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CarlH

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About CarlH

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  1. H! We have a customer in Sweden who are setting up a office in the UK. They want to use the pbxnsip in the UK office as well. Today we have one trunk per account and the dial plan pattern is simply a * Now we need to get some trunks in the UK and make sure that when the users dial Sweden +46 they will use the Swedish trunk otherwise the UK trunk. How do I accomplish this? Regards Carl
  2. I finally got it working! I used both alias on each extension and "try loopback" in the dialplan. I can now call the alias I set in other domains.
  3. This didn't work for me. I made two accounts 43 0842013043 on domain01 and 30 0842016130 on domain02 I still cant call cross domains wit the alias. Do I need to cahnge anything in dialpans or trunks?
  4. We previously used tel:alias to call numbers in other domains but since upgrading to version 3 this doesn't work anymore. I read about the try loopback option as trunk but did not get it to work. What i did was that I disabled loopback detection and then added "Try loopback" to the dialplan with the lowest "pref" and * as "pattern". I'm guessing that's not all there is to it but since I can't find any more info about this I'm asking here
  5. Hi, I would like some more info on this. Is it possible for you to write some type of step by step guide on how to configure cross domain calling? I have read several topics about this but haven't seen any clear information on what to do. Is it possible that you could write some kind of step by step because I don't think I'm the only one with this request. Lets say we have two domains domain1.net and domain2.net in domain 1 we have nr 10-50 in domain 2 we have nr 10-50 as well In the dial plan we currently have pattern * for the trunk. Would it be possible to configure the dial plans so that you call domain2 by pressing eg #1 and then the nr eg #110 for nr 10 in domain2 One of the domains is localhost with domain01.net as alias name. Br Carl
  6. Edit: I'm posting this as a new topic under dialplans instead Hi, I would like som more info on this. Is it possible for you to write some type of step by step guide on how to configure inter domain calling? Lets say we have two domains domain1 and domain2. in domain 1 we have nr 10-50 in domain 2 we have nr 10-50 aswell In the dialplan we currently have pattern * for the trunk. Would it be possible to configure the dialplans so that you call domain2 by pressing eg #1 and then the nr eg #110 for nr 10 in domain2 Br Carl
  7. Which codecs do you recommend?
  8. 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
  9. 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
  10. Firmware-Version: snom-m3-SIP/01.16//03-Jul-08 13:43
  11. Hi, We recently bought a Snom M3 but we cant get DTMF to work with it. Any clue on what setting we should change? Br Carl
  12. CarlH

    3.1

    Yeah, I know and i'm fully aware of that it was an extremely stupid thing to do. I too have done this procedure many times and never had any issues and of course the one time i didn't make a copy this happened
  13. CarlH

    3.1

    I decided to try the latest version tonight after office hours in our live environment. I should NEVER have done that! All settings were deleted and the xml files were overwritten with 0 byte files. I have never experienced this after an upgrade before. Luckily I had a copy thats a few weeks old of the PBX directory. I lost some extensions but I'm working now to restore them.
  14. 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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