SIP Header modification..! + Incoming dial plan!

Asterisk Asterisk Support
1

Dear All,

Athought I’ve made a great research I couldn’t find any helpful infomation regarding how to modify the sip header. The problem I am facing is the following: Although I can register to the sip server whenever I am placing a call, I get the following sip header:

INVITE sip:@ SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2350684e;rport
Max-Forwards: 70
From: “1000” <sip:@[size=150]@192.168.10.2[/size]>;tag=as5b167c1b
To: sip:telconumber@
Contact: sip:telconumber@[size=150]@192.168.10.2[/size]>
Call-ID: 62234e8b778742a36a3f597b05c8ffed@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Tue, 07 Jun 2011 16:03:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

What I really need is to supress @192.168.10.2 (which is the internal LAN address of the asterisk) in order to pass the call over the SIP provider. Any suggestions is appreciated.

Additionally, I can not receive calls and I am getting the 403 code.

Thank you for our support and for reading. :mrgreen:

2

You have NAT misonfigured. In particular, look at the externhost and externip settings in sip.conf, but also make sure that localnet is correct.

You may have other NAT configuration issues.

3

david55,

Thank you for your reply. Unfortunately I didn’t make any progress…! I tried many different configurations and had no luck. I also visited the following url:http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions, and no progress was made. All I want really is to suppress the … :smiling_imp: … local address of the Astersik and pass the call through the trunk. Any suggestion is much appreciated.

Regards,

4

Make sure the trunk has NAT on.

Make sure localnets doesn’t include the trunk.

Make sure that you have an externhost or externip configured.

5

I finally managed to find a solution with the registration! Now the problem I am facing is that I can dial a number get the rigning tone and when I pick up the called device after 1 second the call is being dropped. Additionally, I can not gt incoming calls!!! Any help ip more than welcome. By the way is there any useful dialplan explanation guide. They all sound Greek to me!!! :smiley:

Thank you all for your support.

Regards,
Asteriskos

6

Here’s the sip trace:

<— SIP read from UDP:192.168.1.3:22732 —>
INVITE sip:2XXXXXXXXX@192.168.1.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2df904211e72414e-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1000@192.168.1.3:22732
To: "2XXXXXXXXXX"sip:2XXXXXXXXXXX@192.168.1.4:5060
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 402

v=0
o=- 12953134942167857 1 IN IP4 192.168.1.3
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.3
t=0 0
a=ice-ufrag:7c8100
a=ice-pwd:bbe9e3bb6729c33788edc04630df8078
m=audio 50994 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 50994 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 50995 typ host

<------------->
— (13 headers 14 lines) —
== Using SIP RTP CoS mark 5
Sending to 192.168.1.3 : 22732 (no NAT)
Using INVITE request as basis request - YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
Found peer ‘1000’ for ‘1000’ from 192.168.1.3:22732
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 192.168.1.3:50994
Looking for 2XXXXXXXXXXX in phones (domain 192.168.1.4)
list_route: hop: sip:1000@192.168.1.3:22732

<— Transmitting (no NAT) to 192.168.1.3:22732 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2df904211e72414e-1—d8754z-;received=192.168.1.3;rport=22732
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
To: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:XXXXXXXX@192.168.1.4
Content-Length: 0

<------------>
– Executing [XXXXXXXX@phones:1] NoOp(“SIP/1000-0000001e”, “”) in new stack
– Executing [XXXXXXXX@phones:2] Dial(“SIP/1000-0000001e”, “SIP/XXXX/XXXXXXXX”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 192.168.1.4 port 17110
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to XXXX.XXX.XX.XXX:5060:
INVITE sip:XXXXXXXX@fms.XXXXXXcom.gr SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK10b5b555;rport
Max-Forwards: 70
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
To: sip:XXXXXXXX@fms.XXXXXXcom.gr
Contact: sip:0030XXXXXXXX@192.168.1.4
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Tue, 21 Jun 2011 17:01:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 176737716 176737716 IN IP4 192.168.1.4
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.4
t=0 0
m=audio 17110 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called XXXX/XXXXXXXX

