thr3ads.net - asterisk users - [Asterisk-Users] no audio between sip channels * 1.2.6 [Apr 2006]

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John Millican
2006-Apr-02 10:38 UTC
[Asterisk-Users] no audio between sip channels * 1.2.6

Hello all,
I am running * 1.2.6 I have 2 linksys PAP2 with two phones each.  Until 
recently all was good.  on Friday I was running 1.2.5 when I added the fourth 
phone. I have to admit to initially wiring the rj11(crossed wires) wrong the 
first time but other than that nothing I can think of.  Added the appropriate 
entries in sip.con and on the PAP2.  I then tried to call from one line to 
the other and no sound.  Okay must have screwed something up.  checked 
sip.conf all looked good.  Okay good time to go to 1.2.6, still no audio.  
All phones ring and answer but no audio.  the last thing that apears on the 
console is "attempting native bridge of sip/677-xxxx and sip/699-xxxx"
below is a debug of a call and sip.conf.  Each channel on the PAP2's is set
to
a different port 5060 through 5063.  I can call in to any phone and all is 
good, use any phone to call to POTS line and back in on second POTS line and 
all is good.   I have been looking through the archive of the mail list that 
I keep and have not found anything to fix my problems yet.
i have transfered the registration of both PAP2's to a 1.2.0 system that I 
have and everything works as it should.  moved 1.2.0 configs to 1.2.6 box and 
again no audio between sip channels.

*CLI> sip debug
SIP Debugging enabled
*CLI>
<-- SIP read from 192.168.1.200:5060:
INVITE sip:699@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 101 INVITE
Max-Forwards: 70
Contact: John Millican <sip:677@192.168.1.200:5060>
Expires: 240
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 235
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 143361 143361 IN IP4 192.168.1.200
s=-
c=IN IP4 192.168.1.200
t=0 0
m=audio 16410 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (14 headers 12 lines)---
Using INVITE request as basis request - 23a92fd3-a3463cc9@192.168.1.200
Sending to 192.168.1.200 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.200:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
192.168.1.200:5060;branch=z9hG4bK-75990941;received=192.168.1.200
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as0767a869
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:699@192.168.1.10>
Proxy-Authenticate: Digest realm="asterisk",
nonce="6e91851e"
Content-Length: 0


---
Scheduling destruction of call '23a92fd3-a3463cc9@192.168.1.200' in
15000 ms
Found user '677'

<-- SIP read from 192.168.1.200:5060:
ACK sip:699@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as0767a869
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 101 ACK
Max-Forwards: 70
Contact: John Millican <sip:677@192.168.1.200:5060>
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0


--- (10 headers 0 lines)---

<-- SIP read from 192.168.1.200:5060:
INVITE sip:699@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest 
username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="121d27cf19808e8a097930f0f969d3d7"
Contact: John Millican <sip:677@192.168.1.200:5060>
Expires: 240
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 235
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 143361 143361 IN IP4 192.168.1.200
s=-
c=IN IP4 192.168.1.200
t=0 0
m=audio 16410 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 12 lines)---
Using INVITE request as basis request - 23a92fd3-a3463cc9@192.168.1.200
Sending to 192.168.1.200 : 5060 (non-NAT)
Found user '677'
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.200:16410
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), 
combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 699 in pap2 (domain 192.168.1.10)
list_route: hop: <sip:677@192.168.1.200:5060>
Transmitting (no NAT) to 192.168.1.200:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:699@192.168.1.10>
Content-Length: 0


---
    -- Executing Dial("SIP/677-bd65", "sip/699") in new
stack
We're at 192.168.1.10 port 12042
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 192.168.1.201:5062:
INVITE sip:699@192.168.1.201:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0;rport
From: "John Millican" <sip:677@pap2>;tag=as604497e3
To: <sip:699@192.168.1.201:5062>
Contact: <sip:677@192.168.1.10>
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 02 Apr 2006 17:10:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 158

v=0
o=root 5051 5051 IN IP4 192.168.1.10
s=session
c=IN IP4 192.168.1.10
t=0 0
m=audio 12042 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
    -- Called 699

