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Problem with international numbers


CarlH
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Hi,

 

We are having problems with some international numbers that we have registered to forward calls to our helpdesk. E.g here are extracts from the log when calling to the Norwegian nr +4721031332 and to the Switzerland nr +41435000151. Both these nr are forwarded to 0842014000 which is the Swedish nr. When calling from the Norweigan nr i get connected but I don't hear anything. When I call to the Switz nr i hear the IVR loud and clear.

 

Any help would be greatly appreciated!

 

Here is a an extract from the log when calling to the Norwegian nr.

 

[7] 2008/12/02 09:24:03:SIP Rx udp:195.149.148.40:5060: BYE sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0

Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on>

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2

To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6

Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41

CSeq: 103 BYE

User-Agent: RTC Gateway 2.0

Max-Forwards: 70

Content-Length: 0

P-hint: call from pstn gateway

 

[9] 2008/12/02 09:24:03:Resolve 29794699: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: a udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: udp 195.149.148.40 5060 [7] 2008/12/02 09:24:03:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060

Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on>

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2

To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6

Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41

CSeq: 103 BYE

Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>

User-Agent: pbxnsip-PBX/3.0.1.3023

RTP-RxStat: Dur=9,Pkt=430,Oct=73960,Underun=0

RTP-TxStat: Dur=9,Pkt=441,Oct=75852

Content-Length: 0

 

Here is a an extract from the log when calling to the Switz nr.

 

[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: INVITE sip:0842014000@83.145.6.141:5060;transport=udp;line=02e74f10 SIP/2.0

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>

Contact: <sip:0046707960416@195.138.212.41>

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 INVITE

User-Agent: RTC Gateway 2.0

Max-Forwards: 70

Date: Tue, 02 Dec 2008 08:10:20 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 268

P-hint: call from pstn gateway

P-hint: local sip call

 

v=0

o=root 14929 14929 IN IP4 195.138.212.41

s=session

c=IN IP4 195.138.212.41

t=0 0

m=audio 15002 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=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

[9] 2008/12/02 09:10:59:UDP: Opening socket on port 50050 [9] 2008/12/02 09:10:59:UDP: Opening socket on port 50051 [5] 2008/12/02 09:10:59:Identify trunk (line match) 27 [9] 2008/12/02 09:10:59:Resolve 29788667: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 100 Trying

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 INVITE

Content-Length: 0

 

[7] 2008/12/02 09:10:59:Set packet length to 20 [6] 2008/12/02 09:10:59:Sending RTP for 033369853e67a81277f6e86855db9b4f@195.138.212.41#670aff6470 to 195.138.212.41:15002 [5] 2008/12/02 09:10:59:Trunk RTC 0842014000 sends call to 00 in domain smarthost.se [8] 2008/12/02 09:10:59:Play recordings/ivr79.wav [7] 2008/12/02 09:10:59:Set packet length to 20 [9] 2008/12/02 09:10:59:Resolve 29788668: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 INVITE

Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 228

 

v=0

o=- 46299 46299 IN IP4 83.145.6.141

s=-

c=IN IP4 83.145.6.141

t=0 0

m=audio 50050 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

[9] 2008/12/02 09:10:59:Resolve 29788669: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok

Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 INVITE

Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 228

 

v=0

o=- 46299 46299 IN IP4 83.145.6.141

s=-

c=IN IP4 83.145.6.141

t=0 0

m=audio 50050 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

[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

Via: SIP/2.0/UDP 195.149.148.40;branch=0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK5faf7a1a;rport=5060

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470

Contact: <sip:0046707960416@195.138.212.41>

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 ACK

User-Agent: RTC Gateway 2.0

Max-Forwards: 70

Content-Length: 0

P-hint: call from pstn gateway

 

[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0

Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>

Via: SIP/2.0/UDP 195.149.148.40;branch=0

Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK3cda2bc1;rport=5060

From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486

To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470

Contact: <sip:0046707960416@195.138.212.41>

Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41

CSeq: 102 ACK

User-Agent: RTC Gateway 2.0

Max-Forwards: 70

Content-Length: 0

P-hint: call from pstn gateway

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  • 1 month later...
Yes, that whole topic is addressed in 3.1.1. If you can, get a 3-minute demo key, set up a test server and try the latest &

greatest (http://pbxnsip.com/protect/pbxctrl-3.1.1.3100.exe). Then set your contry code to 47 (if I am right here) and then the numbers should be formatting correctly - automatically.

 

 

Hi!

 

I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK.

 

Any thoughts? All help is very appreciated.

 

Br

Carl

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I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK.

 

Any thoughts? All help is very appreciated.

 

Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG...

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Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG...

Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR.

 

Denmark works Finland doesn't.

 

DENMARK (45) NATIONAL 1 +4569918175

FINLAND (358) HELSINKI (9) 1 +358942419025

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Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR.

 

Denmark works Finland doesn't.

 

DENMARK (45) NATIONAL 1 +4569918175

FINLAND (358) HELSINKI (9) 1 +358942419025

 

Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio).

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