SIP call drops after 10 minutes, 32 seconds using Babytel


#1

I have been using Babytel (a VOIP provider in Canada) with a Sipura 2002 ATA for a while now, and all is fine.

I recently added Asterisk to the mix, sitting between the ATA and Babytel.

Now, any call that goes out to Babytel, it is fine for the first 10 minutes and 32 seconds (10:32). After that, the POTS party at the other end gets dialtone back, and my end just goes quiet.

Internal calls between local extensions (on ATA boxes) are OK. I don’t get incoming calls via the VOIP provider, so I can’t say if it works or not.

Here is what little info I could get off the console:

mtvCLI>
– Executing SetCallerID(“SIP/phoneb-bc41”, “
<1-514-448-xxxx>”) in new stack
– Executing Dial(“SIP/phoneb-bc41”, “SIP/1519xxxxxxx@sipout”) in new
stack
– Called 1xxxxxxxxxx@sipout
– SIP/sipout-9a0e is making progress passing it to SIP/phoneb-bc41
– SIP/sipout-9a0e is ringing
– SIP/sipout-9a0e is making progress passing it to SIP/phoneb-bc41
– SIP/sipout-9a0e answered SIP/phoneb-bc41
– Attempting native bridge of SIP/phoneb-bc41 and SIP/sipout-9a0e
mtv
CLI> ** NO NEW MESSAGES ON CONSOLE WHEN THE CALL DIES **
mtvCLI>
mtv
CLI> ** THIS IS WHAT IT SHOWS WHEN I HANG UP on the ATA AFTER THE CALL GOES DEAD **
== Spawn extension (localgroup, xxxxxxxxxxxxx, 2) exited non-zero on
’SIP/phoneb-bc41’
– Got SIP response 481 “Call/Transaction Does Not Exist” back from
216.18.125.7
– Starting simple switch on ‘Zap/1-1’

I used the sample/demo sip & extension files as my base.

I added: sip.conf:
register => 1514448xxxx:secret@sip.babytel.ca:5065/1514448xxxx

BabyTEL out

[sipout]
context=from-pstn
type=peer
insecure=very
secret=secretpwd
dmtfmode=inband
username=1514448xxxx
fromuser=1514448xxxx
fromdomain=sip.babytel.ca
host=sip.babytel.ca
port=5065

In extensions.conf:
exten => _9.,1,SetCallerID(Jeff <1-514-448-xxxx>)
exten => _9.,2,Dial(SIP/${EXTEN:1}@sipout)

The rest of the additions are for a meetme conference extension, and a couple of ATA ports.

Any ideas of why the call drops thru asterisk?
Thanks
Jeff


#2

have you contacted your provider about this ? there are sometimes “conflicts” between asterisk and other VoIP platforms that can have weird effects, but usually a chat with the techies at their end will fix it.


#3

I have not yet contected Babytel about this issue, as they will no longer give out the SIP passwords, etc., and are reluctant to offer help to those not using their devices.

I was hoping someone might have some ideas before I get in touch with them.

Thanks
Jeff


#4

You need to enable sip debugging and see what message is sent/recieved

Ian


#5

Thanks for the info. I am still learning the SIP protocol, so if you can offer some help, here are the logs for a sample call.

<-- SIP read from 192.168.1.24:5061:
INVITE sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-7cd48196
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 101 INVITE
Max-Forwards: 70
Contact: phonee sip:phonee@192.168.1.24:5061
Expires: 240
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 23424582 23424582 IN IP4 192.168.1.24
s=-
c=IN IP4 192.168.1.24
t=0 0
m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

— (14 headers 19 lines)—
Using INVITE request as basis request - 261ab277-1a0338ba@192.168.1.24
Sending to 192.168.1.24 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-7cd48196;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as76209639
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Proxy-Authenticate: Digest realm=“asterisk”, nonce="5a14dd07"
Content-Length: 0


Scheduling destruction of call ‘261ab277-1a0338ba@192.168.1.24’ in 15000 ms
Found user 'phonee’
mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
ACK sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-7cd48196
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as76209639
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 101 ACK
Max-Forwards: 70
Contact: phonee sip:phonee@192.168.1.24:5061
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0

