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Incoming calls problem on different numbers


snombtw
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Hi,

 

We are meeting a problem with Snom-Pbx and 4-5 Snom phones : All incoming calls arrive on the same phone, direct calls all arrive on the same phone.

 

We want to receive direct calls on the IP-phones, but all calls arrive always on the same phone.

 

Here is the log found on the SnomOnePBX server :

"

[9] 2011/02/03 15:10:29: SIP Rx udp:91.121.X.X:5060:

INVITE sip:003336672XXXX@192.168.X.X:5060;transport=udp;line=c81e728d SIP/2.0

Allow: UPDATE,REFER,INFO

Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net

Contact: <sip:91.121.X.X:5060>

Content-Type: application/sdp

CSeq: 791574727 INVITE

From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583

Max-Forwards: 30

To: <sip:0320xxxxxx@91.121.X.X;user=phone>

User-Agent: Cirpack/v4.42s (gw_sip)

Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8

Content-Length: 481

 

v=0

o=cp10 129674225667 129674225667 IN IP4 10.7.X.X

s=SIP Call

c=IN IP4 91.121.X.X

t=0 0

m=audio 33172 RTP/AVP 18 4 0 8 125 111 101

b=AS:21

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000/1

a=fmtp:4 annexa=no

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:125 CLEARMODE/8000/1

a=rtpmap:111 iLBC/8000/1

a=fmtp:111 mode=30

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

a=sqn:0

a=cdsc: 1 image udptl t38

[9] 2011/02/03 15:10:29: UDP: Opening socket on 0.0.0.0:60170

[9] 2011/02/03 15:10:29: UDP: Opening socket on 0.0.0.0:60171

[9] 2011/02/03 15:10:29: UDP: Opening socket on [::]:60170

[9] 2011/02/03 15:10:29: UDP: Opening socket on [::]:60171

[5] 2011/02/03 15:10:29: Identify trunk (line match) 2

[9] 2011/02/03 15:10:29: Resolve 667271: aaaa udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667271: a udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667271: udp 91.121.129.17 5060

[9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8

From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583

To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c

Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net

CSeq: 791574727 INVITE

Content-Length: 0

 

[7] 2011/02/03 15:10:29: Set packet length to 30

[6] 2011/02/03 15:10:29: Sending RTP for 10360-NL-2f3ab14a-2818812b3@sip.ovh.net to 91.121.X.X:33172, codec not set yet

[8] 2011/02/03 15:10:29: Call from an trunk 2

[8] 2011/02/03 15:10:29: Trunk OVH Sip@snompbx.mydomain.fr has country code 33, area code not set

[9] 2011/02/03 15:10:29: Incoming: formatted From is = "0678XXXXXX" <sip:+33678xxxxxx@sip.ovh.net;user=phone>

[9] 2011/02/03 15:10:29: Incoming: formatted To is = <sip:+3332xxxxx@91.121.X.X;user=phone>

[8] 2011/02/03 15:10:29: To is <sip:+33320xxxxxx@91.121.X.X;user=phone>, user 2, domain 1

[8] 2011/02/03 15:10:29: To user 41

[5] 2011/02/03 15:10:29: Domain trunk OVH Sip@snompbx.mydomain.fr could not identify user for +33366XXXXXX

[7] 2011/02/03 15:10:29: Set packet length to 30

[9] 2011/02/03 15:10:29: Resolve 667272: aaaa udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667272: a udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667272: udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8

From: "0678XXXXXX" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583

To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c

Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net

CSeq: 791574727 INVITE

Contact: <sip:0033366xxxxxx@192.168.X.X:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.2.0.3950

Content-Length: 0

 

[9] 2011/02/03 15:10:29: Resolve 667273: aaaa udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667273: a udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: Resolve 667273: udp 91.121.X.X 5060

[9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8

From: "0678xxxxxx" <sip:0678xxxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583

To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c

Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net

CSeq: 791574727 INVITE

Contact: <sip:0033366xxxxxx@192.168.X.X:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.2.0.3950

Content-Length: 0

 

[9] 2011/02/03 15:10:29: SIP Rx udp:91.121.X.X:5060:

ACK sip:0033366xxxxxx@192.168.X.X:5060;transport=udp;line=c81e728d SIP/2.0

Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net

Contact: <sip:91.121.1X.X:5060>

CSeq: 791574727 ACK

From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583

Max-Forwards: 30

To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c

User-Agent: Cirpack/v4.42s (gw_sip)

Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8

Content-Length: 0

 

[8] 2011/02/03 15:10:29: Hangup: Call 444 not found

"

 

Some help will be very useful,

 

THX

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Hi,

 

Thanks for the quick answer.

Yes i have the DID (with ou without the country code) configured after the internal extension (e.g 41 32047471)

When i set the "send call to extension 41" in the trunk parameters, calls are corectly routed on the end-user phone.

 

As far as i can understand the logs, the DID is recognized as the log gives the correct extension number

[8] 2011/02/03 15:10:29: To is <sip:+33320xxxxxx@91.121.X.X;user=phone>, user 2, domain 1

[8] 2011/02/03 15:10:29: To user 41

 

THis call was for user with extension 41.

It just looks like after, instead of ringing the extension 41, it tries to forward the call to the SIP trunk?

 

Thanks again

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Yes i have the DID (with ou without the country code) configured after the internal extension (e.g 41 32047471)

When i set the "send call to extension 41" in the trunk parameters, calls are corectly routed on the end-user phone.

 

The 32047471 is probably not correct. Either enter 032047471 or to make it 100 % clear +3332047471.

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