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User Interface Programmer Job Opportunity
by Anthony Minessale 09 Dec '10

09 Dec '10

If anyone who lives in the US especially [MI WI OK CA] or would like to move to one of those places and is good at UI design. Contact jobs(a)freeswitch.org with a resume. We need some programmers to work with us on the CudaTel UI. Its a demanding yet rewarding job in software development and you get to help with FreeSWITCH as well. Jquery/Js/Comet type skills are the most important. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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drops calls in the first 60 seconds
by Aloysius Lloyd 09 Dec '10

09 Dec '10

Hi All, How to trouble shoot drops calls in the first 60 seconds ? *FreeSWITCH [Public IP / Hosted ] *<---------> Internet <--------> *Linksys + DD-WRT <------> Aastra 9143i IP Phone* Any help appreciated Thanks Lloyd

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TTS volume issue
by ovvenkat 09 Dec '10

09 Dec '10

Hi to All, I need to increase the volume of TTS. I mean, What ever I am getting output from the TTS, Before playing to user, I need to increase the volume. Is it possible?? -- Regards Venkatesan OV.

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NOTE: OS X memory leak info update
by Mario G 09 Dec '10

09 Dec '10

I have been collecting data for Anthony for a month on this problem, but it changed twice after Git updates. There were 2 different leaks and it seems both may be fixed. I am posting for Anthony and anyone with FreeSWITCH on OS X who may want to update: LEAK 1 The Big Leak - noted in jira 2008 FreeSWITCH allocated 2.3 to 2.3 meg per hour with no calls in or out. This required daily restarts. This leak fixed after Nov 22 Git. LEAK 2 After November 22 2010 Git - I updated before I sent the observations since developers want the latest Git tested. Good thing I did, the previous big leak disappeared! However, each in or out call added 400-500k to FreeSWITCH. It was never released in any 24 hour period. LEAK 3? After December 7 Git - Only a small amount (150-200k) is added per call. But.... I also tested in Linux at the same Git level and it did the same thing so this may be normal. I updated the OS X Wiki with the recommended Git version. Mario G

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OS X lower FS idle processor if launchd used
by Mario G 09 Dec '10

09 Dec '10

Note for OS X users and developers: While working on the memory leak problem I noticed that on OS X FreeSWITCH used more processor utilization at idle while running in background mode. The difference was 2.9 percent vs 4.3 percent. I found that if started with Terminal with or without -nc the utilization was lower than using launchd. I discovered that when i added -nc to the FS plist for launchd processor is lowered to the same level as if started from Terminal. I updated the OS X wiki. Mario G

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G729B and mod_com_g729
by Dan-Cristian Bogos 09 Dec '10

09 Dec '10

Hey Guys, When using mod_com_g729 I found out that FreeSWITCH is rejecting G729B in this form. I know about annexa=no parameter, however have seen quite some UAs sending it in this form. Any work-around I can make FreeSWITCH accepting it? My FreeSWITCH version: FreeSWITCH version: 1.0.head (git-) checked out somewhere in the middle of October. mod_com_g729: fsg729-167-installer Debug of such call: 2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4559 Channel sofia/Ip-Switch.Net/7809163901@2.3.4.5 entering state [received][100] 2010-12-07 13:56:05.371029 [DEBUG] switch_core_state_machine.c:318 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Running State Change CS_NEW 2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4570 Remote SDP: v=0 o=- 24240 28749 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 35892 RTP/AVP 18 4 56 a=rtpmap:18 G729B/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3 a=rtpmap:56 telephone-event/8000 a=ptime:30 a=nortpproxy:yes a=nortpproxy:yes a=direction:active 2010-12-07 13:56:05.371029 [DEBUG] switch_core_state_machine.c:324 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State NEW 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G729B:18:8000:30:8000]/[PCMU:0:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [telephone-event:56:8000:30:0]/[PCMU:0:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf send/recv payload to 56 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G729B:18:8000:30:8000]/[PCMA:8:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [telephone-event:56:8000:30:0]/[PCMA:8:8000:20:64000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf send/recv payload to 56 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G729B:18:8000:30:8000]/[G729:18:8000:20:8000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [G723:4:8000:30:6300]/[G729:18:8000:20:8000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec Compare [telephone-event:56:8000:30:0]/[G729:18:8000:20:8000] 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf send/recv payload to 56 2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf send/recv payload to 56 2010-12-07 13:56:05.371029 [DEBUG] switch_channel.c:2457 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Callstate Change DOWN -> HANGUP 2010-12-07 13:56:05.371029 [NOTICE] sofia.c:4776 Hangup sofia/Ip-Switch.Net/7809163901@2.3.4.5 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2010-12-07 13:56:05.371029 [DEBUG] switch_channel.c:2473 Send signal sofia/Ip-Switch.Net/7809163901@2.3.4.5 [KILL] 2010-12-07 13:56:05.371029 [DEBUG] switch_core_session.c:1057 Send signal sofia/Ip-Switch.Net/7809163901@2.3.4.5 [BREAK] 2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:318 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Running State Change CS_HANGUP 2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:539 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State HANGUP 2010-12-07 13:56:05.371971 [DEBUG] mod_sofia.c:456 Channel sofia/Ip-Switch.Net/7809163901@2.3.4.5 hanging up, cause: INCOMPATIBLE_DESTINATION 2010-12-07 13:56:05.371971 [DEBUG] mod_sofia.c:518 Responding to INVITE with: 488 2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:46 sofia/Ip-Switch.Net/7809163901@2.3.4.5 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:539 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State HANGUP going to sleep 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:337 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State Change CS_HANGUP -> CS_REPORTING 2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1057 Send signal sofia/Ip-Switch.Net/7809163901@2.3.4.5 [BREAK] 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:318 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Running State Change CS_REPORTING 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:599 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State REPORTING 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:53 sofia/Ip-Switch.Net/7809163901@2.3.4.5 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:599 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State REPORTING going to sleep 2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:331 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) State Change CS_REPORTING -> CS_DESTROY 2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1057 Send signal sofia/Ip-Switch.Net/7809163901@2.3.4.5 [BREAK] 2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1224 Session 959 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Locked, Waiting on external entities 2010-12-07 13:56:05.372880 [NOTICE] switch_core_session.c:1242 Session 959 (sofia/Ip-Switch.Net/7809163901@2.3.4.5) Ended Ta in advance for any kind of tip! DanB

