Klaus Peras
2005-Dec-13 05:34 UTC
[Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
Hi Asterisk Users,
i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian
3.1. With a quadbri card installad, wich is running on the bristuff drivers.
Everything seems to be fine so far.
but now i wanted to use the g.729A Codec. I bought 5 licences and
installed them:
asterisk3*CLI> show g729
0/0 encoders/decoders of 5 licensed channels are currently in use
When i do sip to sip calls, everything is working fine (from a snom 190
wich is running with that codec to a sip phone with g.711a), asterisk is
translating correct.
the output on the CLI is:
asterisk3*CLI> show g729
1/0 encoders/decoders of 5 licensed channels are currently in use
But if i try to call a zap channel from that sip phone (snom 190) wich
runs that g729 Codec, i don?t hear anything on the ISDN Phone. the
output on the CLI:
asterisk3*CLI> show g729
1/1 encoders/decoders of 5 licensed channels are currently in use
Here is the output of the show channel command for the SIP Channel and
the ZAP Channel:
asterisk3*CLI> show channel SIP/71-d293
-- General --
Name: SIP/71-d293
Type: SIP
UniqueID: asterisk-2204-1134137006.49
Caller ID: 30071
DNID Digits: 329
State: Up (6)
Rings: 0
NativeFormat: 256
WriteFormat: 256
ReadFormat: 64
1st File Descriptor: 31
Frames in: 7949
Frames out: 7956
Time to Hangup: 0
Elapsed Time: 0h2m39s
-- PBX --
Context: default
Extension: 329
Priority: 2
Call Group: 0
Pickup Group: 0
Application: Dial
Data: Zap/g1/329
Stack: 0
Blocking in: ast_waitfor_nandfds
asterisk3*CLI> show channel Zap/1-1
-- General --
Name: Zap/1-1
Type: Zap
UniqueID: asterisk-2204-1134137006.50
Caller ID: 30071
DNID Digits: 329
State: Up (6)
Rings: 0
NativeFormat: 72
WriteFormat: 64
ReadFormat: 256
1st File Descriptor: 13
Frames in: 8255
Frames out: 8246
Time to Hangup: 0
Elapsed Time: 0h0m0s
-- PBX --
Context: default
Extension: s
Priority: 1
Call Group: 0
Pickup Group: 0
Application: Bridged Call
Data: SIP/71-d293
Stack: -1
Blocking in: ast_waitfor_nandfds
I don?t know what i can do on this problem and would be pleased to get
some help.
Thank you very much!
--
Mit freundlichen Gr??en
With kind regards
Klaus Peras
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Klaus Peras
2005-Dec-13 07:32 UTC
Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
Hi, i just figured out, that there is also a problem by going in a conference with the sip phone that runs the g729a codec. Could it be, that i have timing problems? I don?t have digium hardware installed, but i have ztdummy: asterisk3:/etc/asterisk# lsmod | grep ztdummy ztdummy 3748 0 zaptel 225540 24 ztdummy,qozap Does anybody have a advice for me? Mit freundlichen Gr??en With kind regards Klaus Peras Klaus Peras schrieb:> Hi Asterisk Users, > > i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a > Debian 3.1. With a quadbri card installad, wich is running on the > bristuff drivers. > Everything seems to be fine so far. > but now i wanted to use the g.729A Codec. I bought 5 licences and > installed them: > asterisk3*CLI> show g729 > 0/0 encoders/decoders of 5 licensed channels are currently in use > > When i do sip to sip calls, everything is working fine (from a snom > 190 wich is running with that codec to a sip phone with g.711a), > asterisk is translating correct. > the output on the CLI is: > asterisk3*CLI> show g729 > 1/0 encoders/decoders of 5 licensed channels are currently in use > > But if i try to call a zap channel from that sip phone (snom 190) wich > runs that g729 Codec, i don?t hear anything on the ISDN Phone. the > output on the CLI: > asterisk3*CLI> show g729 > 1/1 encoders/decoders of 5 licensed channels are currently in use > > Here is the output of the show channel command for the SIP Channel and > the ZAP Channel: > > asterisk3*CLI> show channel SIP/71-d293 > -- General -- > Name: SIP/71-d293 > Type: SIP > UniqueID: asterisk-2204-1134137006.49 > Caller ID: 30071 > DNID Digits: 329 > State: Up (6) > Rings: 0 > NativeFormat: 256 > WriteFormat: 256 > ReadFormat: 64 > 1st File Descriptor: 31 > Frames in: 7949 > Frames out: 7956 > Time to Hangup: 0 > Elapsed Time: 0h2m39s > -- PBX -- > Context: default > Extension: 329 > Priority: 2 > Call Group: 0 > Pickup Group: 0 > Application: Dial > Data: Zap/g1/329 > Stack: 0 > Blocking in: ast_waitfor_nandfds > asterisk3*CLI> show channel Zap/1-1 > -- General -- > Name: