Hi david,
Thanks for your answer. Regarding the “retransmission”, I don’t experience that. I just wanted to state that I (most likely) don’t have a NAT issue.
regarding the security issue: I reckon that type=friend is not necessary (I thought it was, but it’s still working with type=peer and insecure=invite). However, I think I need insecure=invite as my voip-provider is not registering itsself with me when it dispatches an incoming call to me.
However, I’ve locked the guest-voip accounts in the context [external] from which it is not possible to make a toll call. Additionally, I’ve limited the IPs allowed to connect to my to my voip-provider’s subnet (both with deny / permit and on a firewall level). I thought that’s pretty much all I can do - am I wrong?
I wish I had a single SIP-trunk, but this would cost me an additional $20 per months, which I’m not willing to spare - everything works at that end, though I probably have a little unnecessary traffic on my wan connection.
Also thanks for your input regarding nat=. I’ve removed nat= completely from my sip.conf - localnet and externhost do all the magic.
Finally, about that sip trace: here it is. The sip trace shows a call from phone 1 to the external number, which then rings on phone2 and is answered. There is no audio going trough. Then the call is ended.
SIP Debugging enabled
<--- SIP read from UDP:192.168.0.233:54974 --->
INVITE sip:0123465789@nas.weily.lan SIP/2.0
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 2658 INVITE
From: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
To: <sip:0123465789@nas.weily.lan>
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bKb9c17f3598fa47bc5ee5ce971264bda7373033;rport
Max-Forwards: 70
Contact: "phone1" <sip:phone1@192.168.0.233:54974;transport=udp>
Content-Type: application/sdp
Content-Length: 299
v=0
o=- 1371933378566 1371933378568 IN IP4 192.168.0.233
s=-
c=IN IP4 192.168.0.233
t=0 0
m=audio 39384 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (10 headers 13 lines) ---
Sending to 192.168.0.233:54974 (NAT)
Using INVITE request as basis request - 9f42d2a53406e625509057427a28d5aa@192.168.0.233
Found peer 'phone1' for 'phone1' from 192.168.0.233:54974
== Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 127
Found unknown media description format GSM-EFR for ID 96
Found unknown media description format AMR for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 127
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.233:39384
Looking for 0123465789 in internal (domain nas.weily.lan)
list_route: hop: <sip:phone1@192.168.0.233:54974;transport=udp>
<--- Transmitting (NAT) to 192.168.0.233:54974 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bKb9c17f3598fa47bc5ee5ce971264bda7373033;received=192.168.0.233;rport=54974
From: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
To: <sip:0123465789@nas.weily.lan>
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 2658 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@192.168.0.3:5060>
Content-Length: 0
<------------>
-- Executing [0123465789@internal:1] Dial("SIP/phone1-0000001e", "SIP/0123465789@0123465789") in new stack
== Using SIP RTP CoS mark 5
Audio is at 13414
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.117.203.47:5060:
INVITE sip:0123465789@guest2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 80.219.136.161:5060;branch=z9hG4bK0a8b87b9;rport
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@80.219.136.161>;tag=as74b9a0bd
To: <sip:0123465789@guest2.voipgateway.org>
Contact: <sip:0123465789@80.219.136.161:5060>
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Date: Sat, 22 Jun 2013 20:36:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1222134708 1222134708 IN IP4 80.219.136.161
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 80.219.136.161
t=0 0
m=audio 13414 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/0123465789@0123465789
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0a8b87b9;rport=5060
To: <sip:0123465789@guest2.voipgateway.org>
From: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0a8b87b9;rport=5060
Record-Route: <sip:212.117.203.47;lr>
To: <sip:0123465789@guest2.voipgateway.org>
From: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="guest2.voipgateway.org",nonce="3cce6f0854c7e80a57ec911beaf9d8b2c80d"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:0123465789@guest2.voipgateway.org> for address/port to send to