— (10 headers 0 lines)—
mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
INVITE sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-2c4922a6
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“phonee”,realm=“asterisk”,nonce=“5a14dd07”,uri="sip:95145551111@192.168.1.9",algorithm=MD5,response="d924c5fbfa8f59bf1ed6041947928737"
Contact: phonee sip:phonee@192.168.1.24:5061
Expires: 240
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 23424582 23424582 IN IP4 192.168.1.24
s=-
c=IN IP4 192.168.1.24
t=0 0
m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

— (15 headers 19 lines)—
Using INVITE request as basis request - 261ab277-1a0338ba@192.168.1.24
Sending to 192.168.1.24 : 5061 (non-NAT)
Found user 'phonee’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.24:16446
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 95145551111 in localgroup (domain 192.168.1.9)
list_route: hop: sip:phonee@192.168.1.24:5061
Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-2c4922a6;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Content-Length: 0


-- Executing SetCallerID("SIP/phonee-5aa9", "Jeff <1-514-448-xxxx>") in new stack
-- Executing Dial("SIP/phonee-5aa9", "SIP/5145551111@sipout") in new stack

We’re at 192.168.1.9 port 18816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
INVITE sip:5145551111@sip.babytel.ca:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK29604b73;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 18 Mar 2006 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 4047 4047 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 18816 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


-- Called 5145551111@sipout

mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 100 Trying
To: sip:5145551111@sip.babytel.ca:5065
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK29604b73
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 102 INVITE
Server: DitechComCorp-PeerPoint/3.5.24 ga-0
Content-Length: 0

— (8 headers 0 lines)—
mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 407 Proxy Authentication Required
To: sip:5145551111@sip.babytel.ca:5065;tag=8280bb0e
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK29604b73
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“sip.babytel.ca”,nonce="441c6017d898038058d3ddf51a2b680d5d88e44a"
Content-Length: 0

— (8 headers 0 lines)—
Transmitting (no NAT) to 216.18.125.7:5065:
ACK sip:5145551111@sip.babytel.ca:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK29604b73;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=8280bb0e
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


We’re at 192.168.1.9 port 18816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
INVITE sip:5145551111@sip.babytel.ca:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2c446fa3;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=“1514448xxxx”, realm=“sip.babytel.ca”, algorithm=MD5, uri=“sip:5145551111@sip.babytel.ca:5065”, nonce=“441c6017d898038058d3ddf51a2b680d5d88e44a”, response=“0365faf28aaf9aa36a88bc7c4ff3792d”, opaque=""
Date: Sat, 18 Mar 2006 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 4047 4048 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 18816 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 100 Trying
To: sip:5145551111@sip.babytel.ca:5065
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2c446fa3
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 INVITE
Server: DitechComCorp-PeerPoint/3.5.24 ga-0
Content-Length: 0

— (8 headers 0 lines)—
mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 183 Session Progress
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2c446fa3
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 INVITE
Contact: sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Content-Type: application/sdp
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.40.200.371
Supported: em, timer, replaces, path
Content-Length: 226

v=0
o=AudiocodesGW 543119 453311 IN IP4 216.18.125.7
s=Phone-Call
c=IN IP4 216.18.125.7
t=0 0
m=audio 23088 RTP/AVP 0 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

— (12 headers 11 lines)—
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.18.125.7:23088
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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)
– SIP/sipout-322c is making progress passing it to SIP/phonee-5aa9
We’re at 192.168.1.9 port 13910
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-2c4922a6;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 4047 4047 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 13910 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 180 Ringing
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2c446fa3
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 INVITE
Contact: sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Content-Type: application/sdp
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.40.200.371
Supported: em, timer, replaces, path
Content-Length: 226

v=0
o=AudiocodesGW 543119 453311 IN IP4 216.18.125.7
s=Phone-Call
c=IN IP4 216.18.125.7
t=0 0
m=audio 23088 RTP/AVP 0 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

— (12 headers 11 lines)—
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.18.125.7:23088
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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)
– SIP/sipout-322c is ringing
Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-2c4922a6;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Content-Length: 0


