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Kristan

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Everything posted by Kristan

  1. Is there any plan for keeping the printed material up to date? I like to give the customer a copy of the user guide so they can do some of the stuff themselves, but pbxnsip moves pretty quick and already the current stuff is out of date. I'd rather not point them to the wiki, so I was wondering if there's any sort of release schedule for the documentation along with the software versions.
  2. Hi All, Bit of a feature request - at the moment failover is either all SIP error codes, or just 5xx codes. Could this be made this a little more flexible? Basically I'd like to be able to failover from a SIP trunk to say ISDN on certain errors, like 401, 404, 408, 480 etc. but not the rest of the 4xx errors. There's no point in my attempting to ring a number on the ISDN if it's just busy, but there is if there is a problem at the voip provider. I'm not sure how I'd like to see this implemented either, maybe initially just a list of codes in the pbx.xml, but I just thought I'd mention it
  3. Has something changed around the hunt/extension setup? Two sites I've done this week, both had similar symptoms. First one I setup hunt groups 600 and 880, and extensions 601, 602 etc. and 881, 882 etc. Dialing into 600 though, I got the "you have reached the mailbox of 600", same for 880. I delete the hunt group, and I found an extension 600 "underneath" the hunt group. Second system I created two hunt groups 711 and 702, and extensions 240-260. All the extensions were set to auto provision (using the inbuilt polycom 2.2 support ) but the first ones tried to register as 711 and 702, but with the passwords for 240 and 241. I deleted the hunt groups, re-provisioned the phones then re-created the hunt groups and all was well. Just seems something somewhere is treating hunt groups as extensions for some part of the logic.
  4. If only it didn't end up as horrible screeching noises after a few days I have to admit, I've not tried it for a few versions, is it fixed?
  5. Looks like there's something not quite right in the polycom_sip.xml, the line : P_300.26.frame.1.duration="0"/> looks like it's missing a <I at the start
  6. I wondered that too. The whole thing is very strange, those Mitels as monster systems, I'll take PBXnSIP anyday
  7. I just get a 404 when trying to download the windows version?
  8. Your wish is my command: [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: INVITE sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 Route: <sip:192.168.10.240:5060;transport=udp;lr> Max-Forwards: 70 Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE Supported: timer,replaces From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240> Call-ID: 281599984-57239804 CSeq: 1 INVITE Min-SE: 90 Session-Expires: 90;Refresher=uas Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> P-Asserted-Identity: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Length: 0 [7] 2007/12/19 10:31:34: UDP: Opening socket on port 61252 [7] 2007/12/19 10:31:34: UDP: Opening socket on port 61253 [5] 2007/12/19 10:31:34: Identify trunk (IP address/port and domain match) 2 [9] 2007/12/19 10:31:34: Resolve destination 69745: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69745: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69745: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Content-Length: 0 [5] 2007/12/19 10:31:34: Trunk Mitel Trunk sends call to 602 [7] 2007/12/19 10:31:34: Calling extension 602 [8] 2007/12/19 10:31:34: Play audio_en/mb_this_is_the_mailbox_of.wav audio_en/bi_6.wav audio_en/bi_0.wav audio_en/bi_2.wav audio_en/mb_leave_msg_after_tone.wav audio_moh/mb_beep.wav [9] 2007/12/19 10:31:34: Resolve destination 69746: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69746: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69746: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Contact: <sip:602@192.168.10.240:5060;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/2.1.2.2292 Content-Type: application/sdp Content-Length: 378 v=0 o=- 17142 17142 IN IP4 192.168.10.240 s=- c=IN IP4 192.168.10.240 t=0 0 m=audio 61252 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH 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=sendrecv [9] 2007/12/19 10:31:34: Resolve destination 69747: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69747: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69747: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Contact: <sip:602@192.168.10.240:5060;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/2.1.2.2292 Content-Type: application/sdp Content-Length: 378 v=0 o=- 17142 17142 IN IP4 192.168.10.240 s=- c=IN IP4 192.168.10.240 t=0 0 m=audio 61252 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH 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=sendrecv [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807 Max-Forwards: 70 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 ACK Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Type: application/sdp Content-Length: 185 v=0 o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3 s=- c=IN IP4 192.168.10.3 t=0 0 m=audio 20199 RTP/AVP 8 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199 [8] 2007/12/19 10:31:34: No codec available for sending There's an option on the Mitel SIP trunk to force SDP in the initial invite, but if I do that the Mitel comes back and says "Error"
  9. Hi All, We have a customer with a Mitel 3300ICP who would to add a remote office with PBXnSIP. We've managed to get some of the mitel setup done (my god it makes you appreciate how simple PBXnSIP is to get running!) but are falling down with the audio. If I make a call from the mitel to an unregistered extension on the PBX (to get voicemail), after the initial invite/trying, the PBX sends the 200 OK and gets this back in response : [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807 Max-Forwards: 70 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 ACK Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Type: application/sdp Content-Length: 185 v=0 o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3 s=- c=IN IP4 192.168.10.3 t=0 0 m=audio 20199 RTP/AVP 8 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199 [8] 2007/12/19 10:31:34: No codec available for sending Mitel is on 192.168.10.2, PBX is on 192.168.10.240 - I've no idea what 192.168.10.3 is, I presume it's the IP of the phone on the mitel side making a call. Now PBXnSIP offers all it's supported codecs, but the mitel isn't negotiating any. If I do the same with one of our asterisk boxes, it looks like it just sends PCMU regardless of what the mitel says (which seems to work). Any ideas???
