I have Asterisk 1.4 Trunk 180682 running on a AMD64 8-core server and
trying to use the following modules (same effect on both):
codec_g729-ast14-gcc4-glibc-x86_64-opteron-sse3.so
or
codec_g729-ast14-gcc4-glibc-x86_64-pentium4.so
The two phones are Granstream GXP2000 running G729 on both of them.
Using an older version of the trunk, older codec_g729-ast14-gcc4-glibc-
core2.so module on a core2 system, this setup works fine.
I use the exact same asterisk configs on the new test server and the
first call I make works perfectly. Clean sound, fast, etc.
Any additional calls, I get no audio and after a minute:
[Mar 9 12:55:37] NOTICE[19126]: chan_sip.c:16436 do_monitor:
Disconnecting call 'SIP/5001-008aa9f0' for lack of RTP activity in 61
seconds
[Mar 9 12:55:37] NOTICE[19126]: chan_sip.c:16436 do_monitor:
Disconnecting call 'SIP/5000-9803c740' for lack of RTP activity in 61
seconds
Like I said, the first call works.
I am trying to figure out how to get more info about the call. Here
is what I have:
192.168.1.6 5001 4b12378336a 00102/00000 0x100
(g729) No Tx: ACK
192.168.168.75 5000 eb7b1d7a27c 00101/51176 0x100
(g729) No Rx: ACK
* SIP Call
Curr. trans. direction: Outgoing
Call-ID:
4b12378336a6a983...@domain.com
Owner channel ID: SIP/5001-008aa9f0
Our Codec Capability: 1294
Non-Codec Capability (DTMF): 1
Their Codec Capability: 256
Joint Codec Capability: 256
Format: 0x100 (g729)
MaxCallBR: 384 kbps
Theoretical Address:
192.168.1.6:5066
Received Address: X.X.X.X:5066
SIP Transfer mode: open
NAT Support: Always
Audio IP: 173.45.232.143 (local)
Our Tag: as55c6a3ff
Their Tag: 1437d1d32bffffa1
SIP User agent: Grandstream GXP2000 1.1.1.14
Username: 5001
Peername: 5001
Original uri:
sip:50...@192.168.1.6:5066
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
Route:
sip:50...@192.168.1.6:5066
DTMF Mode: rfc2833
SIP Options: (none)
* SIP Call
Curr. trans. direction: Incoming
Call-ID:
eb7b1d7a...@domain.com
Owner channel ID: SIP/5000-9804c910
Our Codec Capability: 1294
Non-Codec Capability (DTMF): 1
Their Codec Capability: 287
Joint Codec Capability: 270
Format: 0x100 (g729)
MaxCallBR: 384 kbps
Theoretical Address:
192.168.168.75:5060
Received Address: A.B.C.D:5060
SIP Transfer mode: open
NAT Support: Always
Audio IP: 173.45.232.143 (local)
Our Tag: as66a00724
Their Tag: b9be3b2342cb88aa
SIP User agent: Grandstream GXP2000 1.1.1.14
Username: 5000
Peername: 5000
Original uri:
sip:50...@192.168.168.75:5060
Caller-ID: 5000
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
Route:
sip:50...@192.168.168.75:5060
DTMF Mode: rfc2833
SIP Options: replaces replace timer
With the module unloaded, I cannot hear ulaw prompts on the server.
With the module loaded, I hear the prompts from the server. This
happens even after the first call. So the trans coding does work.
However, it is the pass-through that does not work.
If I unload the module, the NO call works.
If I loaded it again, the next call will work, but the one after that
will not work.
Not sure about bringing this to the dev group until I have something
else they can blame other than this module.
It does seem to be Asterisk and not the module because the module does
work once and then not again.
Any pointers.