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Questions of packages loss
by joy this 18 Mar '11

18 Mar '11

I have some tests for the problem that the voice keeps off and on during the call. I use Wireshark to check the package flow on the caller, callee, and FS sides. The network structure: caller(PCMA)---(Wi-Fi)--->WAN--->FS--->callee(PCMA). FS and the callee are under the same router. The result is: The caller sends 2688 packages to FS, and FS receives 2687 packages. Then FS sends only 2380 packages to the callee, and the callee receives 2380 packages. There is about 11.4% package loss in FS, and the callee hears incomplete voice. I have some questions: 1. I guess FS drops packages from the caller because these packages arrived too late. Is it reasonable? If not, what is the reason? 2. I want to check if FS drops packages because of the late arrival. Could you tell me the packages flow in the source code? 3. I set the jitterbuffer_msec from 10 to 10000, but it seems not effective. Wouldn’t the jitterbuffer_msec affect the result? Sincerely yours, thisjoy.

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very long delays in conference audio
by covici＠ccs.covici.com 18 Mar '11

18 Mar '11

Hi. I have been experiencing very long delays in conference audio -- usually over the internet. For instance, it can take 1 second or more between the time you hit 0 and when it says you are muted or unmuted. I even had this problem during the freeswitch conference. What can be done about this -- any way to tell why this is happening, because it makes conversation on the calls very difficult. I had one last night where they gave up because of this problem. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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mod_conference question
by DJB International 18 Mar '11

18 Mar '11

I wonder whether there is a way to get Conference-Unique-ID to show up when the channel is hang up in CS_REPORTING state. I am trying to find a way to separate and group the cdrs for each conference session. Thank you, -djbinter

1 0

segfault with stun-enabled=false
by Frank Rosengart 17 Mar '11

17 Mar '11

Hi, I'm trying to run local tests with FS and need to switch off all stun stuff. When stun-enabled=false in my sip profile, FS dies when starting mod_sofia. I think it's more user friendly to exit with an error message saying what is wrong. FS git/trunk on Linux/amd64. How can I force FS not to use stun at all? Thanks, Frank

2 1

Multiple profile on same ports
by awais nazir 17 Mar '11

17 Mar '11

Hello I have intermediate level of experience in freeswitch. I want to achieve a scenario that multiple sip-profiles can share 1 sip bindport 5060. And sip profile can be attached to a call in using pattern matching in incoming calls. --Waisee

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Unexpected out of order command/reply '+OK event listener enabled plain' in perl ESL outbound
by Dmitry Sytchev 17 Mar '11

17 Mar '11

In my outbound async full ESL script after some moment there are out-of-order message on every "execute" call, and this breaks my event handling in some way. Content-Type: command/reply.Reply-Text: +OK event listener enabled plain.. I want to execute some commands syncronously, but setEventLock seems to be broken by this behavior. As the workaround I had to write lock_for_execute_complete function, which waits for execute completion of applications that need to be run sync, but it effectively destroys async events by processing them while waiting for CHANNEL_EXECUTE_COMPLETE. here is a part of socket dump, note second "Content-Type: command/reply.Reply-Text: +OK event listener enabled plain.." right after execute. There is no event command in script besides the very beginning, so why these messages appear? T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a pbx_subscriber_customer_id 1 # T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] .. ## T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] Content-Type: api/response.Content-Length: 4.. # T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] +OK. ## T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] api sofia_contact s000002(a)asdasd.com # T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] .. ## T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] Content-Type: api/response.Content-Length: 96.. # T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] sofia/local/sip:s000002@172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061 ## T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length: 96..sofia/local/sip:s000002@172.19.36.54:5061;fs_nat= yes;fs_path=sip%3As000002%40172.19.36.54%3A5061 # T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP] sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg: {originate_timeout=10,hangup_after_bridge=false, ignore_early_media=false}user/s000002(a)asdasd.com # *T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] Content-Type: command/reply.Reply-Text: +OK event listener enabled plain.. * # T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP] Content-Type: command/reply.Reply-Text: +OK.. -- Best regards, Dmitry Sytchev, IT Engineer

3 8

Already hung channels in 'show channels' output never go away
by Dmitry Sytchev 17 Mar '11

17 Mar '11

Hi All I'm using Perl ESL outbound script to bridge incoming call to sip endpoint. I'm doing execute("bridge","sofia/user/somebody"), then processing events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result. Everything works fine, but original incoming call channel is never removed from list: After few calls I see original incoming channels in 'show channels' output: 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11 12:32:27,1299835947,sofia/external/xxxx@85.x.x.199 ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,, xxx.xxx.ru,,,HANGUP,,,, 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11 12:32:50,1299835970,sofia/external/xxxx@85.x.x.199 ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,, xxx.xxx.ru,,,HANGUP,,,, 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11 12:33:29,1299836009,sofia/external/xxxx@85.x.x.199 ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,, xxx.xxx.ru,,,HANGUP,,, Also, when I try to stop freeswitch i see these messages on console: 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s) 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s) 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s) 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s) 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s) 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s) 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s) 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s) 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s) Why these channels are not removed from list? I also noticed that memory consumption by freeswitch process constantly grows call by call. -- Best regards, Dmitry Sytchev, IT Engineer

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Freeswitch database polling
by Yehavi Bourvine 17 Mar '11

17 Mar '11

Hello, A question for curiosity: We use ODBC over MySQL to serve the FS core database. This database is replicated (by MySQL means) to a backup system which runs MySQL also. There is no cluster between the two, only one-way MySQL replication. The FS on on the other system shows an updated current status of all extensions registered (sofia status profile XXX). How does it do so? a timely update/polling of the database? If so, what is the interval, and doesn't it make much load on MySQL database when there are a lot of registered phones? Thanks! __Yehavi:

2 1

Mute / Unmute with Freeswitch Conference
by sharad 17 Mar '11

17 Mar '11

Hi friends As of now when we mute a participant, conference server mutes that participant while RTP keeps on coming from that UA. This consumes the bandwidth & precessing time also is there. This issue elobrates when no. of conference participants are increased. So I am just wondering whether we can mute the conference participants using the SIP SDP `reconly' & whenever we wants to unmute him, FS should again send the INVITE SDP with `sendonly'. Plz let me know if it is feasible. Thanks in advance. regards Sharad

1 0

Want more FS English Sounds?
by Avi Marcus 17 Mar '11

17 Mar '11

mercutioviz says that FS is putting in an order for more sounds... is there a stock sound (not something custom for yourself..) that you think FS would benefit by having? List it here! http://wiki.freeswitch.org/wiki/Sounds_order -Avi

2 1