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Happy

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Posts posted by Happy

  1. If you have the "park orbit' account created and you have the button profile(with the park orbit button) assigned to the extension, the parking should work. (Just as a practice, I would set both park and retrieve as star code, say *85).

    Just reboot the phone if the buttons were not updated properly.

     

    There seems to be some issue with the moh on park orbit. We are looking into it.

     

    What OS are you using? (for the moh fix)

    We are on W32 (Xp).

    I did have problems with hold / Retrieve and Park, but after a complete restart of every extension everything seems normal. We have a normal moh when calls are parked in the park orbit. Hope it stays that way !

  2. Hmm... we will look into it.

    Seems to be another certificate is also not available :

     

    [4] 2010/11/20 05:50:52: Certificate for UTN-USERFirst-Hardware not available

    [4] 2010/11/20 06:50:52: Certificate for Pradeep not available

     

    Maybe this info is helpfull in any way...

     

    Marc

  3. We introduced a new type "Park Orbit". It has pretty much the same functionality like the previous implicit orbit; however you can pick up the call from an extension by just dialling the park orbit number. This one does not "waste" an extension license. And when you send the call from a trunk to the orbit, the PBX will park the call there.

    Ok, I found the object 'park orbit' and created one. But when I link a button using 'Park Orbit' and the number of the park orbit, it does not park. I am clearly missing something!

  4. Hi Matt,

     

    yes I got it in the meanwhile. Many thanks for your answer.

    By the way. Now I have a really fantastic PBX on my windows box. And it is excellent for my small home office and the family. I use a Fritzbox 7270 as the voip gateway. Sound is great and the PBX features as well.

     

    Thanks again,

    Andreas

    Interesting, Andreas : you actually managed to use the Fritzbox 7270 as a gateway between pots/isdn line and your snomone pbx ? I have a fritzbox 7170, but I could not make it a reliable gateway ...

  5. We introduced a new type "Park Orbit". It has pretty much the same functionality like the previous implicit orbit; however you can pick up the call from an extension by just dialling the park orbit number. This one does not "waste" an extension license. And when you send the call from a trunk to the orbit, the PBX will park the call there.

    So you have to setup a park orbit first, then use that number in the extension setup : how do you setup the orbit ? I checked the manual but it was not very helpfull to me regarding park orbits...

  6. Since my move towards Pbxnsip Version 4 (latest version), system is asking me for certificates ?

     

    *----------Logfile----

    [4] 2010/11/18 16:39:09: Certificate for Pradeep not available

    [4] 2010/11/18 16:39:10: Certificate for UTN-USERFirst-Hardware not available

     

    *---------------------

     

    Any explanation ?

     

    Thanks,

    Marc

  7. Can anyone explain the strange codex use and transcoding that seems to be happening since I started up Pbxnsip version 4 (latest version).

     

    *--------Logfile--

    Sending RTP for 3c2babd5d497-evfazu85tsm2 to 192.168.1.122:50922, codec not set yet

    Dialplan "Algemeen Kiesplan": Match 011999999@pbx.praktijk.be to <sip:011999999@ssw0.brussels.weepee.org;user=phone> on trunk WeePee

    Codec pcmu/8000 is chosen for call id 3c2babd5d497-evfazu85tsm2

    Codec gsm/8000 is chosen for call id 6cf614d1@pbx

    Sending RTP for 6cf614d1@pbx to 91.208.12.130:26410, codec gsm/8000

    Different Codecs (local pcmu/8000, remote gsm/8000), callid 3c2babd5d497-evfazu85tsm2, falling back to transcoding

    *--------

     

    Many thanks !

  8. Yes that could be a source for trouble. There are two settings that you can check in admin/ports: The packet size (set it to 30 ms) and the question if the PBX should always send the packet size (set it to yes).

    Change the admin/ports packet size in Pbxnsip would influence connections to all trunks and phones, I suppose.

    Found a phone : a sipura device with a non-standard 30millisec setting!

  9. If you are talking about "....codec not set yet", that's ok. It is not a problem. But if you are taking about the "...falling back to transcoding", then there is some phone or trunk provider doing 30ms packet instead of default 20ms. You need to find out who is doing 30ms and change it to do 20ms.

    I was concerned about both.

    If I understand you correcly, I have to ask the voip provider to change teh 30ms packet to 20ms. There is no way to adapt pbxnsip to the provider's settings ?

  10. Could anyone explain to me the strange codex behaviour, that seems to take place since I upgraded my system to the latest version 4 ?