  10. Thanks, I'll have to do a bit more testing - as I said it was only a small issue and there were bigger things to worry about
  11. The hunt group is setup on a DDI with 310, with the group name "freephone number". If I set the PBXnSIP hunt group to show the groupname and incoming number, on snom/polycoms etc. it'll come up and say "freephone number (0123456789)". On the 942's, it's just giving the number, however the invite looked correct. I have to admit, I didn't spent a lot of time checking as it was only a minor annoyance.
  12. Hi all, Just installed a system with SPA-942's - simple setup with 4 phones and a single hunt group to ring them all. Set the hunt group to show the group name (number), however it only ever shows the number. I've checked the invite and it looks correct, it appears the linksys is just ignoring everything except for the number. Any idea how to change this? I assume it's a setting on the phone but I can't see anything obvious. Also, any ideas if there's a firmware that suppots the reason header? Thanks, Kristan
  13. I've got a system on a dual processor box running 2292, however looking in the settings no affinity mask is set. Does this mean it will be using both processors, or will it default to a single (the first?) one.
  14. Below is a mail we got from our snom supplier - just thought I would share it in case you guys haven't seen it Start of forwarded message: IMPORTANT TECHNICAL INFORMATION This FAQ concerns: • Firmware V7.1.27 / 7.1.28 • snom3x0 Problem: Telephones running the above mentioned FW versions may, in rare cases, exhibit an error which will then render the phone inoperable. Symptom: The phone reboots after a firmware update, but the process is not complete and the phone will hang as a result. Cause: The phone reboots automatically and prematurely during the Erasing / Writing Flash process which is triggered by a firmware update. Prevention: 1. Before the next firmware update, the Watchdog setting must be changed to OFF: 1. Web-Interface Main Menu: Select Advanced, Behavior, scroll down to Watchdog, select OFF. Scroll down and click on Save. 2. Confirm the: Apply Settings changes? Alert by clicking on Reboot, and confirming with Yes. 2. Now unplug the phone from the power source – either Power Adaptor or PoE. 3. Firmware Updates may now be performed as per normal. How to recover a telephone that has been rendered inoperable by the above issue: • A manual TFTP update must be performed.
  15. That's really strange - we've got a 50 user polycom system and not had a transfer to vm since running 2292 (fingers crossed this seems to be running smoothly). We also run it here on our office one and we've not seen it happen since upgrading to 2292. Are you using the out of the box provisioning for the polycoms and the 2.2 sip application on them? I found this didn't work too well and I needed to mangle the config files from the 2.2 firmware and the pbxnsip ones together. Drop me a PM if you want a copy of my configs.
  16. Really? We're running on 2.1.2.2292 and haven't had this at all. And we definitely would have heard about it.
  17. Oh thanks, I've never noticed that before! Does PBXnSIP use old skool ToS instead of DiffServ then?
  18. Is there any way to get the PBX to set the DiffServ bits in the IP header to something that will give traffic priority on switches that honour them? ta
  19. Question about recordings - What happens if a call comes in via a hunt group and gets picked up via pickup codes? Does the recording mechanism still apply or not? I've got a customer who needs everything in and out recording, and I can see calls in the logs which should have been recorded but haven't been. Thanks, Kristan
  20. An AJAX style operator interface would be really nice, showing current calls and the status of extensions (DND especially). A recent call history would be nice too, with AJAX quick filters to search for calls by specific extensions, inbound numbers etc.
  21. This prevents attended transfers though, and I'm assuming 3 way conferences on the phone also... I guess I'm after an equivalent of the snoms "call waiting" feature I can turn off - or any alternatives anyone has?
  22. Kristan

    CDR

    I've actually written a little (quite hacky) vb script to take the CDR logs and put them into an MS-SQL (MSDE) database, so you can pull them out via queries. I then take these and put them into an email and send them to the domain administrator. I'd be happy to share it (but there's no support )
  23. Have a look in the call log section of your PBX - does it have a caller ID in there? You may need to get your telco to enable it, or make sure your gateway is setup correctly to pass it through. What sort of gateway are you using?
  24. Hi All, On the polycoms the default behaviour is to allow a second call to come into the phone, while the first one is proceeding. The user can then reject, ignore or take the call. Does anyone know if it's possible for the call either not to get sent to the phone, or for the phone to reject with a 486 busy message? I know on the snom's there's a setting for this, but I can't find anything in the polycom reference guide. Thanks, Kristan
  25. It's also worth noting I had to change the registration timeout on the polycoms in the config to a very low 30 seconds - left at the defaults the registrations would expire on the PBX and the calls would go to VM (the phone thought it was still registered). Then it would re-register and the phone would start taking calls again. Not sure what the underlying issue is, but that's how I solved it. I don't think we had any other polycom related issues with the install, and it's running fine now, aside from one or two issues.
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