[dualarrow]

Hi,

I have asterisk running on an entirly local network. A sip client on this network tries to call Asterisk. Using the echo test, all works fine.

Using an IAX client also on the local net, works fine for the echo test. Both are using the GSM codec.

When trying to dial a PSTN number via a TDM400, I get 1 way audio (no sound from the SIP client to the PSTN) after the connection is made, but again, IAX works.

This feels like some type of bridging problem betweem SIP and the TDM400, but it was working about a week ago and I don't think I have changed anything that should have affected it.

Does anyone have any suggestions on what might be causing this ?

Thanks,
Andrew

[dswartz]

that is strange indeed. does anything show up in the * logs?

[dualarrow]

This is the relevent lines from /var/log/asterisk/full

Jan 13 08:09:03 DEBUG[3420] chan_zap.c: Dialing '92125532'
Jan 13 08:09:03 DEBUG[3420] chan_zap.c: Deferring dialing...
Jan 13 08:09:03 VERBOSE[3420] logger.c: -- Called 35/92125532
Jan 13 08:09:04 DEBUG[3396] chan_sip.c: Stopping retransmission on '547e281f1a7319fa682f99c5796ce137@10.250.1.10' of Request 102: Match Found
Jan 13 08:09:04 DEBUG[3420] chan_zap.c: Exception on 27, channel 35
Jan 13 08:09:04 DEBUG[3420] chan_zap.c: Got event Hook Transition Complete(12) on channel 35 (index 0)
Jan 13 08:09:05 DEBUG[3420] chan_zap.c: Exception on 27, channel 35
Jan 13 08:09:05 DEBUG[3420] chan_zap.c: Got event Dial Complete(9) on channel 35 (index 0)
Jan 13 08:09:05 DEBUG[3420] chan_zap.c: Enabled echo cancellation on channel 35
Jan 13 08:09:05 DEBUG[3420] chan_zap.c: Engaged echo training on channel 35
Jan 13 08:09:07 DEBUG[3420] chan_zap.c: Exception on 27, channel 35
Jan 13 08:09:07 DEBUG[3420] chan_zap.c: Got event Dial Complete(9) on channel 35 (index 0)
Jan 13 08:09:07 DEBUG[3420] chan_zap.c: Echo cancellation already on
Jan 13 08:09:07 DEBUG[3420] chan_zap.c: Done dialing, but waiting for progress detection before doing more...
Jan 13 08:09:13 NOTICE[3401] chan_iax2.c: Peer 'FWD' is now REACHABLE! Time: 303
Jan 13 08:09:13 DEBUG[3401] chan_iax2.c: Peer lastms 303, historicms 303, maxms 2000
Jan 13 08:09:34 DEBUG[3420] chan_zap.c: Hangup: channel: 35 index = 0, normal = 27, callwait = -1, thirdcall = -1
Jan 13 08:09:34 DEBUG[3420] chan_zap.c: disabled echo cancellation on channel 35
Jan 13 08:09:34 DEBUG[3420] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/35-1
Jan 13 08:09:34 DEBUG[3420] chan_zap.c: Updated conferencing on 35, with 0 conference users
Jan 13 08:09:34 VERBOSE[3420] logger.c: -- Hungup 'Zap/35-1'
Jan 13 08:09:34 DEBUG[3420] app_dial.c: Exiting with DIALSTATUS=CANCEL.
Jan 13 08:09:34 VERBOSE[3420] logger.c: == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/532-dcff' in macro 'dialout-trunk'
Jan 13 08:09:34 VERBOSE[3420] logger.c: == Spawn extension (from-internal, 092125532, 1) exited non-zero on 'SIP/532-dcff'
Jan 13 08:09:34 VERBOSE[3420] logger.c: -- Executing Macro("SIP/532-dcff", "hangupcall") in new stack
Jan 13 08:09:34 VERBOSE[3420] logger.c: -- Executing ResetCDR("SIP/532-dcff", "w") in new stack
Jan 13 08:09:34 DEBUG[3396] chan_sip.c: Stopping retransmission on '8F21DC0B-E70F-4863-8A2F-165268CED3BD@10.250.2.23' of Response 823: Match Found
Jan 13 08:09:34 DEBUG[3420] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jan 13 08:09:34 DEBUG[3420] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel ,lastapp,l
astdata,duration,billsec,disposition,amaflags,acco untcode,uniqueid) VALUES ('2006-01-13 08:09:01','892125532','892125532','092125532','fro m-internal', 'SIP/53
2-dcff','Zap/35-1','ResetCDR','w',33,0,'NO ANSWER',3,'','1137110941.0')
Jan 13 08:09:34 VERBOSE[3420] logger.c: -- Executing NoCDR("SIP/532-dcff", "") in new stack
Jan 13 08:09:34 WARNING[3420] cdr.c: CDR on channel 'SIP/532-dcff' not posted
Jan 13 08:09:34 WARNING[3420] cdr.c: CDR on channel 'SIP/532-dcff' lacks end
Jan 13 08:09:34 VERBOSE[3420] logger.c: -- Executing Wait("SIP/532-dcff", "5") in new stack
Jan 13 08:09:34 VERBOSE[3420] logger.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/532-dcff' in macro 'hangupcall'
Jan 13 08:09:34 VERBOSE[3420] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/532-dcff'
Jan 13 08:09:34 DEBUG[3420] chan_sip.c: update_call_counter(532) - decrement call limit counter
Jan 13 08:09:38 DEBUG[3075] manager.c: Manager received command 'Command'
Jan 13 08:09:38 DEBUG[3075] manager.c: Manager received command 'Command'
Jan 13 08:09:39 DEBUG[3396] chan_sip.c: Stopping retransmission on '53356ddf62488a6d79b8ace149f3fde0@10.250.1.10' of Request 102: Match Found
Jan 13 08:09:43 DEBUG[3396] chan_sip.c: Auto destroying call 'AA9346739B714EDC99C523A107F5E8E5@10.250.1.10'
Jan 13 08:09:47 DEBUG[3396] chan_sip.c: Stopping retransmission on '3598cde870543ca0452fc6d26ab09a5c@10.250.1.10' of Request 102: Match Found
Jan 13 08:09:47 DEBUG[3396] chan_sip.c: Stopping retransmission on '1966c1c919b0e8f451b0009c5e059d49@10.250.1.10' of Request 102: Match Found
Jan 13 08:09:47 DEBUG[3396] chan_sip.c: Stopping retransmission on '6817d9d56aba987e138a50111658e53a@10.250.1.10' of Request 102: Match Found

