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cfcs

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  1. Maybe worth a try: set allow_pass_through to false. (e.g. http://192.168.1.2/reg_settings.htm?save=s..._through=false).

     

     

    You instructed me to folllow these directoin to disable pass-through mode.

    Try disabling the pass-through mode. This seems to be an ongoing pain in the neck, especially with the CS410 PSTN gateway.

     

    You can do this from the web interface (once that you are logged in as admin) with http://1.2.3.4/reg_status.htm?save=save&am...s_through=false.

     

    Which way is correct?

  2. Well, there is a new release 2.1.3 image that should avoid the problem.

     

    I'm having a similar problem only with Polycom phones:

     

    My VM has been recording conversations between the reception and when an SCC client calls. It appears that at times my VM does not pick up and the caller’s call rings back to the front desk. It is at this time that the entire conversation is recorded on my VM.

  3. I can't get the cs410 to boot up properly. I can use winscp to see what's on the device the log file is 80 megs so I think the system is full but when I try to delete the log the system locks up. need help I'm not a linux expert.

     

    I unplugged the power for 5 minutes then it came up on the next boot.

  4. I heared some time ago that receptionist with a lot of calls associate calls with colors. They mark a key with a color. I thought that was an interesting idea.

     

    A less stressing method is to use a ACD for the receptionist. Then he or she can process the calls one by one, and the PBX can even say the initial greeting "welcome to company xy, you are talking to abc".

     

     

    OK I found out more detail about his problem. I have a hunt group that rings extension 500 for 15 seconds then 508 for 15 seconds with a final destination of 700 the auto attendant. The problem is that when 500 and 508 are on the phone and a third call comes in something gets messed up and they are unable to transfer there current calls. Help Please.

  5. I'm looking for some tips on how to hand this situation. At times we have multiple calls comming in and the receptionist will put the first call on hold then pick up the second call. She then has a hard time getting back to the fist call and sometimes ends up dropping all calls on hold. With no caller id it's hard to identify the callers on hold. Suggestions will be appreciated.

  6. Has anyone had problems getting real time support from PBXnSIP. We have a large opportunity that we are trying to get going and we are having major issues with our PBX and getting features to work. Does anyone have an escalation list?

     

    It took pbxnsip a week before they gave me a repsonce to a request I had sent. Although the problem turned out not to be related to the phone system.

  7. We have a snom phone that is used for client surveys and the one key is very close to mute and when they acidently mute the phone and coninue to press 1 or 2 the call will soon disconect. Also the DTMF is not reconized when the mic is muted. I don't have this problems with our Polycom phones. I experiencing this problems with Snom 320 and 300 firmware 6.5.12

  8. I tried that. This adjust the line impedance not DB levels.

     

    I bought one of these and it completely got rid of the echo I had on the line or at least reduced it enough so the built in echo cancelor on my AdioCodes could do it's job.

  9. User reported the sound of dial tone during a call then it disconnected soon after at 15:47

     

    [7] 2007/10/02 15:45:15: UDP: Opening socket on port 49790

    [7] 2007/10/02 15:45:15: UDP: Opening socket on port 49791

    [5] 2007/10/02 15:45:15: Identify trunk (IP address/port and domain match) 1

    [7] 2007/10/02 15:45:15: Set packet length to 20

    [6] 2007/10/02 15:45:15: Sending RTP to 1.1.1.2:2808

    [5] 2007/10/02 15:45:15: Trunk PSTN sends call to 702

    [7] 2007/10/02 15:45:15: Hunt Group: Moving to next stage

    [7] 2007/10/02 15:45:15: Set packet length to 20

    [7] 2007/10/02 15:45:15: UDP: Opening socket on port 56492

    [7] 2007/10/02 15:45:15: UDP: Opening socket on port 56493

    [7] 2007/10/02 15:45:16: Call a80aa7f8@pbx#592129394: Clear last request

    [7] 2007/10/02 15:45:18: Call a80aa7f8@pbx#592129394: Clear last INVITE

    [6] 2007/10/02 15:45:18: Sending RTP to 192.168.100.105:52406

    [7] 2007/10/02 15:45:19: a80aa7f8@pbx#592129394: RTP pass-through mode

    [7] 2007/10/02 15:45:19: 5a5ddbae@fxo#ecf78bc533: RTP pass-through mode

    [7] 2007/10/02 15:45:22: Other Ports: 3

    [7] 2007/10/02 15:45:22: Call Port: 3c2694452710-dfnf0pxszqn8@snom320-000413247E1C#f30ea65493

