Hi,
I'm currently building an asterisk system which should work as gateway
between SIP phones and ISDN. Most parts are working very fine, but one
problem occurs and I am not able to solve or debug it.
Telephony from ISDN to SIP (a Sipura Hardphone) is working very well,
but if the SIP Phone initiates the call, the ISDN phone rings, and a
connection can be established. But no one of the two peers can hear
anything.
I am using asterisk-1.0.7 with chan_capi-0.3.5, a Siemens Octopus
Telephony System and an AVM B1 ISDN card. When I switch on debugging on
the chan_capi, I see the packets flow? But there is still nothing to
hear.
Do you have any suggestions? Are there any deeper (undocumented) debug
possibilities?
Thanks a lot,
Torsten
--
bash$ :(){ :|:&};: ----- pgp: http://www.unixer.de/htor-key.asc -----
My software never has bugs... it just develops random features...
On Tue, 19 Jul 2005, Torsten Hoefler wrote:> Hi, > I'm currently building an asterisk system which should work as gateway > between SIP phones and ISDN. Most parts are working very fine, but one > problem occurs and I am not able to solve or debug it. > > Telephony from ISDN to SIP (a Sipura Hardphone) is working very well, > but if the SIP Phone initiates the call, the ISDN phone rings, and a > connection can be established. But no one of the two peers can hear > anything. > > I am using asterisk-1.0.7 with chan_capi-0.3.5, a Siemens Octopus > Telephony System and an AVM B1 ISDN card. When I switch on debugging on > the chan_capi, I see the packets flow? But there is still nothing to > hear. > > Do you have any suggestions? Are there any deeper (undocumented) debug > possibilities?This sounds like the FD bug in chan_capi-0.3.5, which happens on kernels > 2.6.10 (AFAIR) or e.g. BSD. Try out chan_capi-cm from sourceforge. Armin
Had the same problem and found that chan_capi-cm solved it. Watch out about the change to the dial command syntax which not does not let you specify outgoing MSN. It now seems impossible to specify outgoing MSN. The excellent perl Op_panel also seems unable to show events on a CAPI line button with this new syntax>>> asterisk-users@unixer.de 19/07/05 12:02:33 >>>Hi, I'm currently building an asterisk system which should work as gateway between SIP phones and ISDN. Most parts are working very fine, but one problem occurs and I am not able to solve or debug it. Telephony from ISDN to SIP (a Sipura Hardphone) is working very well, but if the SIP Phone initiates the call, the ISDN phone rings, and a connection can be established. But no one of the two peers can hear anything. I am using asterisk-1.0.7 with chan_capi-0.3.5, a Siemens Octopus Telephony System and an AVM B1 ISDN card. When I switch on debugging on the chan_capi, I see the packets flow? But there is still nothing to hear. Do you have any suggestions? Are there any deeper (undocumented) debug possibilities? Thanks a lot, Torsten -- bash$ :(){ :|:&};: ----- pgp: http://www.unixer.de/htor-key.asc ----- My software never has bugs... it just develops random features... _______________________________________________ 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 Scanned by MailDefender - managed email security from intY - www.maildefender.net