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK10b5b555
To: sip:XXXXXXXX@fms.XXXXXXcom.gr
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK10b5b555
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 INVITE
Contact: sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp
Record-Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Content-Type: application/sdp
Content-Length: 237
Session: Media
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE

v=0
o=BroadWorks 62275919 1 IN IP4 XXXX.XXX.XX.XXX
s=-
c=IN IP4 10.224.64.8
t=0 0
m=audio 50774 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8
a=cpar: a=rtpmap:8 PCMA/8000
a=cdsc: 2 image udptl t38
a=sendrecv

<------------->
— (12 headers 12 lines) —
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.224.64.8:50774
– SIP/XXXX-0000001f is making progress passing it to SIP/1000-0000001e
Audio is at 192.168.1.4 port 15210
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 192.168.1.3:22732 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2df904211e72414e-1—d8754z-;received=192.168.1.3;rport=22732
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
To: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:XXXXXXXX@192.168.1.4
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1223762041 1223762041 IN IP4 192.168.1.4
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.4
t=0 0
m=audio 15210 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=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK10b5b555
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 INVITE
Contact: sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp
Record-Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Content-Type: application/sdp
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE

v=0
o=BroadWorks 62275919 1 IN IP4 XXXX.XXX.XX.XXX
s=-
c=IN IP4 10.224.64.8
t=0 0
m=audio 50774 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8
a=cpar: a=rtpmap:8 PCMA/8000
a=cdsc: 2 image udptl t38
a=sendrecv

<------------->
— (11 headers 12 lines) —
– SIP/XXXX-0000001f is ringing

<— Transmitting (no NAT) to 192.168.1.3:22732 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2df904211e72414e-1—d8754z-;received=192.168.1.3;rport=22732
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
To: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:XXXXXXXX@192.168.1.4
Content-Length: 0

<------------>
– SIP/XXXX-0000001f is making progress passing it to SIP/1000-0000001e

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK10b5b555
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 INVITE
Contact: sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp
Record-Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY
Accept: multipart/mixed
Accept: application/media_control+xml
Accept: application/sdp

v=0
o=BroadWorks 62275919 1 IN IP4 XXXX.XXX.XX.XXX
s=-
c=IN IP4 10.224.64.8
t=0 0
m=audio 50774 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8
a=cpar: a=rtpmap:8 PCMA/8000
a=cdsc: 2 image udptl t38
a=sendrecv

<------------->
— (16 headers 12 lines) —
list_route: hop: sip:XXXX.XXX.XX.XXX;transport=udp;lr
set_destination: Parsing sip:XXXX.XXX.XX.XXX;transport=udp;lr for address/port to send to
set_destination: set destination to XXXX.XXX.XX.XXX, port 5060
Transmitting (no NAT) to XXXX.XXX.XX.XXX:5060:
ACK sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK4dea3f6b;rport
Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Max-Forwards: 70
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
Contact: sip:0030XXXXXXXX@192.168.1.4
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

-- SIP/XXXX-0000001f answered SIP/1000-0000001e

Audio is at 192.168.1.4 port 15210
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 192.168.1.3:22732 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2df904211e72414e-1—d8754z-;received=192.168.1.3;rport=22732
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
To: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:XXXXXXXX@192.168.1.4
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1223762041 1223762042 IN IP4 192.168.1.4
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.4
t=0 0
m=audio 15210 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=ptime:20
a=sendrecv

<------------>
– Native bridging SIP/1000-0000001e and SIP/XXXX-0000001f
set_destination: Parsing sip:XXXX.XXX.XX.XXX;transport=udp;lr for address/port to send to
set_destination: set destination to XXXX.XXX.XX.XXX, port 5060
Audio is at 192.168.1.4 port 17110
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to XXXX.XXX.XX.XXX:5060:
INVITE sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7e5798ad;rport
Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Max-Forwards: 70
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
Contact: sip:0030XXXXXXXX@192.168.1.4
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Require: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 177

v=0
o=root 176737716 176737717 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.3
t=0 0
m=audio 50994 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.3:22732 —>
ACK sip:XXXXXXXX@192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:22732;branch=z9hG4bK-d8754z-2c6b62cc40535d52-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1000@192.168.1.3:22732
To: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
From: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 1 ACK
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0