set_destination: set destination to 212.117.203.47:5060
Transmitting (NAT) to 212.117.203.47:5060:
ACK sip:0123465789@guest2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 80.219.136.161:5060;branch=z9hG4bK0a8b87b9;rport
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@80.219.136.161>;tag=as74b9a0bd
To: <sip:0123465789@guest2.voipgateway.org>
Contact: <sip:0123465789@80.219.136.161:5060>
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Content-Length: 0
---
Audio is at 13414
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.117.203.47:5060:
INVITE sip:0123465789@guest2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 80.219.136.161:5060;branch=z9hG4bK0e8e95ec;rport
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@80.219.136.161>;tag=as74b9a0bd
To: <sip:0123465789@guest2.voipgateway.org>
Contact: <sip:0123465789@80.219.136.161:5060>
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Authorization: Digest username="0123465789", realm="guest2.voipgateway.org", algorithm=MD5, uri="sip:0123465789@guest2.voipgateway.org", nonce="3cce6f0854c7e80a57ec911beaf9d8b2c80d", response="6954cf880fdddf589e3178689dcbd127"
Date: Sat, 22 Jun 2013 20:36:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1222134708 1222134709 IN IP4 80.219.136.161
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 80.219.136.161
t=0 0
m=audio 13414 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0e8e95ec;rport=5060
To: <sip:0123465789@guest2.voipgateway.org>
From: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 103 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:212.117.203.47:5060 --->
INVITE sip:0123465789@192.168.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-ce32f95014328a74-1---d8754z-;rport
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-egcpsxgheepjccle;rport=5061
Max-Forwards: 69
Record-Route: <sip:212.117.203.47;lr>
Contact: "Anonymous"<sip:212.117.203.47:5061>
To: <sip:0123465789@212.117.203.47>
From: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 579 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
h323-conf-id: 3509096997-3091581407-1494187459-2897649152
cisco-GUID: 3509096997-3091581407-1494187459-2897649152
Content-Length: 251
v=0
o=Sippy 738974032 0 IN IP4 80.219.136.161
s=Asterisk PBX 1.8.13.1~dfsg-3
t=0 0
m=audio 13414 RTP/AVP 8 0 101
c=IN IP4 80.219.136.161
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (17 headers 11 lines) ---
Sending to 212.117.203.47:5060 (NAT)
Using INVITE request as basis request - 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
Found peer '0123465789' for '0123465789' from 212.117.203.47:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.219.136.161:13414
Looking for 0123465789 in external (domain 192.168.0.3)
list_route: hop: <sip:212.117.203.47;lr>
<--- Transmitting (NAT) to 212.117.203.47:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-ce32f95014328a74-1---d8754z-;received=212.117.203.47;rport=5060
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-egcpsxgheepjccle;rport=5061
Record-Route: <sip:212.117.203.47;lr>
From: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
To: <sip:0123465789@212.117.203.47>
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 579 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@80.219.136.161:5060>
Content-Length: 0
<------------>
-- Executing [0123465789@external:1] Dial("SIP/0123465789-00000020", "SIP/phone2") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17650
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.107:5060:
INVITE sip:phone2@192.168.0.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK31f9ac2e;rport
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
To: <sip:phone2@192.168.0.107:5060;transport=udp>
Contact: <sip:0123465789@192.168.0.3:5060>
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Date: Sat, 22 Jun 2013 20:36:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 1628457472 1628457472 IN IP4 192.168.0.3
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 192.168.0.3
t=0 0
m=audio 17650 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/phone2
<--- SIP read from UDP:192.168.0.107:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK31f9ac2e;rport=5060;received=192.168.0.3
From: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
