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On Tue, 19 Jan 2010, Sam Grabham wrote:
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.
dtmfmode=rfc2833 is what I use. Looks like it ought to be working, but ...Maybe call from a mobile rather than out via your ISDN lines to avoid any confusion?
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=3600
That's probably OK - if it works to call other phones, it's fine.
had following messages when trying different settings but non gave any audio after the connection made.
No audio is typically a NAT issue, but if you said you've called another phone, then... You did answer it, didn't you? :)
- 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'
This is what it looks like for me:
-- Goto (internal,280,1)
-- Executing NoOp("SIP/4698185-0820b588", "Conference Room Conference") in new
stack
-- Executing SetMusicOnHold("SIP/4698185-0820b588", "meetme") in new stack
-- Executing Answer("SIP/4698185-0820b588", "") in new stack
-- Executing MeetMe("SIP/4698185-0820b588", "280|isM") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '280'
-- Playing 'conf-getpin' (language 'en')
at that point she's speaking the script and off it goes...
One thing - I have /:s as the post-fix to the dial-string when using
chan_misdn - that makes it put DTMF in-band which I find is better on
outgoing ISDN calls... Do the dial-string would look like:
mISDN/g:out/08450042620/:sI presume when you dial extension 700 (to call meetme directly) from 213 it works just fine?
And when you plumb the sipgate input to a regular phone it also works just fine...
Other than that I'm at a bit of a loss... it looks like it ought to be doing the right thing...
My sip.conf for the sipgate bit looks like: ; SIP Trunk to sipgate [sipgate.co.uk] [sipgate-out] context=from-sipgate type=friend nat=no host=sipgate.co.uk insecure=port,invite username=4698185 fromuser=4698185 secret=yearight fromdomain=sipgate.co.uk qualify=yes disallow=all allow=ulaw allow=alaw allow=g726 allow=gsm allow=g723 allow=g729 allow=speex allow=ilbc I have nat=no because my box isn't behind NAT. Gordon -- The Mailing List for the Devon & Cornwall LUG http://mailman.dclug.org.uk/listinfo/list FAQ: http://www.dcglug.org.uk/linux_adm/list-faq.html