones of various models: Polycom IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921, SPA941. I also have snom phones, but they don't reboot spontaneously. It's only the Polycom and Linksys phones that reboot spontaneously, and it seems to be correlated to call volume. Heavy usage of the Polycom and Linksys phones causes them to reboot more frequently (i.e. shorter duration between the reboots). For people making 50 calls a day, the Polycom phones spontaneously reboot every three days or so (and always on a call, it seems). For people making one or two calls a day, the Polycom will go months without a spontaneous reboot. This correlation between call volume and reboot frequency leads me to believe that there is some bug in the Polycom firmware, perhaps some of overflow condition that accumulates over time and eventually causes a crash (and subsequent reboot). I was wondering if anybody else experienced something similar, and whether anyone had developed a solution. One thing I am considering is to set up a script to reboot the Polycom phones in the middle of the night every day, which would hopefully prevent them from spontaneously rebooting during the day by preventing the phones from ever reaching the overflow condition (if that is indeed what causes the reboots). A nightly reboot would be a hackish way of fixing the problem, so I'd also love to hear if anybody has any other speculations on what might be causing the spontaneous reboots. We have phones in multiple buildings, and we've discovered that the spontaneous reboots happen on phones on different networks, so it's not specific to a particular LAN segment. For the most part, we are using the standard config files for the Polycom phones and for the Linksys phones.

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call info display
by Ken Gillett 26 Mar '10

26 Mar '10

Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension. When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what). Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories? So, how does FS handle all the above? What does it pass to the recipient, for both external and internal calls? Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension? Ken G i l l e t t _/_/_/_/_/_/_/_/

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Guidance on debugging audio glitches
by Chris Fowler 25 Mar '10

25 Mar '10

Hi, Here's the problem I'm "hearing": Test Case #1 x1001 calls x1002 - Audio is rock solid Test Case #2 x1001 calls x3001 (Conference Bridge) x1002 calls x3001 (Conference Bridge) - Audio is mostly perfect. However, occasional "glitches" in the audio can be heard Notes: - Bypass media is switched off. FreeSWITCH routes audio for both the bridged, and the conferenced call - rtp-autoflush-during-bridge is "true" on Sofia profile internal.xml - I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see occasional auto flush of rtp packets for *both* test cases. The glitches don't coincide with the flushing. In fact with flushing you can't hear when data gets flushed (2x20ms packets). Environment - CentOS 5.4 - Tested on AWS EC2 Small, Medium, Large instances. On Medium and Large CPU affinity for FreeSWITCH set to the 2nd core. Questions: - What's different between bridging a call and a conference call run through mod_conference in terms of audio processing? - Thoughts on how to debug this? Thanks, Chris.

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Disabling mod_xml_rpc
by Joseph L. Casale 25 Mar '10

25 Mar '10

If the web interface is not needed, can this module be left unloaded or do other services depend on it? Thanks, jlc

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