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Losing Numbers


Megrain
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Hi All,

I have a funny one on one of our system. we are losing the first 3 numbers of the dialed number if the second set starts with a 0. for example. 011 021 3280 the SNOM ONE passes only the 021 3280 on the our Gateway. we don't use any custom dial plans and most of our setting are standard. the dial plan is basically only a * to pass the numbers to the Gateway.

 

If anyone had this in the past of can point me in the right direction to find a solution I would be great full.

[5] 2013/08/08 09:44:55:	SIP Rx tls:192.168.0.92:3265:
INVITE sip:0110213280@192.168.0.12;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.0.92:3265;branch=z9hG4bK-9kjezcrjpcjy;rport
From: "Jannie" <sip:8024@192.168.0.12>;tag=gq2pt4hthz
To: <sip:0110213280@192.168.0.12;user=phone>
Call-ID: 764c0352d891-ww03fy369nl2
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:8024@192.168.0.92:3265;transport=tls;line=xn80aqez>;reg-id=1
X-Serialnumber: 000413416DD8
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom870/8.7.3.7
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 515

v=0
o=root 1947995709 1947995709 IN IP4 192.168.0.92
s=call
c=IN IP4 192.168.0.92
t=0 0
m=audio 63128 RTP/AVP 9 0 8 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rEJtTq3Y6NHG7HfgzUogCPyFRRhNPmeckP7qEnrE
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[5] 2013/08/08 09:44:55:	SIP Tx tls:192.168.0.92:3265:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.0.92:3265;branch=z9hG4bK-9kjezcrjpcjy;rport=3265
From: "Jannie Smit" <sip:8024@192.168.0.12>;tag=gq2pt4hthz
To: <sip:0110213280@192.168.0.12;user=phone>;tag=74feb0062d
Call-ID: 764c0352d891-ww03fy369nl2
CSeq: 1 INVITE
Content-Length: 0
[5] 2013/08/08 09:44:55:	SIP Tx udp:192.168.0.19:5060:
INVITE sip:0213280@192.168.0.19;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-b2748a9160d2521dedb95c35443044a3;rport
From: "ECN" <sip:play@192.168.0.19>;tag=350641965
To: <sip:0213280@192.168.0.19>
Call-ID: 0dae4b18@pbx
CSeq: 1172 INVITE
Max-Forwards: 70
Contact: <sip:play@192.168.0.12:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids
Remote-Party-ID: "Jannie Smit" <sip:8024@192.168.0.12;user=phone>
P-Charging-Vector: icid-value=;icid-generated-at=192.168.0.12;orig-ioi=192.168.0.12
Content-Type: application/sdp
Content-Length: 382

v=0
o=- 315663417 315663417 IN IP4 192.168.0.12
s=-
c=IN IP4 192.168.0.12
t=0 0
m=audio 25804 RTP/AVP 18 0 8 9 2 3 101
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
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] 2013/08/08 09:44:55:	SIP Rx udp:192.168.0.19:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-b2748a9160d2521dedb95c35443044a3;rport=5060;received=192.168.0.12
From: "ECN" <sip:play@192.168.0.19>;tag=350641965
To: <sip:0213280@192.168.0.19>;tag=2649933857
Call-ID: 0dae4b18@pbx
CSeq: 1172 INVITE
Server: Patton SN4960 4E30VRD 00A0BA073946 R5.7 2011-05-13 H323 RBS SIP M5T SIP Stack/4.0.30.30
WWW-Authenticate: Digest realm="192.168.0.19",nonce="ea616592b52344cb50fae25b7569d1b000d2395aC0A4AD2CFE12",algorithm=MD5
Content-Length: 0
[5] 2013/08/08 09:44:55:	SIP Tx udp:192.168.0.19:5060:
INVITE sip:0213280@192.168.0.19;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-012623dcd8561f5050acea3a757ea2d2;rport
From: "ECN" <sip:play@192.168.0.19>;tag=350641965
To: <sip:0213280@192.168.0.19>
Call-ID: 0dae4b18@pbx
CSeq: 1173 INVITE
Max-Forwards: 70
Contact: <sip:play@192.168.0.12:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids
Remote-Party-ID: "Jannie Smit" <sip:8024@192.168.0.12;user=phone>
P-Charging-Vector: icid-value=;icid-generated-at=192.168.0.12;orig-ioi=192.168.0.12
Authorization: Digest realm="192.168.0.19",nonce="ea616592b52344cb50fae25b7569d1b000d2395aC0A4AD2CFE12",response="ef9489e0caf9af0341749eb5fc373eae",username="play",uri="sip:0213280@192.168.0.19;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 382

v=0
o=- 315663417 315663417 IN IP4 192.168.0.12
s=-
c=IN IP4 192.168.0.12
t=0 0
m=audio 25804 RTP/AVP 18 0 8 9 2 3 101
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
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] 2013/08/08 09:44:55:	SIP Tx tls:192.168.0.92:3265:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.0.92:3265;branch=z9hG4bK-9kjezcrjpcjy;rport=3265
From: "Jannie Smit" <sip:8024@192.168.0.12>;tag=gq2pt4hthz
To: <sip:0110213280@192.168.0.12;user=phone>;tag=74feb0062d
Call-ID: 764c0352d891-ww03fy369nl2
CSeq: 1 INVITE
Contact: <sip:8024@192.168.0.12:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 459

v=0
o=- 1372248197 1372248197 IN IP4 192.168.0.12
s=-
c=IN IP4 192.168.0.12
t=0 0
m=audio 20430 RTP/AVP 18 0 8 9 99 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6+fAOsq0hZHyrfukvT+7444mUUJOq2PV4wmBstJk
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2013/08/08 09:44:55:	SIP Rx tls:192.168.0.92:3265:
PRACK sip:8024@192.168.0.12:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.0.92:3265;branch=z9hG4bK-immnxhlvh55n;rport
From: "Jannie Smit" <sip:8024@192.168.0.12>;tag=gq2pt4hthz
To: <sip:0110213280@192.168.0.12;user=phone>;tag=74feb0062d
Call-ID: 764c0352d891-ww03fy369nl2
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:8024@192.168.0.92:3265;transport=tls;line=xn80aqez>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0
[5] 2013/08/08 09:44:55:	SIP Tx tls:192.168.0.92:3265:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.0.92:3265;branch=z9hG4bK-immnxhlvh55n;rport=3265
From: "Jannie Smit" <sip:8024@192.168.0.12>;tag=gq2pt4hthz
To: <sip:0110213280@192.168.0.12;user=phone>;tag=74feb0062d
Call-ID: 764c0352d891-ww03fy369nl2
CSeq: 2 PRACK
Contact: <sip:8024@192.168.0.12:5061;transport=tls>
User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids
Content-Length: 0

Regards,

J

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The point here is probably the country code. This tells the PBX how to read number, but also how to represent numbers (canonical form). For example if you have set country code to 1 and dial 0110213280, the PBX will read that as "dial international, country code 0213" and that's where it probably gets a little hiccup (there is no country code starting with 0).

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For the area code, you have three possibilities:

  1. Just the area code like 2262, which means that numbers are represented locally (lets say in Germany=49) like +492262123
  2. Area code with a 0 might also work; just try sometimes this ends up with a different representation
  3. Area code set to "-" means that all numbers in the country start with the number directly, without any 0 in the front

In any way, turn the logging levels up then you see what the PBX feeds into the dial plan logic, so that you can see what you should match,

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