Tony Davidson
2005-Nov-16 20:02 UTC
[Asterisk-Users] Asterisk drops call when calling other VOIP
I'm having an issue when Asterisk calls what I believe to be other VOIP
connections.
I can call the number from a normal sip phone, but when I attempt to
connect via Asterisk the call is dropped immediately. Checking my call
logs I can tell the call has connected but I think Asterisk is trying
something when it connects that immediately causes a dropout. My VOIP
connection is via a SIP account.
Tony
The log of the call is:
-- Called engin/03XXXX0888
-- SIP/engin-f91f answered SIP/203-add5
== Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'SIP/203-add5' in macro 'dialout-trunk'
== Spawn extension (from-internal, 0392210888, 1) exited non-zero on
'SIP/203-add5'
-- Executing Macro("SIP/203-add5", "hangupcall") in new
stack
-- Executing ResetCDR("SIP/203-add5", "w") in new stack
-- Executing NoCDR("SIP/203-add5", "") in new stack
-- Executing Wait("SIP/203-add5", "5") in new stack
-- Executing Hangup("SIP/203-add5", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/203-add5' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/203-add5'
asterisk1*CLI>
Tom Vile
2005-Nov-16 22:39 UTC
[Asterisk-Users] Asterisk drops call when calling other VOIP
its possible that your provider is not setup to use asterisk for your account. I know a some providers that need to know if you are using a regular SIP phone or Asterisk. On 11/16/05, Tony Davidson <tony@davidson.id.au> wrote:> I'm having an issue when Asterisk calls what I believe to be other VOIP > connections. > > I can call the number from a normal sip phone, but when I attempt to > connect via Asterisk the call is dropped immediately. Checking my call > logs I can tell the call has connected but I think Asterisk is trying > something when it connects that immediately causes a dropout. My VOIP > connection is via a SIP account. > > Tony > > The log of the call is: > > -- Called engin/03XXXX0888 > -- SIP/engin-f91f answered SIP/203-add5 > == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on > 'SIP/203-add5' in macro 'dialout-trunk' > == Spawn extension (from-internal, 0392210888, 1) exited non-zero on > 'SIP/203-add5' > -- Executing Macro("SIP/203-add5", "hangupcall") in new stack > -- Executing ResetCDR("SIP/203-add5", "w") in new stack > -- Executing NoCDR("SIP/203-add5", "") in new stack > -- Executing Wait("SIP/203-add5", "5") in new stack > -- Executing Hangup("SIP/203-add5", "") in new stack > == Spawn extension (macro-hangupcall, s, 4) exited non-zero on > 'SIP/203-add5' in macro 'hangupcall' > == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-add5' > asterisk1*CLI> > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856