To: <sip:phone2@192.168.0.107:5060;transport=udp>;tag=2512611503
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Phone2" <sip:phone2@192.168.0.107:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A2812>"
Server: Aastra 55i/3.2.2.3077
Supported: path
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:phone2@192.168.0.107:5060;transport=udp>
-- SIP/phone2-00000021 is ringing
<--- Transmitting (NAT) to 212.117.203.47:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-ce32f95014328a74-1---d8754z-;received=212.117.203.47;rport=5060
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-egcpsxgheepjccle;rport=5061
Record-Route: <sip:212.117.203.47;lr>
From: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
To: <sip:0123465789@212.117.203.47>;tag=as3a9cd917
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 579 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@80.219.136.161:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0e8e95ec;rport=5060
Record-Route: <sip:212.117.203.47;lr>
To: <sip:0123465789@guest2.voipgateway.org>;tag=oyefjjxy5eqerzze.i
From: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:212.117.203.47;lr>
-- SIP/0123465789-0000001f is ringing
<--- Transmitting (NAT) to 192.168.0.233:54974 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bKb9c17f3598fa47bc5ee5ce971264bda7373033;received=192.168.0.233;rport=54974
From: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
To: <sip:0123465789@nas.weily.lan>;tag=as30a62f26
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 2658 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@192.168.0.3:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.0.107:5060 --->
SUBSCRIBE sip:192.168.0.3:5060 SIP/2.0
Accept: application/dialog-info+xml
Via: SIP/2.0/UDP 192.168.0.107;branch=z9hG4bKea12bc7c0fe47fd26
Max-Forwards: 70
From: "Phone 2" <sip:phone2@192.168.0.3:5060>;tag=ab5a85c7f2
To: <sip:192.168.0.3:5060>
Call-ID: 3cfc4c0de2aecf1f
CSeq: 168 SUBSCRIBE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Phone 2" <sip:phone2@192.168.0.107:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A2812>"
Event: dialog
Expires: 593
Supported: path
User-Agent: Aastra 55i/3.2.2.3077
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Creating new subscription
Sending to 192.168.0.107:5060 (NAT)
list_route: hop: <sip:phone2@192.168.0.107:5060;transport=udp>
Found peer 'phone2' for 'phone2' from 192.168.0.107:5060
Looking for s in subscribe (domain 192.168.0.3)
<--- Transmitting (NAT) to 192.168.0.107:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.107;branch=z9hG4bKea12bc7c0fe47fd26;received=192.168.0.107;rport=5060
From: "Phone 2" <sip:phone2@192.168.0.3:5060>;tag=ab5a85c7f2
To: <sip:192.168.0.3:5060>;tag=as78a83697
Call-ID: 3cfc4c0de2aecf1f
CSeq: 168 SUBSCRIBE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
------------>
Really destroying SIP dialog '3cfc4c0de2aecf1f' Method: SUBSCRIBE
<--- SIP read from UDP:192.168.0.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK31f9ac2e;rport=5060;received=192.168.0.3
From: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
To: <sip:phone2@192.168.0.107:5060;transport=udp>;tag=2512611503
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Phone 2" <sip:phone2@192.168.0.107:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A2812>"
Server: Aastra 55i/3.2.2.3077
Supported: path, replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=MxSIP 0 1 IN IP4 192.168.0.107
s=SIP Call
c=IN IP4 192.168.0.107
t=0 0
m=audio 3000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.107:3000
list_route: hop: <sip:phone2@192.168.0.107:5060;transport=udp>
set_destination: Parsing <sip:phone2@192.168.0.107:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.107:5060
Transmitting (NAT) to 192.168.0.107:5060:
ACK sip:phone2@192.168.0.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK44138b25;rport
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
To: <sip:phone2@192.168.0.107:5060;transport=udp>;tag=2512611503
Contact: <sip:0123465789@192.168.0.3:5060>
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Content-Length: 0
---
-- SIP/phone2-00000021 answered SIP/0123465789-00000020
Audio is at 13444
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 212.117.203.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-ce32f95014328a74-1---d8754z-;received=212.117.203.47;rport=5060
