Gilberto Verástegui
2011-Dec-28 21:32 UTC
[asterisk-users] MFCR2 Long distance calls not connected
Calls to long distance get disconnected before answer.
Telco: Alestra
Country: Mexico
System: Elastix 2.2
Digital Card: Digium TE122
Log:
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing
[+525552622900 at default:1] Set("SIP/OCS_TRUNK-000001bf",
"EXT=015552622900") in new stack
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing
[+525552622900 at default:2] Dial("SIP/OCS_TRUNK-000001bf",
"DAHDI/g1/015552622900,60") in new stack
[Dec 28 14:37:44] VERBOSE[4586] app_dial.c: -- Called
DAHDI/g1/015552622900
[Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan 1
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 call
on chan 1
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted in
openr2 cause 6/Normal Clearing
[Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: -- Hungup
'DAHDI/1-1'
[Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn extension
(default, +525552622900, 2) exited non-zero on 'SIP/OCS_TRUNK-000001bf'
[Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on
channel 1
Found this email list, but I think is too old.
http://www.mail-archive.com/asterisk-users at lists.digium.com/msg205765.html
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Moises Silva
2012-Jan-03 04:25 UTC
[asterisk-users] MFCR2 Long distance calls not connected
On Wed, Dec 28, 2011 at 3:32 PM, Gilberto Ver?stegui <gilbertovm at ti-m.com.mx> wrote:> Calls to long distance get disconnected before answer. > Telco: Alestra > Country: Mexico > System: Elastix 2.2 > Digital Card: Digium TE122 > > Log: > > [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing > [+525552622900 at default:1] Set("SIP/OCS_TRUNK-000001bf", > "EXT=015552622900") in new stack > [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing > [+525552622900 at default:2] Dial("SIP/OCS_TRUNK-000001bf", > "DAHDI/g1/015552622900,60") in new stack > [Dec 28 14:37:44] VERBOSE[4586] app_dial.c: -- Called > DAHDI/g1/015552622900 > [Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan > 1 > [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 > call on chan 1 > [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted > in openr2 cause 6/Normal Clearing > [Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: -- Hungup 'DAHDI/1-1' > [Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn > extension (default, +525552622900, 2) exited non-zero on > 'SIP/OCS_TRUNK-000001bf' > [Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on > channel 1 > > Found this email list, but I think is too old. > > http://www.mail-archive.com/asterisk-users at lists.digium.com/msg205765.html > >You would be better off asking this questions in asterisk-r2 mailing list. I will answer the same way that I answered back then. You need to enable protocol debugging. Without protocol debugging there is no way to tell what is happening to the call. Read the sample chan_dahdi.conf included with Asterisk and search for mfcr2 logging options. Having said that, it is possible in international calls you need to specify a different caller category. *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120102/25fdb11f/attachment.htm>