     

    Log details below (don't find how to make attachements)

    ------------------

    [4] 2010/10/25 08:13:08: Certificate for UTN-USERFirst-Hardware not available

    [6] 2010/10/25 08:41:09: Sending RTP for 3c26f037f183-pmfuk042dnj0 to 192.168.1.111:60040, codec not set yet

    [6] 2010/10/25 08:41:09: Codec pcmu/8000 is chosen for call id 3c26f037f183-pmfuk042dnj0

    [6] 2010/10/25 08:45:37: Last message repeated 2 times

    [6] 2010/10/25 08:45:37: Sending RTP for 3c26f1443a60-eo3jprew4xti to 192.168.1.111:51430, codec not set yet

    [6] 2010/10/25 08:45:38: Codec pcmu/8000 is chosen for call id 3c26f1443a60-eo3jprew4xti

    [6] 2010/10/25 08:45:42: Last message repeated 2 times

    [6] 2010/10/25 08:45:42: Sending RTP for 3c26f1493088-l0rfsie8ywpl to 192.168.1.111:58142, codec not set yet

    [6] 2010/10/25 08:45:43: Codec pcmu/8000 is chosen for call id 3c26f1493088-l0rfsie8ywpl

    [6] 2010/10/25 09:01:52: Last message repeated 2 times

    [6] 2010/10/25 09:01:52: Sending RTP for 3c2794bb2bf4-n6h6xmpisry6 to 192.168.1.23:55170, codec not set yet

    [6] 2010/10/25 09:01:52: Codec pcmu/8000 is chosen for call id 3c2794bb2bf4-n6h6xmpisry6

    [6] 2010/10/25 09:02:14: Last message repeated 2 times

    [6] 2010/10/25 09:02:14: Sending RTP for 824d122fbf0cbdbd to 192.168.1.2:5238, codec not set yet

    [3] 2010/10/25 09:02:14: Account 33 is not a Service Flag

    [3] 2010/10/25 09:02:14: Hunt group 94 wants to add 6 members

    [6] 2010/10/25 09:02:26: Codec pcmu/8000 is chosen for call id 6712bbf7@pbx

    [6] 2010/10/25 09:02:26: Sending RTP for 6712bbf7@pbx to 192.168.1.21:55946, codec pcmu/8000

    [6] 2010/10/25 09:02:26: Codec pcmu/8000 is chosen for call id 824d122fbf0cbdbd

    [6] 2010/10/25 09:04:55: Sending RTP for 3d48f974f3b4f453 to 192.168.1.2:5240, codec not set yet

    [3] 2010/10/25 09:04:56: Account 33 is not a Service Flag

    [3] 2010/10/25 09:04:56: Hunt group 94 wants to add 6 members

    [6] 2010/10/25 09:05:10: Codec pcmu/8000 is chosen for call id ca1a80bc@pbx

    [6] 2010/10/25 09:05:10: Sending RTP for ca1a80bc@pbx to 192.168.1.21:49168, codec pcmu/8000

    [6] 2010/10/25 09:05:10: Codec pcmu/8000 is chosen for call id 3d48f974f3b4f453

    [6] 2010/10/25 09:06:24: Sending RTP for 05622ea9cd2c311c to 192.168.1.2:5242, codec not set yet

    [3] 2010/10/25 09:06:24: Account 33 is not a Service Flag

    [3] 2010/10/25 09:06:24: Hunt group 94 wants to add 6 members

    [6] 2010/10/25 09:06:36: Codec pcmu/8000 is chosen for call id 1646ddb2@pbx

    [6] 2010/10/25 09:06:36: Sending RTP for 1646ddb2@pbx to 192.168.1.23:57510, codec pcmu/8000

    [6] 2010/10/25 09:06:36: Codec pcmu/8000 is chosen for call id 05622ea9cd2c311c

    [6] 2010/10/25 09:07:53: Sending RTP for 52C428498BB2667E@192.168.1.1 to 192.168.1.1:7078, codec not set yet

    [3] 2010/10/25 09:07:53: Hunt group 21 wants to add 6 members

    [6] 2010/10/25 09:07:53: Codec pcmu/8000 is chosen for call id 52C428498BB2667E@192.168.1.1

    [6] 2010/10/25 09:07:57: Sending RTP for 3c279627ef8b-w9hqobmv4z5o to 192.168.1.23:51182, codec not set yet

    [6] 2010/10/25 09:07:57: Codec pcmu/8000 is chosen for call id 3c279627ef8b-w9hqobmv4z5o

    [6] 2010/10/25 09:08:59: Last message repeated 2 times

    [6] 2010/10/25 09:08:59: Sending RTP for 795D34D4A0751049@192.168.1.1 to 192.168.1.1:7078, codec not set yet

    [3] 2010/10/25 09:08:59: Hunt group 21 wants to add 6 members

    [6] 2010/10/25 09:08:59: Codec pcmu/8000 is chosen for call id 795D34D4A0751049@192.168.1.1