Zap/1-1 > Type: Zap > UniqueID: asterisk-2204-1134137006.50 > Caller ID: 30071 > DNID Digits: 329 > State: Up (6) > Rings: 0 > NativeFormat: 72 > WriteFormat: 64 > ReadFormat: 256 > 1st File Descriptor: 13 > Frames in: 8255 > Frames out: 8246 > Time to Hangup: 0 > Elapsed Time: 0h0m0s > -- PBX -- > Context: default > Extension: s > Priority: 1 > Call Group: 0 > Pickup Group: 0 > Application: Bridged Call > Data: SIP/71-d293 > Stack: -1 > Blocking in: ast_waitfor_nandfds > > I don?t know what i can do on this problem and would be pleased to get > some help. > > Thank you very much! > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-------------- next part -------------- A non-text attachment was scrubbed... Name: klaus.peras.vcf Type: text/x-vcard Size: 264 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051213/aef18295/klaus.peras.vcf
Klaus Peras
2005-Dec-14 10:49 UTC
Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
I thougt i have some problems with ztdummy and removed that # in front of ztdummy in the zaptel Makefile before compiling. But still no change. I even tried it with another Phone, a Planet VIP-150T. Still the same Problem, i don?t hear anything from the SIP Phone on the ISDN Phone, but i hear everything fine the other way. Any Ideas? Thanks a lot for help. regards Klaus Peras Klaus Peras schrieb:> Hi, i just figured out, that there is also a problem by going in a > conference with the sip phone that runs the g729a codec. > Could it be, that i have timing problems? I don?t have digium hardware > installed, but i have ztdummy: > > asterisk3:/etc/asterisk# lsmod | grep ztdummy > ztdummy 3748 0 > zaptel 225540 24 ztdummy,qozap > > Does anybody have a advice for me? > > Mit freundlichen Gr??en > With kind regards > > Klaus Peras > > > > > > > Klaus Peras schrieb: > >> Hi Asterisk Users, >> >> i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a >> Debian 3.1. With a quadbri card installad, wich is running on the >> bristuff drivers. >> Everything seems to be fine so far. >> but now i wanted to use the g.729A Codec. I bought 5 licences and >> installed them: >> asterisk3*CLI> show g729 >> 0/0 encoders/decoders of 5 licensed channels are currently in use >> >> When i do sip to sip calls, everything is working fine (from a snom >> 190 wich is running with that codec to a sip phone with g.711a), >> asterisk is translating correct. >> the output on the CLI is: >> asterisk3*CLI> show g729 >> 1/0 encoders/decoders of 5 licensed channels are currently in use >> >> But if i try to call a zap channel from that sip phone (snom 190) >> wich runs that g729 Codec, i don?t hear anything on the ISDN Phone. >> the output on the CLI: >> asterisk3*CLI> show g729 >> 1/1 encoders/decoders of 5 licensed channels are currently in use >> >> Here is the output of the show channel command for the SIP Channel >> and the ZAP Channel: >> >> asterisk3*CLI> show channel SIP/71-d293 >> -- General -- >> Name: SIP/71-d293 >> Type: SIP >> UniqueID: asterisk-2204-1134137006.49 >> Caller ID: 30071 >> DNID Digits: 329 >> State: Up (6) >> Rings: 0 >> NativeFormat: 256 >> WriteFormat: 256 >> ReadFormat: 64 >> 1st File Descriptor: 31 >> Frames in: 7949 >> Frames out: 7956 >> Time to Hangup: 0 >> Elapsed Time: 0h2m39s >> -- PBX -- >> Context: default >> Extension: 329 >> Priority: 2 >> Call Group: 0 >> Pickup Group: 0 >> Application: Dial >> Data: Zap/g1/329 >> Stack: 0 >> Blocking in: ast_waitfor_nandfds >> asterisk3*CLI> show channel Zap/1-1 >> -- General -- >> Name: Zap/1-1 >> Type: Zap >> UniqueID: asterisk-2204-1134137006.50 >> Caller ID: 30071 >> DNID Digits: 329 >> State: Up (6) >> Rings: 0 >> NativeFormat: 72 >> WriteFormat: 64 >> ReadFormat: 256 >> 1st File Descriptor: 13 >> Frames in: 8255 >> Frames out: 8246 >> Time to Hangup: 0 >> Elapsed Time: 0h0m0s >> -- PBX -- >> Context: default >> Extension: s >> Priority: 1 >> Call Group: 0 >> Pickup Group: 0 >> Application: Bridged Call >> Data: SIP/71-d293 >> Stack: -1 >> Blocking in: ast_waitfor_nandfds >> >> I don?t know what i can do on this problem and would be pleased to >> get some help. >> >> Thank you very much! >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-------------- next part -------------- A non-text attachment was scrubbed... Name: klaus.peras.vcf Type: text/x-vcard Size: 264 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051214/74cf5f49/klaus.peras.vcf