    [7] 2007/10/02 15:45:22: Call Port: 5a5ddbae@fxo#ecf78bc533

    [7] 2007/10/02 15:45:22: Call Port: d045298a@pbx#865348473

    [7] 2007/10/02 15:45:22: Call 5a5ddbae@fxo#ecf78bc533: Clear last request

    [5] 2007/10/02 15:45:22: BYE Response: Terminate 5a5ddbae@fxo

    [7] 2007/10/02 15:45:22: Other Ports: 2

    [7] 2007/10/02 15:45:22: Call Port: 3c2694452710-dfnf0pxszqn8@snom320-000413247E1C#f30ea65493

    [7] 2007/10/02 15:45:22: Call Port: d045298a@pbx#865348473

    [7] 2007/10/02 15:45:39: UDP: Opening socket on port 64326

    [7] 2007/10/02 15:45:39: UDP: Opening socket on port 64327

    [5] 2007/10/02 15:45:39: Identify trunk (IP address/port and domain match) 1

    [7] 2007/10/02 15:45:39: Set packet length to 20

    [6] 2007/10/02 15:45:39: Sending RTP to 1.1.1.2:2810

    [5] 2007/10/02 15:45:39: Trunk PSTN sends call to 702

    [7] 2007/10/02 15:45:39: Hunt Group: Moving to next stage

    [7] 2007/10/02 15:45:40: Set packet length to 20

    [7] 2007/10/02 15:45:40: UDP: Opening socket on port 52124

    [7] 2007/10/02 15:45:40: UDP: Opening socket on port 52125

    [7] 2007/10/02 15:45:40: Call e8c6e0ff@pbx#2128230563: Clear last request

    [7] 2007/10/02 15:45:47: Call e8c6e0ff@pbx#2128230563: Clear last INVITE

    [6] 2007/10/02 15:45:47: Sending RTP to 192.168.100.105:64584

    [7] 2007/10/02 15:45:47: e8c6e0ff@pbx#2128230563: RTP pass-through mode

    [7] 2007/10/02 15:45:47: eb90a18c@fxo#19c97cd0ec: RTP pass-through mode

    [7] 2007/10/02 15:46:17: Set packet length to 20

    [7] 2007/10/02 15:46:17: e8c6e0ff@pbx#2128230563: Media-aware pass-through mode

    [7] 2007/10/02 15:46:17: eb90a18c@fxo#19c97cd0ec: Media-aware pass-through mode

    [5] 2007/10/02 15:46:18: Redirecting call to 502

    [7] 2007/10/02 15:46:18: eb90a18c@fxo#19c97cd0ec: RTP pass-through mode

    [7] 2007/10/02 15:46:18: Calling extension 502

    [7] 2007/10/02 15:46:18: UDP: Opening socket on port 63348

    [7] 2007/10/02 15:46:18: UDP: Opening socket on port 63349

    [7] 2007/10/02 15:46:18: eb90a18c@fxo#19c97cd0ec: Media-aware pass-through mode

    [7] 2007/10/02 15:46:19: Call e8c6e0ff@pbx#2128230563: Clear last request

    [5] 2007/10/02 15:46:19: BYE Response: Terminate e8c6e0ff@pbx

    [7] 2007/10/02 15:46:19: Other Ports: 4

    [7] 2007/10/02 15:46:19: Call Port: 2d00d77f@pbx#1514598187

    [7] 2007/10/02 15:46:19: Call Port: 3c2694452710-dfnf0pxszqn8@snom320-000413247E1C#f30ea65493

    [7] 2007/10/02 15:46:19: Call Port: d045298a@pbx#865348473

    [7] 2007/10/02 15:46:19: Call Port: eb90a18c@fxo#19c97cd0ec

    [7] 2007/10/02 15:46:19: Call 2d00d77f@pbx#1514598187: Clear last request

    [7] 2007/10/02 15:46:38: Last message repeated 2 times

    [7] 2007/10/02 15:46:38: Call 2d00d77f@pbx#1514598187: Clear last INVITE

    [7] 2007/10/02 15:46:55: Other Ports: 2

    [7] 2007/10/02 15:46:55: Call Port: 3c2694452710-dfnf0pxszqn8@snom320-000413247E1C#f30ea65493

    [7] 2007/10/02 15:46:55: Call Port: d045298a@pbx#865348473

    [7] 2007/10/02 15:47:34: Call d045298a@pbx#865348473: Clear last INVITE

    [7] 2007/10/02 15:47:39: UDP: Opening socket on port 49472

    [7] 2007/10/02 15:47:39: UDP: Opening socket on port 49473

    [7] 2007/10/02 15:47:39: Set packet length to 20

    [6] 2007/10/02 15:47:39: Sending RTP to 192.168.100.105:57492

    [5] 2007/10/02 15:47:39: Dialplan: Match 3919429@192.168.100.9 to <sip:3919429@localhost;user=phone> on trunk PSTN