<-- SIP read from 192.168.1.201:5062:
SIP/2.0 100 Trying
To: <sip:699@192.168.1.201:5062>
From: "John Millican" <sip:677@pap2>;tag=as604497e3
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.1.201:5062:
SIP/2.0 180 Ringing
To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0
From: "John Millican" <sip:677@pap2>;tag=as604497e3
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/699-8c1a is ringing
Transmitting (no NAT) to 192.168.1.200:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as7abbbe62
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:699@192.168.1.10>
Content-Length: 0


---

<-- SIP read from 192.168.1.201:5062:
SIP/2.0 200 OK
To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0
From: "John Millican" <sip:677@pap2>;tag=as604497e3
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0
Contact: Sentinel Communications <sip:699@192.168.1.201:5062>
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 13282 13282 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 16444 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.201:16444
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), 
combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:699@192.168.1.201:5062>
set_destination: Parsing <sip:699@192.168.1.201:5062> for address/port to
send
to
set_destination: set destination to 192.168.1.201, port 5062
Transmitting (no NAT) to 192.168.1.201:5062:
ACK sip:699@192.168.1.201:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3db4b100;rport
From: "John Millican" <sip:677@pap2>;tag=as604497e3
To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0
Contact: <sip:677@192.168.1.10>
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/699-8c1a answered SIP/677-bd65
We're at 192.168.1.10 port 11982
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.200:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as7abbbe62
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:699@192.168.1.10>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 5051 5051 IN IP4 192.168.1.10
s=session
c=IN IP4 192.168.1.10
t=0 0
m=audio 11982 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/677-bd65 and SIP/699-8c1a

<-- SIP read from 192.168.1.200:5060:
ACK sip:699@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-e7eb5b44
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as7abbbe62
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest 
username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="36f62fd39cac4190c11bc2b1fa7cb31b"
Contact: John Millican <sip:677@192.168.1.200:5060>
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0


--- (11 headers 0 lines)---

<-- SIP read from 192.168.1.200:5060:
BYE sip:699@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-cd302779
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as7abbbe62
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest 
username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="fe5a3ed0d6a5e2b18071b0784b4c0c80"
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.1.200 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.200:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.1.200:5060;branch=z9hG4bK-cd302779;received=192.168.1.200
From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0
To: <sip:699@192.168.1.10>;tag=as7abbbe62
Call-ID: 23a92fd3-a3463cc9@192.168.1.200
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:699@192.168.1.10>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:699@192.168.1.201:5062> for address/port to
send
to
set_destination: set destination to 192.168.1.201, port 5062
Reliably Transmitting (no NAT) to 192.168.1.201:5062:
BYE sip:699@192.168.1.201:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69a7640a;rport
From: "John Millican" <sip:677@pap2>;tag=as604497e3
To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0
Contact: <sip:677@192.168.1.10>
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (pap2, 699, 1) exited non-zero on 'SIP/677-bd65'

<-- SIP read from 192.168.1.201:5062:
SIP/2.0 200 OK
To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0
From: "John Millican" <sip:677@pap2>;tag=as604497e3
Call-ID: 400784d110e4e314373334be31ee3576@pap2
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69a7640a
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '400784d110e4e314373334be31ee3576@pap2'
Destroying call '23a92fd3-a3463cc9@192.168.1.200'

*CLI> sip no debug


[general]
context=default			
port=5060			
bindaddr=0.0.0.0		
srvlookup=yes

[677]
type=friend
;username=John Millican
;fromuser=677
secret=hell99
host=dynamic
callerid=John <677>
fromdomain=pap2
context=pap2
dtmfmode=inband
disallow=all
allow=ulaw
mailbox=4700
canreinvite=no
nat=no

[666]
type=friend
secret=xxxxxx
host=dynamic
callerid=Kelly <666>
fromdomain=pap2
context=pap2
dtmfmode=inband
disallow=all
allow=ulaw
mailbox=4800
canreinvite=no
nat=no

[699]
type=friend
secret=sentinel1
host=dynamic
callerid=Sentinel 1 <603xxxxxxx>
fromdomain=pap2
context=pap2
dtmfmode=inband
disallow=all
allow=ulaw
mailbox=9010
canreinvite=no
nat=no

<flame suit> If I have missed something stupid feel free to rub my nose in
it</flame suit>
Any help would be greatly appreciated.
John M
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