[Asterisk-Users] X-Lite to Zap, no Voice on other phone!

Etienne Pretorius etiennep at kingsley.co.za
Mon Apr 4 09:49:02 MST 2005 

Skipped content of type multipart/alternative-------------- next part --------------
    -- Executing Dial("SIP/Reception-a39c", "Zap/g2/0836851650") in new stack
    -- Called g2/0836851650
We're at 192.168.5.71 port 12192
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>;tag=as692cd138
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0836851650 at 192.168.5.71>
Content-Type: application/sdp
Content-Length: 213

v=0p*CLI>
o=root 1994 1994 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 12192 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
CANCEL sip:0836851650 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 CANCEL
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.5.39 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>;tag=as692cd138
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0836851650 at 192.168.5.71>
Content-Length: 0

voip*CLI>
---
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>;tag=as692cd138
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0836851650 at 192.168.5.71>
Content-Length: 0


---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0836851650, 1) exited non-zero on 'SIP/Reception-a39c'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:0836851650 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>;tag=as692cd138
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:0836851650 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:reception at 192.168.5.71>;tag=2918742336
To: <sip:0836851650 at 192.168.5.71>;tag=as692cd138
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39
CSeq: 1371 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '35303165-B432-4857-B05A-22DC8B2943CC at 192.168.5.39'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
-------------- next part --------------
    -- Executing Dial("SIP/Reception-4949", "Zap/g2/0217619930") in new stack
    -- Called g2/0217619930
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 13200
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bKF7B88DB1B9E242C0AC8BBD5F5ABAB233
From: none <sip:reception at 192.168.5.71>;tag=488136580
To: <sip:0217619930 at 192.168.5.71>;tag=as4153afdd
Call-ID: C842AD4D-7CB3-4BE5-846E-3061FC31E9C7 at 192.168.5.39
CSeq: 10930 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0217619930 at 192.168.5.71>
Content-Type: application/sdp
Content-Length: 213

v=0p*CLI>
o=root 1999 1999 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 13200 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-------------- next part --------------
    -- Executing Dial("SIP/Reception-5cf7", "Zap/g2/0828000495") in new stack
    -- Called g2/0828000495
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 15180
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK6F44B5DB845640A18B3AFAE9073643DE
From: none <sip:reception at 192.168.5.71>;tag=2938417032
To: <sip:0828000495 at 192.168.5.71>;tag=as7a0a3237
Call-ID: 5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39
CSeq: 53237 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0828000495 at 192.168.5.71>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1963 1963 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 15180 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Zap/3-1 answered SIP/Reception-5cf7
We're at 192.168.5.71 port 15180
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK6F44B5DB845640A18B3AFAE9073643DE
From: none <sip:reception at 192.168.5.71>;tag=2938417032
To: <sip:0828000495 at 192.168.5.71>;tag=as7a0a3237
Call-ID: 5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39
CSeq: 53237 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0828000495 at 192.168.5.71>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1963 1964 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 15180 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:0828000495 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bKF7CCFB94A67F42669B3C8683544CDCCA
From: none <sip:reception at 192.168.5.71>;tag=2938417032
To: <sip:0828000495 at 192.168.5.71>;tag=as7a0a3237
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39
CSeq: 53237 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0828000495, 1) exited non-zero on 'SIP/Reception-5cf7'
set_destination: Parsing <sip:reception at 192.168.5.39:5060> for address/port to send to
set_destination: set destination to 192.168.5.39, port 5060
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
BYE sip:reception at 192.168.5.39:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.71:5060;branch=z9hG4bK1d37c962;rport
From: <sip:0828000495 at 192.168.5.71>;tag=as7a0a3237
To: none <sip:reception at 192.168.5.71>;tag=2938417032
Contact: <sip:0828000495 at 192.168.5.71>
Call-ID: 5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.5.71:5060;branch=z9hG4bK1d37c962;rport
From: <sip:0828000495 at 192.168.5.71>;tag=as7a0a3237
To: none <sip:reception at 192.168.5.71>;tag=2938417032
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39
CSeq: 102 BYE
Server: X-Lite release 1103m
Content-Length: 0


--- (9 headers 0 lines)---
Response message is BYE
Destroying call '5F9ABEF0-B783-408F-A9A1-ABE3A7B31B58 at 192.168.5.39'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
-------------- next part --------------
--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
INVITE sip:0217619930 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 195

v=0
o=reception 1251812 1251843 IN IP4 192.168.5.39
s=X-Lite
c=IN IP4 192.168.5.39
t=0 0
m=audio 8000 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (11 headers 9 lines)---
Using latest request as basis request
Sending to 192.168.5.39 : 5060 (non-NAT)
Found user 'Reception'
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.5.39:8000
Found description format gsm
Found description format telephone-event
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 0217619930 in sip
list_route: hop: <sip:reception at 192.168.5.39:5060>
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0217619930 at 192.168.5.71>
Content-Length: 0


---
    -- Executing Dial("SIP/Reception-33d9", "Zap/g2/0217619930") in new stack
    -- Called g2/0217619930
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 11850
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>;tag=as6093bccf
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0217619930 at 192.168.5.71>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1958 1958 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 11850 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
CANCEL sip:0217619930 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 CANCEL
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.5.39 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>;tag=as6093bccf
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0217619930 at 192.168.5.71>
Content-Length: 0
voip*CLI>

---
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>;tag=as6093bccf
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0217619930 at 192.168.5.71>
Content-Length: 0


---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0217619930, 1) exited non-zero on 'SIP/Reception-33d9'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:0217619930 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>;tag=as6093bccf
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:0217619930 at 192.168.5.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:reception at 192.168.5.71>;tag=1058526819
To: <sip:0217619930 at 192.168.5.71>;tag=as6093bccf
Contact: <sip:reception at 192.168.5.39:5060>
Call-ID: 3B1077B2-FDE6-43EA-9EEC-95B5E70AE9CD at 192.168.5.39
CSeq: 23215 ACK
Max-Forwards: 70
Content-Length: 0

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