-- SIP/sipout-322c is making progress passing it to SIP/phonee-5aa9

mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 200 OK
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2c446fa3
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 INVITE
Contact: sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Content-Type: application/sdp
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.40.200.371
Supported: em, timer, replaces, path
Content-Length: 230

v=0
o=AudiocodesGW 543119 2118728360 IN IP4 216.18.125.7
s=Phone-Call
c=IN IP4 216.18.125.7
t=0 0
m=audio 23088 RTP/AVP 0 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

— (12 headers 11 lines)—
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.18.125.7:23088
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
set_destination: Parsing sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
Transmitting (no NAT) to 216.18.125.7:5065:
ACK sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3a7052a4;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


-- SIP/sipout-322c answered SIP/phonee-5aa9

We’re at 192.168.1.9 port 13910
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-2c4922a6;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 4047 4048 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 13910 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


-- Attempting native bridge of SIP/phonee-5aa9 and SIP/sipout-322c

set_destination: Parsing sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
We’re at 192.168.1.9 port 18816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 16 lines
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
INVITE sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3b3dee19;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
ontent-Type: application/sdp
Content-Length: 364

v=0
o=root 4047 4049 IN IP4 192.168.1.24
s=session
c=IN IP4 192.168.1.24
t=0 0
m=audio 16446 RTP/AVP 0 4 8 111 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
ACK sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-58a873b8
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“phonee”,realm=“asterisk”,nonce=“5a14dd07”,uri="sip:95145551111@192.168.1.9",algorithm=MD5,response="d7b8d43e5dd091298b5c4d37df2cba9c"
Contact: phonee sip:phonee@192.168.1.24:5061
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0

— (11 headers 0 lines)—
set_destination: Parsing sip:phonee@192.168.1.24:5061 for address/port to send to
set_destination: set destination to 192.168.1.24, port 5061
We’re at 192.168.1.9 port 13910
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.1.24:5061:
INVITE sip:phonee@192.168.1.24:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5411a79b;rport
From: sip:95145551111@192.168.1.9;tag=as4ff193fd
To: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
Contact: sip:95145551111@192.168.1.9
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 4047 4049 IN IP4 216.18.125.7
s=session
c=IN IP4 216.18.125.7
t=0 0
m=audio 23088 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
SIP/2.0 200 OK
To: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
From: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5411a79b
Contact: phonee sip:phonee@192.168.1.24:5061
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 235
Content-Type: application/sdp

v=0
o=- 23425961 23425961 IN IP4 192.168.1.24
s=-
c=IN IP4 192.168.1.24
t=0 0
m=audio 16446 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

— (10 headers 12 lines)—
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.24:16446
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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:phonee@192.168.1.24:5061
set_destination: Parsing sip:phonee@192.168.1.24:5061 for address/port to send to
set_destination: set destination to 192.168.1.24, port 5061
Transmitting (no NAT) to 192.168.1.24:5061:
ACK sip:phonee@192.168.1.24:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK73d9d24e;rport
From: sip:95145551111@192.168.1.9;tag=as4ff193fd
To: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
Contact: sip:95145551111@192.168.1.9
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 200 OK
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3b3dee19
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 104 INVITE
Contact: sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Content-Type: application/sdp
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.40.200.371
Supported: em, timer, replaces, path
Content-Length: 230

v=0
o=AudiocodesGW 543119 1903294079 IN IP4 216.18.125.7
s=Phone-Call
c=IN IP4 216.18.125.7
t=0 0
m=audio 23088 RTP/AVP 0 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

— (12 headers 11 lines)—
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.18.125.7:23088
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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)
set_destination: Parsing sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
Transmitting (no NAT) to 216.18.125.7:5065:
ACK sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK56859ab5;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
BYE sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-cfb0289
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username=“phonee”,realm=“asterisk”,nonce=“5a14dd07”,uri="sip:95145551111@192.168.1.9",algorithm=MD5,response="b12082b4a1ec9b3af7e594d54b53ceea"
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0