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Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)
by Peter Olsson 09 Dec '10

09 Dec '10

Hello everyone, I have a question about the reliability of heartbeat events, it should be sent from FreeSWITCH every 20 seconds - right? If you look below in my log sample (from my own process), you can see that it comes 12:07:01, then 21, then 41, but after this it suddenly takes 31 seconds for it to show up (12:08:12). I've also done a Wireshark trace, and it shows the same thing - so it is really delayed from FS. The load at the time was zero calls, and the server itself was using maybe 5% CPU. After the delayed event it continues to show up every 20 seconds. [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive. [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive. [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive. [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive. [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive. [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive. [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive. Should the HEARTBEAT event be 100% reliable, or could something cause a delay like this? This server is not on latest GIT, but it's not more then one month old. I can debug this further myself, but I just want to know if it always should be triggered every 20 seconds, or if these events for some reasons might be delayed. By the way - no clock skew or anything is detected on the machine, so that's not causing it. Thanks, Peter

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sip-t
by budi wibowo 09 Dec '10

09 Dec '10

hi does freeswitch support sip-t protocol? regards budi

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get error on skypopen module
by ha do 09 Dec '10

09 Dec '10

Hi list i setup freeswitch and skypopen running fine there are 2 skype clients run on freeswitch freeswitch@internal> sk list sk console is NOT yet assigned F ID Name IB (F/T) OB (F/T) State CallFlw UUID = ==== ======== ======= ======= ====== ============ ====== 1 [interface1] 0/1 3/7 IDLE IDLE 2 [interface2] 0/5 1/5 IDLE IDLE the skype clients are used the same username + password of skype account the skypopen works fine but i get error below in the debug mode 2010-12-07 04:24:35.794611 [DEBUG] switch_core_state_machine.c:462 (skypopen/interface1) State DESTROY going to sleep 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173 [|] [DEBUG_SKYPE 173 ][interface1 ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed||| 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228 [|] [DEBUG_SKYPE 228 ][interface1 ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed||| 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173 [|] [DEBUG_SKYPE 173 ][interface1 ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed||| 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228 [|] [DEBUG_SKYPE 228 ][interface1 ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed||| How to fix it Thank you Ha`

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No sound on incomming calls
by Daniel Bareiro 09 Dec '10

09 Dec '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm new in the list and I'm starting to test FreeSWITCH at home to learn how to use it thinking as a alternative to Asterisk. I've compiled FreeSwitch 1.0.6 and I have been doing some testing with Iptel.org and FreeSWICH. Outgoing calls to Iptel.org accounts ran without problems, however I am having trouble with incoming calls. If, for example, I call from a softphone registered on Iptel.org with a second account to the Iptel.org account that was registered from FreeSWITCH, it answers the call but no sound. The FreeSWITCH server is behind a firewall, but I do not think it's a NAT problem because then there would be no sound for outgoing calls made with the softphone registered on Iptel.org. Could it be a codecs problem? When FreeSWITCH receives the call, it is redirected to a Grandstream BT200 phone. To answer the call in the BT200, the softphone indicates that G711a is being used, but no sound. Thanks in advance for your replies. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAk0ACREACgkQZpa/GxTmHTd9GQCcCoM4TzkSfZyVPthU7v9hzqw4 aSgAn1rg4mjSqj8yzIGE77NDO5oOXKPq =+GKJ -----END PGP SIGNATURE-----

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