[Asterisk-Users] DTMF broken

Sun May 9 12:56:16 MST 2004

On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote:
> Mark,
> Could you please add a SIP debug message with the SIP INFO?

I've done a debug with a working asterisk (V1.0) and the non-working
asterisk. The trace is attached.  :-)    (debug - ascii text)

When you say "SIP INFO" - what else are you asking for???
If its one of the 'sip show' commands - which one, and at what instance
of time?

-- 
  .  .     ___. .__      Posix Systems - Sth Africa
 /| /|       / /__       mje at posix.co.za  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

-------------- next part --------------
This is a debug trace of Asterisk v1-0_stable - I'm dialing '310' which in extensions.conf looks like..
; 310 = Access Voicemail - with full prompting
exten => 310,1,VoicemailMain()

I'm hanging up after 'dialing' 203
... the 'bad' one follows after....

*CLI> sip debug
SIP Debugging Enabled
*CLI> 

Sip read: 
INVITE sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2408 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

12 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as3564c06e
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2408 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Proxy-Authenticate: Digest realm="asterisk", nonce="6d4d7372"
Content-Length: 0

 to 160.124.48.121:5060

Sip read: 
ACK sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as3564c06e
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2408 ACK
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

11 headers, 0 lines

Sip read: 
INVITE sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at asterisk.posix.co.z
a;user=phone", nonce="6d4d7372", response="0142fb85eda2d7497992a0149d78e828"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2409 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

13 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 310 in sip
list_route: hop: <sip:phone1 at 160.124.48.121;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2409 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
    -- Executing VoiceMailMain("SIP/phone1-a1b1", "") in new stack
We're at 160.124.48.24 port 15044
Answering with capability 2
Answering with capability 4
Answering with capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2409 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 32123 32123 IN IP4 160.124.48.24
s=session
c=IN IP4 160.124.48.24
t=0 0
m=audio 15044 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

 to 160.124.48.121:5060
    -- Playing 'vm-login' (language 'en')

Sip read: 
ACK sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", non
ce="6d4d7372", response="9d79430d735b4601caae79a73aabed83"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2409 ACK
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 149
v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:40

20 headers, 0 lines

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK1efc17b00000497e
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", non
ce="6d4d7372", response="9d79430d735b4601caae79a73aabed83"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2410 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=2
Duration=960
13 headers, 2 lines
Receiving DTMF!
DTMF received: '2'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK1efc17b00000497e
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2410 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK0000000000000000
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", non
ce="6d4d7372", response="9d79430d735b4601caae79a73aabed83"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2411 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 23

Signal=0
Duration=1280
13 headers, 2 lines
Receiving DTMF!
DTMF received: '0'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK0000000000000000
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2411 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK716bcd0a16f54e24
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", non
ce="6d4d7372", response="9d79430d735b4601caae79a73aabed83"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2412 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=3
Duration=960
13 headers, 2 lines
Receiving DTMF!
DTMF received: '3'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK716bcd0a16f54e24
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2412 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
    -- Playing 'vm-password' (language 'en')

Sip read: 
BYE sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK9e28494e17b0e9c5
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", non
ce="6d4d7372", response="642c2d9b096aca4bbcc7f49495dece8b"
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2413 BYE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

12 headers, 0 lines
Sending to 160.124.48.121 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK9e28494e17b0e9c5
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=63f98f4e24e20f2f
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as2363ae73
Call-ID: 73484ad634d872c6 at 160.124.48.121
CSeq: 2413 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
May  9 21:28:02 WARNING[376854]: app_voicemail.c:2764 vm_execmain: Unable to read password
  == Spawn extension (sip, 310, 1) exited non-zero on 'SIP/phone1-a1b1'
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 160.124.48.24:5060;branch=z9hG4bK6875710d
From: <sip:292951 at fwd.pulver.com>;tag=as58f1c922
To: <sip:292951 at fwd.pulver.com>
Call-ID: 6b8b4567327b23c6643c986966334873 at 160.124.48.24
CSeq: 190 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1000 at 160.124.48.24>
Event: registration
Content-Length: 0

 (no NAT) to 192.246.69.223:5060

--------------------------------End of 'good' trace ------------------------
Bad Asterisk....