<------------->
— (10 headers 0 lines) —
set_destination: Parsing sip:1000@192.168.1.3:22732 for address/port to send to
set_destination: set destination to 192.168.1.3, port 22732
Audio is at 192.168.1.4 port 15210
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 192.168.1.3:22732:
INVITE sip:1000@192.168.1.3:22732 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK02a8e6d4;rport
Max-Forwards: 70
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Contact: sip:XXXXXXXX@192.168.1.4
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1223762041 1223762043 IN IP4 10.224.64.8
s=Asterisk PBX 1.6.2.11
c=IN IP4 10.224.64.8
t=0 0
m=audio 50774 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=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.3:22732 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK02a8e6d4;rport=5060
Contact: sip:1000@192.168.1.3:22732
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 373

v=0
o=- 12953134942167857 2 IN IP4 192.168.1.3
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.3
t=0 0
a=ice-ufrag:7c8100
a=ice-pwd:bbe9e3bb6729c33788edc04630df8078
m=audio 50994 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 50994 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 50995 typ host

<------------->
— (12 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 192.168.1.3:50994
set_destination: Parsing sip:1000@192.168.1.3:22732 for address/port to send to
set_destination: set destination to 192.168.1.3, port 22732
Transmitting (no NAT) to 192.168.1.3:22732:
ACK sip:1000@192.168.1.3:22732 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK3952bb5a;rport
Max-Forwards: 70
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Contact: sip:XXXXXXXX@192.168.1.4
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK7e5798ad
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4;rport=5061;branch=z9hG4bK7e5798ad
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 103 INVITE
Contact: sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY
Accept: multipart/mixed
Accept: application/media_control+xml
Accept: application/sdp

v=0
o=BroadWorks 62275919 2 IN IP4 XXXX.XXX.XX.XXX
s=-
c=IN IP4 10.224.64.8
t=0 0
m=audio 50774 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8
a=cpar: a=rtpmap:8 PCMA/8000
a=cdsc: 2 image udptl t38
a=sendrecv

<------------->
— (15 headers 12 lines) —
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.224.64.8:50774
set_destination: Parsing sip:XXXX.XXX.XX.XXX;transport=udp;lr for address/port to send to
set_destination: set destination to XXXX.XXX.XX.XXX, port 5060
Transmitting (no NAT) to XXXX.XXX.XX.XXX:5060:
ACK sip:sgc_c@XXXX.XXX.XX.XXX;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK0645f456;rport
Route: sip:XXXX.XXX.XX.XXX;transport=udp;lr
Max-Forwards: 70
From: “1000” sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
To: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
Contact: sip:0030XXXXXXXX@192.168.1.4
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

<— SIP read from UDP:XXXX.XXX.XX.XXX:5060 —>
BYE sip:0030XXXXXXXX@192.168.1.4:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP XXXX.XXX.XX.XXX:5060;branch=z9hG4bKp2mmfe6qpyidovndtx6kc2s7d
To: sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
From: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 1 BYE
Reason: SIP ;cause=503;text="Service Unavailable (10:211)"
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to XXXX.XXX.XX.XXX : 5060 (no NAT)

<— Transmitting (no NAT) to XXXX.XXX.XX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXXX.XXX.XX.XXX:5060;branch=z9hG4bKp2mmfe6qpyidovndtx6kc2s7d;received=XXXX.XXX.XX.XXX
From: sip:XXXXXXXX@fms.XXXXXXcom.gr;tag=h7g4Esbg_365288422-1308661343643
To: sip:0030XXXXXXXX@fms.XXXXXXcom.gr;tag=as6c9ac3fe
Call-ID: 4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr
CSeq: 1 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
set_destination: Parsing sip:1000@192.168.1.3:22732 for address/port to send to
set_destination: set destination to 192.168.1.3, port 22732
Audio is at 192.168.1.4 port 15210
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 192.168.1.3:22732:
INVITE sip:1000@192.168.1.3:22732 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7db67d7d;rport
Max-Forwards: 70
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Contact: sip:XXXXXXXX@192.168.1.4
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1223762041 1223762044 IN IP4 192.168.1.4
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.4
t=0 0
m=audio 15210 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=ptime:20
a=sendrecv

== Spawn extension (phones, XXXXXXXX, 2) exited non-zero on 'SIP/1000-0000001e’
Scheduling destruction of SIP dialog ‘YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.’ in 32000 ms (Method: ACK)

