shopcomputer
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Posts posted by shopcomputer
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I contacted Snom, they said the problem may happen, if a user pushes transfer again instead of check/confirm key.
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Snom 360 7.30
If they have 2 calls on hold and try to pick 1 up and transfer them to another extension, the 2 callers get conferenced, and the user gets knocked out of the call.
call join on xfer is set to off on the phone.
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I can confirm this is a bug in at least the last few releases, you can find several posts in the forum about this. I you an IVR node instead fo the sub attendant, to work around the issue.
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We use http://www.vikingelectronics.com door box, hooked to an fxo port.
we also tried http://www.its-tel.com/info.asp?id=48 and have http://www.cyberdata.net/products/voip/voip-intercom.html on order, it will be availible in a few weeks.
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Actually I can almost confirm this to be a problem, just tried leaving myself a voicemail, and the call with the VM notification, came in from unknown.
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Did you try playing with the Remote Party/Privacy Indication:on the trunk?
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yes
[0] 2008/03/19 19:09:55: SIP Tx tcp:192.168.0.25:2229:
MESSAGE sip:25@192.168.0.25:2229;transport=tcp;reg-id=1 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.10:5060;branch=z9hG4bK-121be68286544064cfbd4c20411129fc;rport
From: "PAC " <sip:25@localhost>;tag=17103
To: "PAC " <sip:25@localhost>
Call-ID: bhby5n1k@pbx
CSeq: 55969 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.0.10:5060;transport=tcp>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 37
k=34
c=on
x=ext
m=0/0
d=Test
[0] 2008/03/19 19:09:56: SIP Tx tcp:192.168.0.25:2229:
MESSAGE sip:25@192.168.0.25:2229;transport=tcp;reg-id=1 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.10:5060;branch=z9hG4bK-f2c10bf40dc8efecdc564a75afb730d4;rport
From: "PAC " <sip:25@localhost>;tag=50925
To: "PAC " <sip:25@localhost>
Call-ID: fvhot3lm@pbx
CSeq: 12784 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.0.10:5060;transport=tcp>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 37
k=34
c=on
x=ext
m=0/0
d=Test
[0] 2008/03/19 19:09:56: SIP Tx tcp:192.168.0.25:2229:
MESSAGE sip:25@192.168.0.25:2229;transport=tcp;reg-id=1 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.10:5060;branch=z9hG4bK-01b1cd79c3e9062e449a06b99a3dcb0c;rport
From: "PAC " <sip:25@localhost>;tag=55992
To: "PAC " <sip:25@localhost>
Call-ID: 02ajx3ab@pbx
CSeq: 16212 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.0.10:5060;transport=tcp>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 29
k=L4
c=on
x=line
l=co4
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I am using a vitelity wholesale account on my in house system. I beleive the brodvox trunks at my clients location also supports passing the caller id. All I needed to do was thet the Remote Party/Privacy Indication: to remote-party-id.
I wish I would be able to manipulate it to change the first digit or 2 of the are code with a code, telling me it is the pbx or mailbox calling.
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As I have extension numbers that start with a 4, the # puts a 2 second timeout before the redirection, so a user can enter ext 45, before it redirects to the 4 shortcut.
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When a call redirects to my cell phone, it shows the original caller ID which is good. Is there any way to set a prefix to the caller ID, so I know the call is coming from the pbx rather than someone dialing my cell directly.
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Thanks Chuck,
That did get the application to show the extensions and lines. The extensions seem to work, however the lines are not showing anything.
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Can the PAC monitor service flag, or park orbit button, etc.?
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Is there a hard limit to the amount of buttons that can be added to a button group? are only line an extension buttons supported, or all other button types are also suported?
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Is any way to that when a call comes in it should display both the name and the number. It is only displaying the name on my test system, if the name is not availible it displays the number.
I have a prospect who has this as a required feature.
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Is that the same issue we are having with the prerilli phones, not stopping to ring?
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2.1.5 on the pbx, not sure about the phone.
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I have a client demoing pbxnsip for the past 3 weeks, I just contacted them to see how things are going, they are running version 2.15 on Ubuntu running on a fanless computer, I got from PBxnsip. Here a problem they have and want a resolution before purchasing.
About 2 weeks ago, when a call was made both incoming and outgoing, it took several seconds for the call to be connected, you did not hear any voice until about 3 seconds in to the call, same was for call transfers as well. A reboot of the PBX fixed this issue, however you want to make sure it does not reoccur. Do you happen to know if same was for internal calls, for say if you called from your desk to the next desk? Or you did not try that.
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When a call comes in, and you transfer the call to another extension, many times the Grandstream phone lights up the line 1 and line 2 on the phones during the transfer. This does not happen on every transfer however it does happen very often, when you pick up the call it does connect the call properly.
Also any way get Busy lamp working on this phone?
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Cisco has several gigabit models, for ex. 7961G-GE.
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Not yet, I did not want to disable all unused codecs remotely, and I did not visit the client since.
it does work fine with version 2.1.5, so you can downgrade and get it working.
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Which ports did you open? Are you using a sip aware router.
Also look at http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses
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I found a listing of 311 call centers here, http://www.911dispatch.com/info/311map.html I set my dial plan to use the local 311 phone number and now 311 works. Does anyone have a list of 911 numbers to share, or at least the New York 911 number?
Several pages in the wiki regarding emergency calling point here, http://wiki.pbxnsip.com/index.php?title=Emergency_Calling however this is a blank page.
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Do you have competitive list, to compete with Fonality?
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The pbxnsip is set to 0 8 18 2 3, the Pirelli was set preffered codec g729, I changed it to g711u. I don't recall seeing a setting to restrict to g711 only preffered, although I will double check next time I am at the clients location.
Should I change the pbxnsip to 0 only? Broadvox the ITSP as well as the audiocodes gateway, snom, polycom and aastra phones we use all support G711u.
One trunk with several numbers
in Trunk Setup
Posted
You can leave the route to field empty on the trunk, just add an alias on the extensions, with the DID that needs to be routed to that extension.