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Question about execute_on_answer
by Davide Colombo 14 Jun '10

14 Jun '10

Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: <action application="export" data="nolocal:execute_on_answer=transfer myivrpresentation XML default"/> <action application="bridge" data="user/1019@${domain_name}"/> When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/1000(a)fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards

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Two bridged FreeSWITCH servers - both in bypass_media mode - possible?
by Vitalii Colosov 14 Jun '10

14 Jun '10

Hi, I am trying to make 2 freeswitches (bridged) to work both in bypass media mode - currently with no luck. Would appreciate your advice. Let say I have 2 FreeSWITCH servers - one acts as PROXY (in bypass_media_after_bridge mode) and second processes media. I make a call from one sip client to another via these 2 servers (let's say G711 is used everywhere). SIP CLIENT1 -> FS1 (proxy) -> FS2 (media) -> SIP CLIENT2 Proxy server (FS1) sends a re-INVITE and excludes itself from the media path after the call was answered (I am using bypass_media_after_bridge). So, the media goes like this: SIP CLIENT1 <-> FS2 <-> SIP CLIENT2 Now, I want to try to exclude the media server (FS2) as well (just experimenting, but if both clients use same codec, I suppose this can be legitimately done?). I put bypass_media_after_bridge=true on the media server (I did it in few places in Lua script). *But as per sip trace, it does not sends any reinvite after the call was answered.* *Only FS1 sends reINVITE.* So, looks like I cannot remove FS2 from media path like this. Do you think it is not possible to acheive this kind of configuration so both servers will work in bypass media? Thank you, Vitalie

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How to limit call attemp time?
by Sergey Okhapkin 14 Jun '10

14 Jun '10

Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.

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odbc on windows
by babak yakhchali 14 Jun '10

14 Jun '10

Hi I configured my sofia profile internal to use odbc. it connects and creates tables but after that I get lots of errors like bellow these are just a few of them: 2010-06-06 14:09:04.000000 [ERR] switch_odbc.c:427 ERR: [BEGIN] [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Inc orrect syntax near 'BEGIN'. ] 2010-06-06 14:09:04.000000 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 C ODE 8180 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Statement(s) coul d not be prepared. ] thanx all

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Dropped Call During VM Email Delivery?
by Jerry Richards 14 Jun '10

14 Jun '10

We getting dropped calls recently. They seem to appear in the vicinity where mod_voicemail is delivering a voice mail Email (at least that's where we've seen them). Below, I show a snippet from the Freeswitch log that shows two calls (on extensions 1028 and 1063) getting simultaneously disconnected. Do you know what might be the cause? 2010-06-14 08:59:02.495697 [DEBUG] mod_voicemail.c:2394 Deliver VM to 1036(a)192.168.72.141 2010-06-14 08:59:02.500634 [DEBUG] ozmod_sangoma_boost.c:1174 1:23 STATE [DOWN] 2010-06-14 08:59:02.500634 [WARNING] sangoma_boost_client.c:221 TX EVENT (N): CALL_STOPPED_ACK:(86) [w1g23] Rc=0 CSid=0 Seq=766 2010-06-14 08:59:02.500634 [DEBUG] zap_io.c:1388 channel done 1:23 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1028@192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1063@192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1028@192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1063@192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1028@192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:1/18007297580@g1 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1032@192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1032@192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1032@192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:3/19403232644@g1 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1063@192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:2/14253491003@g1 [BREAK] 2010-06-14 08:59:04.501359 [DEBUG] switch_utils.c:631 Emailed file [/tmp/mail.12765311428da7] to [chuck.wichser(a)teotech.com] 2010-06-14 08:59:04.501359 [DEBUG] mod_voicemail.c:2562 Sending message to chuck.wichser(a)teotech.com Best Regards, Jerry

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major problem with latest version
by Christian Löschenkohl 14 Jun '10

14 Jun '10

hello i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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LED-Controlling on Snom phones. Missing Subject header
by Helmut Kuper 14 Jun '10

14 Jun '10

Hello, I try to control the LEDs on Snom 370 phones using LUA and event type SEND_MESSAGE. The phone receives this message: MESSAGE sip:2850@85.16.245.213:1051 SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKeer7H9cFa85DN Max-Forwards: 70 From: <sip:2850@85.16.246.6>;tag=yH2p954XZZ1tj To: <sip:2850@85.16.246.6> Call-ID: 2a6ed9ba-ef6d-122d-3d83-00144fe6e332 CSeq: 131979517 MESSAGE Contact: <sip:mod_sofia@85.16.246.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/x-buttons Content-Length: 10 k=1 c=on and Snom acknowledges this with 200 OK. Unfortunately nothing happens. Snom's howto (http://wiki.snom.com/Category:HowTo:LED_Remote_Control) describes a message like this: MESSAGE sip:44@172.20.25.101:2048;transport=tls;line=44qsudyt SIP/2.0 Via: SIP/2.0/TLS 172.20.25.102:5061;branch=z9hG4bK-2909fe081f8466c9836c3b673f284842;rport From: <sip:44@localhost>;tag=17820 To: <sip:44@localhost> Call-ID: umceg9jb@pbx CSeq: 35160 MESSAGE Max-Forwards: 70 Contact: <sip:172.20.25.102:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 7 k=3 l=42 Main difference is the subject header containing "buttons" (despite of the body) Is there a way to add subject header as above to the message generated by FS? regards Helmut

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help about a system with login logout
by babak yakhchali 14 Jun '10

14 Jun '10

Hi I need to implement a system in which users should login and logout to be able to place or receive calls using cisco 7941 ip phones on freeswitch (first I tried to somehow program one of soft keys as login and logout shortcut but I couldn't do that). Any suggestions would be appreciated. thanx

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Anti-tromboning in FreeSWITCH?
by Ron Avriel 14 Jun '10

14 Jun '10

Hi, Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? My scenario is similar to image in link above. Currently, when user A calls B I get two calls and media passing twice through FS. Is there any way for media to pass directly between A and B? Thanks, Ron

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position of members in the fifo
by Abubacker siddiq 14 Jun '10

14 Jun '10

Dear all, I am working in fifo using outbound sockets in that my requirement is to say the queue number for the waiting customer for every '5' seconds, I tried this with fifo_position variable but the issue is when ever the customer calls has been answered and hangup then also the queue number is not decremented , is it a correct behaviour ? I can do this using fifo list and fifo list_verbose by parsing its xml structure , but I want to know whether we have a easy way to achieve this. Thanks in advance !

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