[Asterisk-Users] Problem SIP no audio just noise

Thu Jul 8 07:00:35 MST 2004

I'm trying to call from XLite phone to PSTN
(I've tried this from internet and from local network the same)
The Xlite doesn't write that it is connected but receives excelent audio.
At the other end comes only noise. Some times only for a second you can 
here the
caller  voice , but this was only one time :)

I saw with ethereal that UDP packets are coming and going to the 
asterisk box.

Sorry for the long logs.

Sip read:
INVITE sip:99826816 at 10.1.1.2 SIP/2.0
Via: SIP/2.0/UDP 
10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2
From: damian <sip:damian at 10.1.1.2>;tag=2667644054
To: <sip:99826816 at 10.1.1.2>
Contact: <sip:damian at 10.1.1.11:5060>
Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11
CSeq: 42510 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 291

v=0
o=damian 23894728 23894788 IN IP4 10.1.1.11
s=X-Lite
c=IN IP4 10.1.1.11
t=0 0
m=audio 8000 RTP/AVP 0 8 3 97 110 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 13 lines
Using latest request as basis request
Sending to 10.1.1.11 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Peer RTP is at port 10.1.1.11:0
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - 
audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 
0xe(GSM|ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Found peer 'phone1010'
Jul  8 16:47:21 DEBUG[65541]: chan_sip.c:4851 check_user: Setting NAT on 
RTP to 0
Jul  8 16:47:21 DEBUG[65541]: chan_sip.c:6424 handle_request: Check for 
res for damian
Jul  8 16:47:21 DEBUG[65541]: chan_sip.c:1386 update_user_counter: 
damian is not a local user
Looking for 99826816 in default
Jul  8 16:47:21 DEBUG[65541]: chan_sip.c:4115 build_route: build_route: 
Contact hop: <sip:damian at 10.1.1.11:5060>
list_route: hop: <sip:damian at 10.1.1.11:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2
From: damian <sip:damian at 10.1.1.2>;tag=2667644054
To: <sip:99826816 at 10.1.1.2>;tag=as5b6158bb
Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11
CSeq: 42510 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:99826816 at 10.1.1.2:0>
Content-Length: 0

  to 10.1.1.11:5060
     -- Executing Dial("SIP/damian-ff45", "Zap/4/9826816") in new stack
Jul  8 16:47:21 DEBUG[262159]: chan_zap.c:1576 zt_call: Dialing '9826816'
Jul  8 16:47:21 DEBUG[262159]: chan_zap.c:1633 zt_call: Deferring dialing...
     -- Called 4/9826816
Jul  8 16:47:21 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception 
on 21, channel 4
Jul  8 16:47:21 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got 
event Hook Transition Complete(12) on channel 4 (index 0)
Jul  8 16:47:23 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception 
on 21, channel 4
Jul  8 16:47:23 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got 
event Dial Complete(9) on channel 4 (index 0)
Jul  8 16:47:23 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No 
echocancellation requested
Jul  8 16:47:23 DEBUG[262159]: chan_zap.c:1185 zt_train_ec: No echo 
training requested
Jul  8 16:47:24 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception 
on 21, channel 4
Jul  8 16:47:24 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got 
event Dial Complete(9) on channel 4 (index 0)
Jul  8 16:47:24 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No 
echocancellation requested
Jul  8 16:47:24 DEBUG[262159]: chan_zap.c:3007 zt_handle_event: Done 
dialing, but waiting for progress detection before doing more...
We're at 10.1.1.2 port 10524
Answering with capability 0x2(GSM)
Answering with capability 0x4(ULAW)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2
From: damian <sip:damian at 10.1.1.2>;tag=2667644054
To: <sip:99826816 at 10.1.1.2>;tag=as5b6158bb
Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11
CSeq: 42510 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:99826816 at 10.1.1.2:0>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 586 586 IN IP4 10.1.1.2
s=session
c=IN IP4 10.1.1.2
t=0 0
m=audio 10524 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