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-egcpsxgheepjccle;rport=5061
Record-Route: <sip:212.117.203.47;lr>
From: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
To: <sip:0123465789@212.117.203.47>;tag=as3a9cd917
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 579 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@80.219.136.161:5060>
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1569894218 1569894218 IN IP4 80.219.136.161
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 80.219.136.161
t=0 0
m=audio 13444 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
-- Locally bridging SIP/0123465789-00000020 and SIP/phone2-00000021
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0e8e95ec;rport=5060
Record-Route: <sip:212.117.203.47;lr>
Contact: "Anonymous"<sip:212.117.203.47:5061>
To: <sip:0123465789@guest2.voipgateway.org>;tag=oyefjjxy5eqerzze.i
From: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 103 INVITE
Content-Type: application/sdp
Server: Sippy
Content-Length: 250
v=0
o=Sippy 21596856 2 IN IP4 80.219.136.161
s=Asterisk PBX 1.8.13.1~dfsg-3
t=0 0
m=audio 13444 RTP/AVP 8 0 101
c=IN IP4 80.219.136.161
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.219.136.161:13444
list_route: hop: <sip:212.117.203.47;lr>
set_destination: Parsing <sip:212.117.203.47;lr> for address/port to send to
set_destination: set destination to 212.117.203.47:5060
Transmitting (NAT) to 212.117.203.47:5060:
ACK sip:212.117.203.47:5061 SIP/2.0
Via: SIP/2.0/UDP 80.219.136.161:5060;branch=z9hG4bK759c81fb;rport
Route: <sip:212.117.203.47;lr>
Max-Forwards: 70
From: "Phone 1" <sip:0123465789@80.219.136.161>;tag=as74b9a0bd
To: <sip:0123465789@guest2.voipgateway.org>;tag=oyefjjxy5eqerzze.i
Contact: <sip:0123465789@80.219.136.161:5060>
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Content-Length: 0
---
-- SIP/0123465789-0000001f answered SIP/phone1-0000001e
Audio is at 10242
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.0.233:54974 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bKb9c17f3598fa47bc5ee5ce971264bda7373033;received=192.168.0.233;rport=54974
From: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
To: <sip:0123465789@nas.weily.lan>;tag=as30a62f26
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 2658 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0123465789@192.168.0.3:5060>
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 362708731 362708731 IN IP4 192.168.0.3
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 192.168.0.3
t=0 0
m=audio 10242 RTP/AVP 8 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=sendrecv
<------------>
-- Locally bridging SIP/phone1-0000001e and SIP/0123465789-0000001f
<--- SIP read from UDP:212.117.203.47:5060 --->
ACK sip:0123465789@192.168.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-50556d5e44897152-1---d8754z-;rport
Via: SIP/2.0/UDP 212.117.203.47:5061;rport=5061;branch=z9hG4bK-z5w52vfrek2tjdpa
Max-Forwards: 69
To: <sip:0123465789@212.117.203.47>;tag=as3a9cd917
From: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 579 ACK
User-Agent: Sippy
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.233:54974 --->
ACK sip:0123465789@192.168.0.3:5060 SIP/2.0
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 2658 ACK
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bK52d087a35bf086d47b980841c0055a7e373033
From: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
To: <sip:0123465789@nas.weily.lan>;tag=as30a62f26
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.233:54974 --->
OPTIONS sip:0123465789@nas.weily.lan SIP/2.0
Call-ID: 7f600b6f5dd49f3191485b2221770e6e@192.168.0.233
CSeq: 3370 OPTIONS
From: "phone1" <sip:phone1@nas.weily.lan>;tag=486596409
To: <sip:0123465789@nas.weily.lan>
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bK247566f945c8d4f4475e75d2b55ae09d373033;rport
Max-Forwards: 70
Contact: "phone1" <sip:phone1@192.168.0.233:54974;transport=udp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Looking for 0123465789 in unauthenticated (domain nas.weily.lan)
<--- Transmitting (NAT) to 192.168.0.233:54974 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.233:54974;branch=z9hG4bK247566f945c8d4f4475e75d2b55ae09d373033;received=192.168.0.233;rport=54974
From: "phone1" <sip:phone1@nas.weily.lan>;tag=486596409
To: <sip:0123465789@nas.weily.lan>;tag=as7ef73b98
