anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making progress passing it to SIP/102-767c Jun 28 10:51:21 NOTICE[278545]: channel.c:1654 ast_set_read_format: Unable to find a path from G723 to ALAW Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from ULAW to G723 -- CAPI[contr1/2003002]/0 is ringing Jun 28 10:51:21 WARNING[278545]: chan_sip.c:1788 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 8/4) Jun 28 10:51:21 WARNING[278545]: channel.c:1485 ast_prod: Prodding channel 'SIP/102-767c' failed Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from SLINR to G723 Jun 28 10:51:21 WARNING[278545]: indications.c:76 playtones_alloc: Unable to set 'SIP/102-767c' to signed linear format (write) -- CAPI Hangingup == Spawn extension (from-sip, 9, 1) exited non-zero on 'SIP/102-767c'
Klaus-Peter Junghanns
2004-Jun-29 01:49 UTC
[Asterisk-Users] sip to isdn-capi call problem
Hi Tomaz, make sure you disable the G723.1 codec in your SIP device, asterisk does not support G723.1. Use G711 (alaw, ulaw)! best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-28 um 10.52 schrieb Tomaz:> anyone has idea what problem can be here, > > something with codec but i have today CVS version and grandstream phone > with 1.5.0 firmware.I try to change codec in phone and also in > asterisk-sip.conf but the same. > What can be problem ? > > tnx, > Tomaz > > > > > *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack > -- Called 2:5 > -- CAPI[contr1/2003002]/0 is making progress passing it to SIP/102-767c > Jun 28 10:51:21 NOTICE[278545]: channel.c:1654 ast_set_read_format: > Unable to find a path from G723 to ALAW > Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: > Unable to find a path from ULAW to G723 > -- CAPI[contr1/2003002]/0 is ringing > Jun 28 10:51:21 WARNING[278545]: chan_sip.c:1788 sip_write: Asked to > transmit frame type 4, while native formats is 1 (read/write = 8/4) > Jun 28 10:51:21 WARNING[278545]: channel.c:1485 ast_prod: Prodding > channel 'SIP/102-767c' failed > Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: > Unable to find a path from SLINR to G723 > Jun 28 10:51:21 WARNING[278545]: indications.c:76 playtones_alloc: > Unable to set 'SIP/102-767c' to signed linear format (write) > -- CAPI Hangingup > == Spawn extension (from-sip, 9, 1) exited non-zero on 'SIP/102-767c' > http://lists.digium.com/mailman/listinfo/asterisk-users
Klaus-Peter Junghanns wrote:>Hi Tomaz, > >make sure you disable the G723.1 codec in your SIP device, asterisk >does not support G723.1. Use G711 (alaw, ulaw)! > >best regards > >Klaus > >yes ,this was a problem . thank you. tomaz