I can't see anything wrong myself, but then again..........

Thanks,
Andrew

[dswartz]

nothing obvious (to me at any rate.) what shows up under sip peers when the call is active?

[dualarrow]

sip show peers relevent line is

532/532 10.250.2.23 D 5060 OK (1 ms)

sip show channels

Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
10.250.2.23 532 82694515-CD 00101/08997 ulaw No Rx: INVITE
1 active SIP channel


sip show settings

Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS: 0x0
OSP Support: No
SIP realtime: Disabled
asterisk1*CLI>
Global Signalling Settings:
---------------------------
Codecs: ulaw,alaw
Relax DTMF: No
Compact SIP headers: No
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
asterisk1*CLI>
Default Settings:
-----------------
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
Musicclass: default
Voice Mail Extension: asterisk


Thanks

[dswartz]

well, sad to say i don't see anything wrong here, but i'm far from an asterisk expert...

[dualarrow]

Thanks anyway dswartz.
Andrew

[dualarrow]

Some more info.

When listening on the PSTN, it sounds like the audio is muted. Asterisk doesnt seem to think the line has been answered.

If I start making loud sounds in the PSTN phone, asterisk seems to figure out the line has been answered and the audio comes to life and starts working perfectly, but only for that call. Each call needs this sound to kick it into action.

I seem to recall another thread with similar problems, so I'll hunt for it.

Asterisk CLI reports

-- Executing Dial("SIP/532-30d8", "ZAP/35/92125532") in new stack
-- Called 35/92125532
-- Zap/35-1 answered SIP/532-30d8

It's the last line that starts the audio functioning, but only seems to start if I make a lot of noise on the PSTN line.

Andrew