— (10 headers 0 lines)—
Sending to 192.168.1.24 : 5061 (non-NAT)
Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-cfb0289;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=bead2f534fa9818eo1
To: sip:95145551111@192.168.1.9;tag=as4ff193fd
Call-ID: 261ab277-1a0338ba@192.168.1.24
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


set_destination: Parsing sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
We’re at 192.168.1.9 port 18816
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
INVITE sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5a205304;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 210

v=0
o=root 4047 4050 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 18816 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


== Spawn extension (localgroup, 95145551111, 2) exited non-zero on 'SIP/phonee-5aa9’
mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 500 Server Internal Error
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
From: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5a205304
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 105 INVITE
Content-Length: 0

— (7 headers 0 lines)—
– Got SIP response 500 “Server Internal Error” back from 216.18.125.7
set_destination: Parsing sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
Transmitting (no NAT) to 216.18.125.7:5065:
ACK sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5a205304;rport
From: “Jeff” sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
To: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Contact: sip:1514448xxxx@192.168.1.9
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 105 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Destroying call '7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
Destroying call '261ab277-1a0338ba@192.168.1.24’
mtv*CLI>
<-- SIP read from 216.18.125.7:5065:
BYE sip:1514448xxxx@192.168.1.9 SIP/2.0
To: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
From: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-2fe3e652ca481c50a6b7-1-cHBiZTM1NGEzZWM4ZDEwNjdlNzM0ZA…-d87543-
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 176906050 BYE
Contact: sip:GR110PkGoj2scHJ1iGmgGeKVCCn3hQqFrs.@216.18.125.7:5065
Max-Forwards: 70
Content-Length: 0

mtv*CLI>
— (9 headers 0 lines)—
Sending to 216.18.125.7 : 5065 (non-NAT)
Transmitting (no NAT) to 216.18.125.7:5065:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-2fe3e652ca481c50a6b7-1-cHBiZTM1NGEzZWM4ZDEwNjdlNzM0ZA…-d87543-;received=216.18.125.7
From: sip:5145551111@sip.babytel.ca:5065;tag=5a6a4926
To: "Jeff"sip:1514448xxxx@sip.babytel.ca;tag=as2cdb8e63
Call-ID: 7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
CSeq: 176906050 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Destroying call '7870ee7b43a0e8aa785b494a4d25c8ff@sip.babytel.ca
mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
INVITE sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-fcf5f529
From: phonee sip:phonee@192.168.1.9;tag=b8d1843b9d33f959o1
To: sip:95145551111@192.168.1.9
Call-ID: 4230df63-9d341851@192.168.1.24
CSeq: 101 INVITE
Max-Forwards: 70
Contact: phonee sip:phonee@192.168.1.24:5061
Expires: 240
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 23492057 23492057 IN IP4 192.168.1.24
s=-
c=IN IP4 192.168.1.24
t=0 0
m=audio 16448 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

— (14 headers 19 lines)—
Using INVITE request as basis request - 4230df63-9d341851@192.168.1.24
Sending to 192.168.1.24 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.24:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-fcf5f529;received=192.168.1.24
From: phonee sip:phonee@192.168.1.9;tag=b8d1843b9d33f959o1
To: sip:95145551111@192.168.1.9;tag=as71efa4cb
Call-ID: 4230df63-9d341851@192.168.1.24
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:95145551111@192.168.1.9
Proxy-Authenticate: Digest realm=“asterisk”, nonce="748cca6c"
Content-Length: 0


Scheduling destruction of call ‘4230df63-9d341851@192.168.1.24’ in 15000 ms
Found user 'phonee’
mtv*CLI>
<-- SIP read from 192.168.1.24:5061:
ACK sip:95145551111@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5061;branch=z9hG4bK-fcf5f529
From: phonee sip:phonee@192.168.1.9;tag=b8d1843b9d33f959o1
To: sip:95145551111@192.168.1.9;tag=as71efa4cb
Call-ID: 4230df63-9d341851@192.168.1.24
CSeq: 101 ACK
Max-Forwards: 70
Contact: phonee sip:phonee@192.168.1.24:5061
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0



#6

I have not been able to solve this as of yet.

I disabled the ‘ReInvite’ feature, but it did not help.

Any suggestions would be great!

Thanks
Jeff