*CLI> show version
Asterisk CVS-04/05/04-09:58:21 built by root at asterisk on a i686 running Linux

*CLI> sip debug
SIP Debugging Enabled
*CLI> 

Sip read: 
INVITE sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe6303ef16f54c20e
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29106 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

12 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 15
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe6303ef16f54c20e
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as174e1ffb
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Proxy-Authenticate: Digest realm="asterisk", nonce="150a7b85"
Content-Length: 0

 to 160.124.48.121:5060
Scheduling destruction of call '67c953c4d947368d at 160.124.48.121' in 15000 ms
Found user 'phone1'

Sip read: 
ACK sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe6303ef16f54c20e
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as174e1ffb
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29106 ACK
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

11 headers, 0 lines

Sip read: 
INVITE sip:310 at asterisk.posix.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf444ad4a3ef10de2
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at asterisk.posix.co.za;user=phone", nonce="150a7b85", response="5da7b9abb824ddf96ef073102fac068b"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29107 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

13 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 15
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Found user 'phone1'
Looking for 310 in sip
list_route: hop: <sip:phone1 at 160.124.48.121;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf444ad4a3ef10de2
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29107 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
    -- Executing VoiceMailMain("SIP/phone1-72a7", "") in new stack
We're at 160.124.48.24 port 18550
Answering with capability 2
Answering with capability 4
Answering with capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf444ad4a3ef10de2
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29107 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 32242 32242 IN IP4 160.124.48.24
s=session
c=IN IP4 160.124.48.24
t=0 0
m=audio 18550 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

 to 160.124.48.121:5060

Sip read: 
ACK sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf444ad4a3ef10de2
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", nonce="150a7b85", response="be76d4fff32991e1a2744f57676063c0"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29107 ACK
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 149
v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:40

20 headers, 0 lines
    -- Playing 'vm-login' (language 'en')

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK71f18b358247de8b
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", nonce="150a7b85", response="be76d4fff32991e1a2744f57676063c0"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29108 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=2
Duration=960
13 headers, 2 lines
Receiving DTMF!
May  9 21:40:23 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from 67c953c4d947368d at 160.124.48.121
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK71f18b358247de8b
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29108 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK71f18b358247de8b
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29108 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe927ea2d00009b0a
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", nonce="150a7b85", response="be76d4fff32991e1a2744f57676063c0"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29109 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=0
Duration=960
13 headers, 2 lines
Receiving DTMF!
May  9 21:40:23 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from 67c953c4d947368d at 160.124.48.121
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe927ea2d00009b0a
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29109 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKe927ea2d00009b0a
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29109 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060

Sip read: 
INFO sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK0000000000000000
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", nonce="150a7b85", response="be76d4fff32991e1a2744f57676063c0"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29110 INFO
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=3
Duration=640
13 headers, 2 lines
Receiving DTMF!
May  9 21:40:24 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from 67c953c4d947368d at 160.124.48.121
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK0000000000000000
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29110 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK0000000000000000
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29110 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

 to 160.124.48.121:5060
    -- Username not entered

Sip read: 
BYE sip:310 at 160.124.48.24 SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK7714fe148b35acf8
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Contact: <sip:phone1 at 160.124.48.121;user=phone>
Proxy-Authorization: DIGEST username="phone1", realm="asterisk", algorithm=MD5, uri="sip:310 at 160.124.48.24", nonce="150a7b85", response="daf8e0820515e56931958956deebf344"
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29111 BYE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

12 headers, 0 lines
Sending to 160.124.48.121 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK7714fe148b35acf8
From: "Phone One" <sip:phone1 at asterisk.posix.co.za;user=phone>;tag=bb2c08bc80decffb
To: <sip:310 at asterisk.posix.co.za;user=phone>;tag=as55e008c2
Call-ID: 67c953c4d947368d at 160.124.48.121
CSeq: 29111 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:310 at 160.124.48.24>
Content-Length: 0

------------------ End of BAD Trace- Phone has been hungup ----------------------