02-26-2007 07:32 PM - edited 03-14-2019 08:13 PM
myphone -> as5350xm -> pstn
if i enable all codecs in myphone, "show call active voice brief" shows g729r8. outbound call is ok.
But, if we enable only g711ulaw, call is dropped immediately (ring once or no ring happened). I do have the same problem with Asterisk if i enable only g711ulaw.
Since it is a LAN, i want to use low complexity codec only.
For dial-peer voip, i could specify codec preference or codec used. But for dial-peer pots, i do not know how to enable a specific codec.
pls advise.
**** running-config ***
dial-peer voice 100 pots
destination-pattern 49
no digit-strip
direct-inward-dial
port 2/1:D
forward-digits 8
02-26-2007 09:01 PM
Configure a voip dial-peer for incoming called-number. Can be the same for outbound. Configure codec or voice class codecs in there.
02-26-2007 10:46 PM
I have added a dial-peer voip as below:
voice service voip
h323
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
dial-peer voice 100 pots
destination-pattern 49
no digit-strip
direct-inward-dial
port 2/1:D
forward-digits 8
dial-peer voice 4 voip
voice-class codec 1
incoming called-number .
Call is still not established. Attached is the debug log.
Myphone calls dial-peer 100 by typing "49xxxxx@192.168.1.19".
pls kindly advise.
02-27-2007 07:42 AM
can u attach all the following debugs. From what you have attahced, does confirm that H245 is causing the call failure.
debug voip ccapi inout
h225 q931
h245 asn
cch323 session
02-28-2007 03:00 AM
03-01-2007 02:51 AM
pls advise what configuration could make the outbound call work. i have tried either or both of the following but without success:
h245 caps mode restricted
h245 tunnel disable
If we allow g723 or g729, there is no problem at all.
*** asterisk config ***
h323.conf
disallow=all
allow=ulaw
**** show voice dsp for myphone/gridborg/asterisk ***
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ========= ======= ===== ======= === == ========= == ==== ============
C5510 001 01 None 8.4.1 busy idle 0 0 23 0 0/0
If we use sip (using gridborg/asterisk), the codec is automatically g711ulaw and call is ok.
*** asterisk config ***
h323.conf
allow=all
attached pls find the debug log for asterisk.
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