    [7] 2007/10/02 15:47:39: UDP: Opening socket on port 52372

    [7] 2007/10/02 15:47:39: UDP: Opening socket on port 52373

    [7] 2007/10/02 15:47:39: Set packet length to 20

    [6] 2007/10/02 15:47:39: Sending RTP to 1.1.1.2:2812

    [7] 2007/10/02 15:47:42: Call 12b4bcd2@pbx#1294427488: Clear last INVITE

    [7] 2007/10/02 15:47:42: Set packet length to 20

    [7] 2007/10/02 15:47:43: 12b4bcd2@pbx#1294427488: RTP pass-through mode

    [7] 2007/10/02 15:47:43: 3c28276330d4-z5m5zdnniw54@snom320-00041324C675#83d8bd4ebc: RTP pass-through mode

    [7] 2007/10/02 15:48:17: Set packet length to 20

    [7] 2007/10/02 15:48:17: Call d045298a@pbx#865348473: Clear last request

    [5] 2007/10/02 15:48:17: BYE Response: Terminate d045298a@pbx

    [7] 2007/10/02 15:48:17: Other Ports: 2

    [7] 2007/10/02 15:48:17: Call Port: 12b4bcd2@pbx#1294427488

    [7] 2007/10/02 15:48:17: Call Port: 3c28276330d4-z5m5zdnniw54@snom320-00041324C675#83d8bd4ebc

    [7] 2007/10/02 15:48:19: UDP: Opening socket on port 63696

    [7] 2007/10/02 15:48:19: UDP: Opening socket on port 63697

    [7] 2007/10/02 15:48:19: Set packet length to 20

    [6] 2007/10/02 15:48:19: Sending RTP to 192.168.100.111:58066

    [5] 2007/10/02 15:48:19: Dialplan: Match 5093462174@192.168.100.9 to <sip:15093462174@localhost;user=phone> on trunk PSTN

    [7] 2007/10/02 15:48:19: UDP: Opening socket on port 63050

    [7] 2007/10/02 15:48:19: UDP: Opening socket on port 63051

    [7] 2007/10/02 15:48:19: Set packet length to 20

    [6] 2007/10/02 15:48:19: Sending RTP to 1.1.1.2:2814

    [7] 2007/10/02 15:48:24: Set packet length to 20

    [7] 2007/10/02 15:48:24: fec499c9@pbx#1596244034: RTP pass-through mode

    [7] 2007/10/02 15:48:24: 3c26952d7530-n5uh992z5kbv@snom320-000413247E1C#2e0b360259: RTP pass-through mode

    [6] 2007/10/02 15:48:26: Received DTMF 5

    [7] 2007/10/02 15:48:30: Call fec499c9@pbx#1596244034: Clear last request

    [7] 2007/10/02 15:48:30: Call fec499c9@pbx#1596244034: Clear last INVITE

    [7] 2007/10/02 15:48:30: Other Ports: 3

    [7] 2007/10/02 15:48:30: Call Port: 12b4bcd2@pbx#1294427488

    [7] 2007/10/02 15:48:30: Call Port: 3c28276330d4-z5m5zdnniw54@snom320-00041324C675#83d8bd4ebc

    [7] 2007/10/02 15:48:30: Call Port: fec499c9@pbx#1596244034

  10. I know how to ftp the file over but what directory do I put it. Instructions would be helpful.

     

    I know how to do for the tar file from the wiki:

    cd /pbx

    tar xvfz update.tgz

    cd update

    ./pbx_install.sh

    sync;reboot;exit

     

    but that doesn't work on this file. I'm getting RTP timeouts with the tar update. I need to install the newer version but don't know how. Any help would be greatly appreciated.

     

    Jason

  11. I'm having the same problem with call dropping if muted longer land 2.5 minutes. I'm using Snom 320 phones with latest firmware. We really need to have mute availbe for confernce calls. What's the best way to make it so it does not drop the calls.

  12. My system is handing up about once a day. I can call internally but incomming and outgoing calls do not work. The only thing I can do is reboot the sysetm and it works fine.

     

    My question: Is there a way to have the system reboot on a scheduled time?

  13. Hi,

     

    I need help with setting up my AudioCodes gatway with my CS 410 appliance. I can dial in and out of the CS 410 with no problems but I'm having problems getting incomming and outgoing call to work with the audiocodes gateway. I bought the equipment through atacomm would I be better off getting professional paid support from them or purchasing 1 -3 hrs of support from pbxnsip.

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