Call-ID: 7f600b6f5dd49f3191485b2221770e6e@192.168.0.233
CSeq: 3370 OPTIONS
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7f600b6f5dd49f3191485b2221770e6e@192.168.0.233' in 32000 ms (Method: OPTIONS)
<--- SIP read from UDP:192.168.0.107:5060 --->
BYE sip:0123465789@192.168.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107;branch=z9hG4bKee9b7f770c9b3cea2
Max-Forwards: 70
From: <sip:phone2@192.168.0.107:5060;transport=udp>;tag=2512611503
To: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 11673 BYE
User-Agent: Aastra 55i/3.2.2.3077
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.0.107:5060 (NAT)
Scheduling destruction of SIP dialog '76397d3a4927da5c7a9420376b208397@192.168.0.3:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 192.168.0.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.107;branch=z9hG4bKee9b7f770c9b3cea2;received=192.168.0.107;rport=5060
From: <sip:phone2@192.168.0.107:5060;transport=udp>;tag=2512611503
To: "Phone 1" <sip:0123465789@192.168.0.3>;tag=as1eb64ae6
Call-ID: 76397d3a4927da5c7a9420376b208397@192.168.0.3:5060
CSeq: 11673 BYE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (external, 0123465789, 1) exited non-zero on 'SIP/0123465789-00000020'
Scheduling destruction of SIP dialog '1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:212.117.203.47;lr> for address/port to send to
set_destination: set destination to 212.117.203.47:5060
Reliably Transmitting (NAT) to 212.117.203.47:5060:
BYE sip:212.117.203.47:5061 SIP/2.0
Via: SIP/2.0/UDP 80.219.136.161:5060;branch=z9hG4bK636b87d4;rport
Route: <sip:212.117.203.47;lr>
Max-Forwards: 70
From: <sip:0123465789@212.117.203.47>;tag=as3a9cd917
To: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:212.117.203.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK636b87d4;rport=5060
To: "Phone 1"<sip:0123465789@212.117.203.47>;tag=pe7dwc57mtiy3xyo.o
From: <sip:0123465789@212.117.203.47>;tag=as3a9cd917
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o
CSeq: 102 BYE
Server: Sippy
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060~2o' Method: ACK
<--- SIP read from UDP:212.117.203.47:5060 --->
BYE sip:0123465789@192.168.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-f6f7ec3310257d69-1---d8754z-;rport
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-xlewl3mpp7bncjxt;rport=5061
Max-Forwards: 69
Contact: "Anonymous"<sip:212.117.203.47:5061>
To: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
From: <sip:0123465789@guest2.voipgateway.org>;tag=oyefjjxy5eqerzze.i
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 428 BYE
User-Agent: Sippy
h323-conf-id: 692572566-3680965090-2692252715-735722158
cisco-GUID: 692572566-3680965090-2692252715-735722158
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 212.117.203.47:5060 (NAT)
Scheduling destruction of SIP dialog '1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 212.117.203.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.117.203.47:5060;branch=z9hG4bK-d8754z-f6f7ec3310257d69-1---d8754z-;received=212.117.203.47;rport=5060
Via: SIP/2.0/UDP 212.117.203.47:5061;branch=z9hG4bK-xlewl3mpp7bncjxt;rport=5061
From: <sip:0123465789@guest2.voipgateway.org>;tag=oyefjjxy5eqerzze.i
To: "Phone 1"<sip:0123465789@192.168.0.3:5060>;tag=as74b9a0bd
Call-ID: 1158ce590452a15e0a5291aa425f4658@80.219.136.161:5060
CSeq: 428 BYE
Server: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (internal, 0123465789, 2) exited non-zero on 'SIP/phone1-0000001e'
Scheduling destruction of SIP dialog '9f42d2a53406e625509057427a28d5aa@192.168.0.233' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:phone1@192.168.0.233:54974;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.233:54974
Reliably Transmitting (NAT) to 192.168.0.233:54974:
BYE sip:phone1@192.168.0.233:54974;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK074f501c;rport
Max-Forwards: 70
From: <sip:0123465789@nas.weily.lan>;tag=as30a62f26
To: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:192.168.0.233:54974 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK074f501c;rport=5060;received=192.168.0.3
From: <sip:0123465789@nas.weily.lan>;tag=as30a62f26
To: "phone1" <sip:phone1@nas.weily.lan>;tag=1104846907
Call-ID: 9f42d2a53406e625509057427a28d5aa@192.168.0.233
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '9f42d2a53406e625509057427a28d5aa@192.168.0.233' Method: ACK