<— SIP read from UDP:192.168.1.3:22732 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7db67d7d;rport=5060
Contact: sip:1000@192.168.1.3:22732
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 373

v=0
o=- 12953134942167857 3 IN IP4 192.168.1.3
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.3
t=0 0
a=ice-ufrag:7c8100
a=ice-pwd:bbe9e3bb6729c33788edc04630df8078
m=audio 50994 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 50994 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 50995 typ host

<------------->
— (12 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 192.168.1.3:50994
set_destination: Parsing sip:1000@192.168.1.3:22732 for address/port to send to
set_destination: set destination to 192.168.1.3, port 22732
Transmitting (no NAT) to 192.168.1.3:22732:
ACK sip:1000@192.168.1.3:22732 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK4af64dd9;rport
Max-Forwards: 70
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Contact: sip:XXXXXXXX@192.168.1.4
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

set_destination: Parsing sip:1000@192.168.1.3:22732 for address/port to send to
set_destination: set destination to 192.168.1.3, port 22732
Reliably Transmitting (no NAT) to 192.168.1.3:22732:
BYE sip:1000@192.168.1.3:22732 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK657b208b;rport
Max-Forwards: 70
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 104 BYE
ser-Agent: Asterisk PBX 1.6.2.11
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Scheduling destruction of SIP dialog ‘YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.’ in 32000 ms (Method: ACK)
Really destroying SIP dialog '4cedeb890b8bbd474666301c4cccb290@fms.XXXXXXcom.gr’ Method: BYE

<— SIP read from UDP:192.168.1.3:22732 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK657b208b;rport=5060
Contact: sip:1000@192.168.1.3:22732
To: "1000"sip:1000@192.168.1.4:5060;tag=973b6b2f
From: "XXXXXXXX"sip:XXXXXXXX@192.168.1.4:5060;tag=as010b1c56
Call-ID: YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.
CSeq: 104 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘YWViMzJkY2FkOTM4MDk1ZmI3MWZiOGVhZThiMWMyMGU.’ Method: ACK

<— SIP read from UDP:192.168.1.3:22732 —>

<------------->

7

Typically means you are not permitted to do what you asked to do.

Looks like you may have tried to externally bridge to a private address (canreinvite/direct media, possibly interacting with a faulty NAT setting).

8

Thank you for your support david55,

Given that I am short of comfused will it help if I post the config I am using, to clean it up from excessive information ?

Regards,

9

Here’s the scenario I need to implement. I am starting a home working business and I will use X-lite softphones (in that case only one, when I will make progress I will add some extra softphones). In order to be able to do that I bought a sip service from a local provider which seems to me very reliable. I started a step by step configuration and here’s how far I went:
sip.conf

[general]
srvlookup =yes
registerattempts = 1

register => telconum@telcodomain:password@telcodomain~300

[1000]
type = friend
context = phones
nat=no
host = dynamic
disallow = all
allow = alaw

[Trunk_SIP]
type = friend
username = telcousername
host = telcodomain
fromuser = telcouser
secret = userpass
;context=incoming_calls
context = phones
;---------------MyADDONS-----
externhost = telecohostaddress
localnet = no
bindport = 5060
bindaddr = 0.0.0.0
canreinvite = nonat
qualify = no
nat = yes
outboundproxy = outbound proxy of telco
;-----------------------------
dtmfmode = rfc2833
disallow=all
allow = alaw
insecure = invite

extensions.conf

[general]
autofallthrough = yes

[default]
;exten => s,1,Verbose(1|Unrouted call handler)
;exten => s,n,Answer()
;exten => s,n,Wait(10)
;exten => s,n,Playback(tt-weasels)
;exten => s,n,Hangup()

[incoming_calls]
exten => _X.,1.NoOp()
exten => 1000,3,Dial(SIP/Trunk_SIP)

[outgoing_calls]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/Trunk_SIP/${EXTEN},30)

[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()

exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()

[phones]
include => internal
include => outgoing_calls
include => incoming_calls

So far, I was only capable of making partial outgoing calls, and listen to the music every time I was calling myself… I was unable to receive incoming calls. :exclamation: :exclamation:

Any valuable help will be much appreciated.

Thank you. :open_mouth: :open_mouth:

10

externhost = telecohostaddress
localnet = no

externhost should be a name and it should be yours, not the telco’s.

locanet should be a subnet pattern x.x.x.x/x

Not looked any further.