    [6] 2010/10/25 09:09:21: Sending RTP for 3c2e50241a19-cgdthhcerdpi to 192.168.1.22:64732, codec not set yet

    [6] 2010/10/25 09:09:21: Codec pcmu/8000 is chosen for call id 3c2e50241a19-cgdthhcerdpi

    [6] 2010/10/25 09:13:08: Last message repeated 2 times

    [4] 2010/10/25 09:13:08: HTTP client: Timeout on 173.166.77.221:443

    [4] 2010/10/25 09:13:08: Certificate for UTN-USERFirst-Hardware not available

    [6] 2010/10/25 09:14:47: Sending RTP for b53e7ab21fa8d6b0 to 192.168.1.2:5244, codec not set yet

    [3] 2010/10/25 09:14:47: Account 33 is not a Service Flag

    [3] 2010/10/25 09:14:47: Hunt group 94 wants to add 6 members

    [6] 2010/10/25 09:15:02: Codec pcmu/8000 is chosen for call id b53e7ab21fa8d6b0

    [3] 2010/10/25 09:15:11: Hunt group 95 wants to add 6 members

    [6] 2010/10/25 09:15:17: Codec pcma/8000 is chosen for call id e15efdaf@pbx

    [6] 2010/10/25 09:15:17: Sending RTP for e15efdaf@pbx to 192.168.1.26:49152, codec pcma/8000

    [6] 2010/10/25 09:15:18: Different packet size (30 and 20), callid b53e7ab21fa8d6b0, falling back to transcoding

    [6] 2010/10/25 09:15:18: Different packet size (20 and 30), callid e15efdaf@pbx, falling back to transcoding

    [6] 2010/10/25 09:17:11: Received bindRequest for user pbx.praktijk.be\13

    [6] 2010/10/25 09:17:18: Sending RTP for 3c279852a67d-jqmf892xc6fd to 192.168.1.23:57276, codec not set yet

     

    --------------------------------------

  11. I am trying to setup an ext to give only information to the user (currently using v 3.x).

     

    I do not want them to leave a vm, so I can't setup an ext and disable the vm. I tried an IVR but it will not timeout just plays over and over and over. (could use this if I can say hit zero and go back to the main menu or hang up) Auto Attendant asks the user to enter an option. I would use the AA and just timeout to the main menu but it still asks to enter your choice.

     

    Thanks in advance for any information.

     

    Michael,

     

    The following line in the IVR DTMF list will cause the IVR

  12. In order to avoid endless loops the PBX needs to count how many times a call has been redirected. The global variable (see pbx.xml) with the name "max_loop" does that. The default is 10, you probably need to set it to a higher value here. See https://www.pbxnsipsupport.com/index.php?_m...kbarticleid=413 on how to set a global parameter from the web interface.

    I have been testing that solution, setting max_loop to 30. It made not any difference.

    There does not seem to be much logic behind it, but when I start up far enough in the chain, I get to the end. Otherwise the line gets cut.

    Could a clean setup of Pbxnsip be a solution to my problem : 6 steps to get a call to its destination surely can't be a problem related to pbxnsip ?

  13. I would like to upload a ringtone.xml file to pbxnsip, having alert info to link Custom1 to Custom5 ringtones to internal Snom320 ringtones.

     

    Since Pbxnsip uses its own ringtone info to differentiate between internal and external calls, and I don't know where this info is stored, I am a bit concerned about the possibility that my ringtone.xml file will erase somehow this information.

     

    Can anyone tell me what happens with the ringtone.xml file once it is uploaded to the pbx ? :)

     

    Many thanks,

    Marc

  14. It looks like that the chain of hunt groups, that I am using without any problem in version 3, blocks in version 4.

     

    What I am using when call gets to the pbx :

    HG 92 -> HG94 -> IVR 81 -> HG95 -> IVR85 -> HG96

     

    The IVR's are just for announcements, no choices have to be made (Dtmf-list : !.!95! !E!95!).

     

    When I start dialing up the chain, this is what happens :

    81 -> ok all the way to 96

    94 -> ok all the way to 95 (call gets terminated thereafter)

    92 -> ok all the way to 81 (idem)

     

    I noticed that Stage 1 and 2 of the hunt groups seem to accept only extension numbers. Calls get terminated if HG or IVR are put in for example stage 2, you have to put these in the last stage. There is a good logic for that.

     

    But I don't see any reason / setting why my chain gets broken. Could someone verify this, please. Version 4 is not usable for me unless I can get this fixed.

  15. It would be nice to have virtual UNLICENCED extensions.. something with no SIP registration but can collect voicemails.

    I would like the same feature : you can solve some of the stuff by using a hunt group, but unregistered extensions - out of the licence - would be a great idea !!

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