  to 10.1.1.11:5060
Jul  8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format 
changed from UNKN to ULAW
Jul  8 16:47:24 DEBUG[262159]: chan_sip.c:1976 sip_rtp_read: Oooh, 
format changed to 2
Jul  8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format 
changed from ULAW to GSM
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.1.1.11 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.2:0;branch=z9hG4bK6eac0d88
From: "asterisk" <sip:asterisk at 10.1.1.2:0>;tag=as5fdf9f82
To: <sip:10.1.1.11>
Contact: <sip:asterisk at 10.1.1.2:0>
Call-ID: 2b4d7a39423d1c805053483b6fb5367a at 10.1.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 08 Jul 2004 13:47:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

  (no NAT) to 10.1.1.11:5060
voipgw*CLI>

Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.1.1.2:0;branch=z9hG4bK6eac0d88
From: "asterisk" <sip:asterisk at 10.1.1.2:0>;tag=as5fdf9f82
To: <sip:10.1.1.11>;tag=2355563749
Contact: <sip:damian at 10.1.1.11:5060>
Call-ID: 2b4d7a39423d1c805053483b6fb5367a at 10.1.1.2
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY
CSeq: 102 OPTIONS
Server: X-Lite release 1103a
Content-Length: 0

10 headers, 0 lines
Jul  8 16:47:31 DEBUG[65541]: chan_sip.c:752 __sip_ack: Stopping 
retransmission on '2b4d7a39423d1c805053483b6fb5367a at 10.1.1.2' of Request 
102: Found
Destroying call '2b4d7a39423d1c805053483b6fb5367a at 10.1.1.2'
voipgw*CLI>

Sip read:
CANCEL sip:99826816 at 10.1.1.2 SIP/2.0
Via: SIP/2.0/UDP 
10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2
From: damian <sip:damian at 10.1.1.2>;tag=2667644054
To: <sip:99826816 at 10.1.1.2>
Contact: <sip:damian at 10.1.1.11:5060>
Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11
CSeq: 42510 CANCEL
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0

10 headers, 0 lines
Sending to 10.1.1.11 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2
From: damian <sip:damian at 10.1.1.2>;tag=2667644054
To: <sip:99826816 at 10.1.1.2>;tag=as5b6158bb
Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11
CSeq: 42510 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:99826816 at 10.1.1.2:0>
Content-Length: 0

  to 10.1.1.11:5060
Jul  8 16:47:36 DEBUG[262159]: chan_zap.c:1876 zt_hangup: Hangup: 
channel: 4 index = 0, normal = 21, callwait = -1, thirdcall = -1
Jul  8 16:47:36 DEBUG[262159]: chan_zap.c:2272 zt_setoption: Set option 
TDD MODE, value: OFF(0) on Zap/4-1
Jul  8 16:47:36 DEBUG[262159]: chan_zap.c:1141 update_conf: Updated 
conferencing on 4, with 0 conference users
     -- Hungup 'Zap/4-1'
   == Spawn extension (default, 99826816, 1) exited non-zero on 
'SIP/damian-ff45'
Jul  8 16:47:36 DEBUG[262159]: cdr_addon_mysql.c:181 mysql_log: 
cdr_mysql: inserting a CDR record.
Jul  8 16:47:36 DEBUG[262159]: cdr_addon_mysql.c:200 mysql_log: 
cdr_mysql: SQL command as follows:  INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) 
VALUES ('2004-07-08 16:47:36','\"damian\" 
<damian>','damian','99826816','default', 
'SIP/damian-ff45','Zap/4-1','Dial','Zap/4/9826816',15,0,'NO ANSWER',3,'')
Jul  8 16:47:36 DEBUG[262159]: chan_sip.c:1508 sip_hangup: 
update_user_counter(damian) - decrement inUse counter
Jul  8 16:47:36 DEBUG[262159]: chan_sip.c:1386 update_user_counter: 
damian is not a local user
Destroying call '912EEDDD-2BDC-4CF8-A627-DECC35793EA5 at 10.1.1.11'
voipgw*CLI>