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Bintec R1200 as gateway for Snom ONE


Hubertus R.
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Hello,

 

I'm having troubles in getting a Bintec R1200 to run as a gateway for incoming calls. I successfully set up the connection using it as a SIP Registration. Also the Call Identification from P-Asserted-Identify works fine. Only the routing to the correct extension doesn't work. So far I can only set a fixed routing using the "Send call to extension" field.

 

Maybe someone can give me an advise, what I have to change to get this running. Here are the SIP messages, that might help. The called ISDN-MSN was the 25, which should be routed to the internal extension 25.

 

[5] 2012/10/03 14:33:12:	SIP Rx udp:192.168.151.250:5060:
INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038
To: "25" <sip:25@192.168.151.250>
Call-ID: 229AE3B3293926108DB4190009190038
CSeq: 1 INVITE
Contact: <sip:snomone@192.168.151.250:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: fec sip stack
Allow-Events: refer, message-summary, dialog
Content-Type: application/sdp
Content-Length: 183

v=0
o=- 23506 1 IN IP4 192.168.151.250
s=SIP call
c=IN IP4 192.168.151.250
t=0 0
m=audio 10010 RTP/AVP 8 18
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
[5] 2012/10/03 14:33:12:	SIP Tx udp:192.168.151.250:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038
To: "25" <sip:25@192.168.151.250>;tag=7a36527553
Call-ID: 229AE3B3293926108DB4190009190038
CSeq: 1 INVITE
Content-Length: 0

As can be seen, the system cannot find the target extension.

[5] 2012/10/03 14:33:12:	SIP Tx udp:192.168.151.250:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038
To: "25" <sip:25@192.168.151.250>;tag=7a36527553
Call-ID: 229AE3B3293926108DB4190009190038
CSeq: 1 INVITE
Contact: <sip:snomone@192.168.150.16:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Length: 0

[5] 2012/10/03 14:33:12:	SIP Rx udp:192.168.151.250:5060:
ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038
To: "25" <sip:25@192.168.151.250>;tag=7a36527553
Call-ID: 229AE3B3293926108DB4190009190038
CSeq: 1 ACK
Contact: <sip:snomone@192.168.151.250:5060;transport=udp>
Content-Length: 0

 

 

In case I set the "Send call to extension" field manually to 25, the successful invite to sip extension 25 looks like this:

[5] 2012/10/03 14:30:42:	SIP Tx tcp:192.168.222.26:58226:
INVITE sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x SIP/2.0
Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport
From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668
To: "Sip User 25" <sip:25@snom.mydomain>
Call-ID: abdee4b8@pbx
CSeq: 3565 INVITE
Max-Forwards: 70
Contact: <sip:25@192.168.150.16:5060;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Alert-Info: <http://127.0.0.1/Bellcore-dr3>
Content-Type: application/sdp
Content-Length: 256

v=0
o=- 50613 50613 IN IP4 192.168.150.16
s=-
c=IN IP4 192.168.150.16
t=0 0
m=audio 51504 RTP/AVP 8 3 101
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2012/10/03 14:30:42:	SIP Rx tcp:192.168.222.26:58226:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport=5060
From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668
To: "Sip User 25" <sip:25@snom.mydomain>;tag=zeczbc5izk
Call-ID: abdee4b8@pbx
CSeq: 3565 INVITE
Contact: <sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x>;reg-id=1
Content-Length: 0

 

So as long as I just want to have one MSN being sent to one extension, this is ok. But I have of course multiple ones I would like to use, without configuring a trunk for each.

 

Best

HR

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Thanks for the quick response, her the new log:

 

[5] 2012/10/03 16:29:07:	SIP Rx udp:192.168.151.250:5060:
INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038
To: "25" <sip:25@192.168.151.250>
Call-ID: EEB910E5393926108DC8190009190038
CSeq: 1 INVITE
Contact: <sip:snomone@192.168.151.250:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: fec sip stack
Allow-Events: refer, message-summary, dialog
Content-Type: application/sdp
Content-Length: 182

v=0
o=- 7872 1 IN IP4 192.168.151.250
s=SIP call
c=IN IP4 192.168.151.250
t=0 0
m=audio 10024 RTP/AVP 8 18
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
[8] 2012/10/03 16:29:07:	Allocating for call port 15, SIP call id EEB910E5393926108DC8190009190038
[5] 2012/10/03 16:29:07:	SIP Tx udp:192.168.151.250:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038
To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2
Call-ID: EEB910E5393926108DC8190009190038
CSeq: 1 INVITE
Content-Length: 0

[7] 2012/10/03 16:29:07:	Set packet length to 20
[6] 2012/10/03 16:29:07:	Call-leg 15: Sending RTP for EEB910E5393926108DC8190009190038 to 192.168.151.250:10024, codec not set yet
[8] 2012/10/03 16:29:07:	Incoming call: Request URI sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c, To is "25" <sip:25@192.168.151.250>
[8] 2012/10/03 16:29:07:	Trunk R1200-10@snom.mydomain has country code 49, area code 2581
[8] 2012/10/03 16:29:07:	call port 15: state code from 0 to 404
[7] 2012/10/03 16:29:07:	Set packet length to 20
[5] 2012/10/03 16:29:07:	SIP Tx udp:192.168.151.250:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038
To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2
Call-ID: EEB910E5393926108DC8190009190038
CSeq: 1 INVITE
Contact: <sip:snomone@192.168.150.16:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1090 Epsilon Geminids
Content-Length: 0

[5] 2012/10/03 16:29:07:	SIP Rx udp:192.168.151.250:5060:
ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0
Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport
From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038
To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2
Call-ID: EEB910E5393926108DC8190009190038
CSeq: 1 ACK
Contact: <sip:snomone@192.168.151.250:5060;transport=udp>
Content-Length: 0

[8] 2012/10/03 16:29:07:	Clearing call port 15, SIP call id EEB910E5393926108DC8190009190038
[8] 2012/10/03 16:29:07:	Hangup: Call 15 not found

 

 

I replaced my phone number by 0123123456. Everything else is untouched. I already tried to figure out using the manual, but so far I'm not lucky. Our other gateways didn't make any of those problems (berofix).

 

Thanks in advance

Hubertus

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