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Hi Tried the sample code below, but still couldn't get it to work.When you watch the console you see the call getting connected, but you don't hear any voice messages and connection times out.
i wondered if it had anything to do with dtmfmode=rfc2833 and tried other mode but non worked.
sip.conf [1042620] ;Telephone-No.08450042620 ;type=friend ;peer type=peer insecure=very ; otherwise I get authentication errors nat=Yes username=1042620 authuser=1042620 fromuser=1042620 fromdomain=sipgate.co.uk secret=XXXXXXXX ;host=sipgate.co.uk ;outboundproxy=sipgate.co.uk ;qualify=no qualify=yes ;dtmfmode=info dtmfmode=rfc2833 context=incoming-sipgate ;canreinvite=no disallow=all allow=alaw allow=ulaw ;allow=gsm ;allow=g729 ;allow=slinear ;maxexpiry=3600had following messages when trying different settings but non gave any audio after the connection made.
- Called g:out/08450042620
-- mISDN/1-u5 is proceeding passing it to SIP/213-08204ca8
-- Executing Answer("SIP/sipgate.co.uk-0820e108", "") in new stack
-- Executing MeetMe("SIP/sipgate.co.uk-0820e108", "700|isM") in new
stack
== Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '700' -- Playing 'conf-getpin' (language 'en') -- mISDN/1-u5 is ringing -- mISDN/1-u5 answered SIP/213-08204ca8 -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en')== Spawn extension (default, 908450042620, 3) exited non-zero on 'SIP/213-08204ca8'
-- Hungup 'Zap/pseudo-1478991532'== Spawn extension (default, 1042620, 2) exited non-zero on 'SIP/sipgate.co.uk-0820e108'
*************************************************************-- Executing Dial("SIP/213-08204ca8", "misdn/g:out/08450042620") in new stack
-- Called g:out/08450042620
-- mISDN/1-u6 is proceeding passing it to SIP/213-08204ca8
-- Executing Answer("SIP/sipgate.co.uk-0820e408", "") in new stack
-- Executing MeetMe("SIP/sipgate.co.uk-0820e408", "700|isM") in new
stack
== Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '700'***************** -- Executing Set("SIP/213-082161c0", "CALLERID(num)=881913") in new stack -- Executing Set("SIP/213-082161c0", "CALLERID(name)=Software Science") in new stack -- Executing Dial("SIP/213-082161c0", "misdn/g:out/08450042620") in new stack
-- Called g:out/08450042620 -- mISDN/1-u21 is proceeding passing it to SIP/213-082161c0-- Executing Goto("SIP/sipgate.co.uk-0821b700", "internal|700|1") in new stack
-- Goto (internal,700,1)-- Executing SetMusicOnHold("SIP/sipgate.co.uk-0821b700", "meetme") in new stack
-- Executing Answer("SIP/sipgate.co.uk-0821b700", "") in new stack
-- Executing MeetMe("SIP/sipgate.co.uk-0821b700", "700|isMprD") in
new stack
-- Playing 'conf-getpin' (language 'en') -- mISDN/1-u21 is ringing -- mISDN/1-u21 answered SIP/213-082161c0 -- Created MeetMe conference 1023 for conference '700'> Starting recording of MeetMe Conference 700 into file meetme-conf-rec-700-1263911652.142.wav.
-- Recording -- Playing 'vm-rec-name' (language 'en') -- Playing 'beep' (language 'en')-- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-700-1 format: sln, 0x81ee380
-- Took too long, cutting it short... -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en')== Spawn extension (default, 908450042620, 3) exited non-zero on 'SIP/213-082161c0'
-- Hungup 'Zap/pseudo-392673834'== Spawn extension (internal, 700, 3) exited non-zero on 'SIP/sipgate.co.uk-0821b700'
**************** -- Playing 'conf-getpin' (language 'en') -- mISDN/1-u6 is ringing -- mISDN/1-u6 answered SIP/213-08204ca8 -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en')== Spawn extension (default, 908450042620, 3) exited non-zero on 'SIP/213-08204ca8'
-- Hungup 'Zap/pseudo-356181767'== Spawn extension (default, 1042620, 2) exited non-zero on 'SIP/sipgate.co.uk-0820e408'
************************************************************************-- Executing Dial("SIP/213-08204ca8", "misdn/g:out/08450042620") in new stack
-- Called g:out/08450042620
-- mISDN/1-u7 is proceeding passing it to SIP/213-08204ca8
-- Executing Answer("SIP/sipgate.co.uk-08210058", "") in new stack
-- Executing MeetMe("SIP/sipgate.co.uk-08210058", "700|isM") in new
stack
== Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '700' -- Playing 'conf-getpin' (language 'en') -- mISDN/1-u7 is ringing -- mISDN/1-u7 answered SIP/213-08204ca8 -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en')******************** -- parse_srv: SRV mapped to host sipgate.co.uk, port 5060
-- Executing SetMusicOnHold("SIP/213-082078c8", "meetme") in new stack
-- Executing Answer("SIP/213-082078c8", "") in new stack
-- Executing MeetMe("SIP/213-082078c8", "700|isMprD") in new stack
-- Playing 'conf-getpin' (language 'en')
-- Created MeetMe conference 1023 for conference '700'
> Starting recording of MeetMe Conference 700 into file
meetme-conf-rec-7
00-1263903894.87.wav.
-- Recording -- Playing 'vm-rec-name' (language 'en') -- Playing 'beep' (language 'en')-- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-700-1 form at: sln, 0x81e7c00
-- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Playing 'vm-msgsaved' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Started music on hold, class 'default', on SIP/213-082078c8**********************
any ideas were to start debugging my prob? Regards Sam Gordon Henderson wrote:
On Mon, 18 Jan 2010, Sam Grabham wrote:Hi Has anyone ever used a Sipgate connection to a MeetMe conf connection?Yes....I had thought it was going to be as easy as a Dial(<the meeting No>), but no, that doesn't work for me. If you do a dial to a valid internal SIP number then works fine, but no to a mettme.I've removed the commented out lines:[incoming-sipgate] exten => 1042620,1,Answer() exten => 1042620,n,MeetMe(700|cdM|1234)I'd drop the 'd' and make sure I had a conference defined in meetme.conf. I've never gotten on with dynamic conferences myself.exten => 1042620,n,Playback(vm-goodbye)I don't think this ever will be called.exten => 1042620,n,Hangup()Unneeded, but ...Do you have a timing source in your system - either ztdummy/dahdi_dummy or a TDM card of some sorts? I think it's essential to make MeetMe work.I've just added one to one of my systems - you can call it via sipgate on 4698185 if you want...meetme.conf has: [rooms] ; Conference conf => 280,1234,1234 and the dialplan looks like: ; Conference exten => 280,1,Noop(Conference Room Conference) exten => 280,n,SetMusicOnHold(meetme) exten => 280,n,Answer() exten => 280,n,MeetMe(280,isM)However, I have a jump from the sipgate input to this extension which looks like:; 4698185 -> 280 [Conference]exten => 4698185,1,Noop(DDI Incoming on 4698185 to 280 - Conference - Prefix is SG); irrelevant stuff deleted exten => 4698185,n,Goto(internal,280,1) So in reality, it's not much different from yours. What's the console output when you set verbose 4 like?Also has anyone used a GUI writer for Asterisk?I came across a tool at http://www.apstel.com, but it's not free, it's the first one i have ever came across.Hm. Cute. Lego for asterisk...I wrote my own webby/phpy front-end to my own dial-plan. Have heard of these things, but I'm not convinced they're really neccessary. 95% of all the systems I sell/install are more or less the same configuration.Gordon
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