Art King
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Posts posted by Art King
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We are in trouble. Our formerly stable pbxnsip configuration is now giving us grief.
We are running 2.1.12.2489 (Win32) with about 30 active extensions using snom 360 and 300 phones. Trunking is a T1 via an AudioCodes TP-260.
Last week we opened the web interface to add a new extension. The new extension grabbed the Alias and settings of an existing extension. We then deleted the new extension, and the existing one and attempted to re-create them both. This time two other existing extensions were corrupted. The more changes we made, the more unexpected results showed up. Finally we gave up and did a complete restore from the previous night's backup of the pbx folder. Things were then stable again, and we didn't try to make any changes that day.
Yesterday I created a new Trunk and Dial Plan for testing. A few minutes later we noticed that one person's phone wasn't working, and when we looked at the Account list, his extension had disappeared. We again did a complete restore from backup and restart.
At this point, we are afraid to make any changes via the web interface, for fear of other unexpected results.
I have a backup "pbx.tar" file that I can sent if that would help someone debug this problem.
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In version 2.1, you can use the "parameter2" also for ANI. It is a little dirty trick that we put in some time.
In version 3 we cleaned that area up and introduced a new parameter called "ANI" that solves the problem the right way.
This is the second post I've seen indicating that this problem is solved in version 3. When is version 3 going to be released? Is there a beta?
The latest version I see on the web site is 2.1.12.2489
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Well, the rules are defined in http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk.
In the next version we'll have a new setting called "ANI" for every account, where you can explicity put the caller-ID that you want to this specific account.
I think we need to write something up that explains how this whole Caller-ID presentation works in theory and in real ITSP-operator life.
We just switched providers and they block any call that doesn't have ANI that matches one of ours. Our providor is concerned about Caller-ID spoofing, and they won't relax this restriction.
We are running version 2.1.10
Your documentation says: "If the call is redirected from another incoming call, it will use the original number."
We can't forward calls to cell phones, or even manually transfer calls to outside numbers since the PBX always tries to use the ANI from the incoming call.
We need a setting for "If the call is redirected from another incoming call, use the ANI for the the station doing the redirecting" Help!
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All our users have their mailbox configured to send voice mail messages direct to our email system as WAV attachments.
What we would like is to have a system that will deliver both the WAV file, and also an email with the text of the message, converted by a speech recognition engine.
I assumed I could find a service that would allow me to configure the PBX to send my WAV files to them, and they would return the converted text to my inbox. I can’t find such a service.
Do you know of anyone integrating this capability with PBXNSIP?
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Strange. Well, usually any call should be cleared after two hours, unless yo uchange the setting on system level. What happens if you hit the delete button for that link?
My system level setting is 4 hours. I clicked on the delete button and it disappeared.
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Here's a screen shot of the status screen: http://download.faxback.com/temp/pbxstatus.jpg
The status screen shows an active call 20 hours after the start time indicated. I checked my T1 interface and there were no active ports, so there was actually no call going on. The snom telephone at this extension shows no line lights are no activity.
What's going on?
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There is really not much difference between moderator and participant in 2.1.0.2115. In the 2.1.1.2209 we added the possiblity to kick out all other participants (*9) and send an email to the moderators email address with the list of current participants (*1).
This is more of a general question. Where can I look on the forum to find the latest version? I thought 2.1.0.2115 was the latest. I did a forum search for "2.1.1.2209" and found nothing.
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I recently upgraded to version 2.1.0.2115 (Win32) and I'm looking at the updated features for Ad-hock conferencing. I've set up a conference extension that looks like this:
I can enter the conference using either "Moderator" or "Participant" access codes. I can't see any differences between using the two codes from my testing. I looked on the Wiki and I can't find any new documentation for the Moderator access code.
How does it work? What is the "Dialog Permissions" setting?
Thank you.
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We are running version 2.0.3.1715 (Win32)
When PBXnSIP sends an email with a voice mail WAV file attached, the body of the email includes several broken hyperlinks. Is there a way to control the content or formatting of the email template? How do I fix the broken links?
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Ehh... Not sure if it is available in 2.0.3.1715. It would be in the System Admin/Settings page. But it will be for sure in 2.1.
In not, there might be a setting called "camp_enabled" in the pbx.xml file. If it is there, change it to "false". That would require a restart of the service.
Thanx! xml file update works.
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Just go to the admin settings and turn "Offer Camp On" off.
Please be a little more explicit about where this setting is found. I've already looked everywhere. I'm running version 2.0.3.1715.
Thank you.
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I call an extension that has a standard voice mail setup. After a few rings the call rolls to voice mail and the first prompt says;
"There was no answer to your request. To receive a call back when the extension becomes available, press 1. To leave a message press 2."
If I press 1 nothing happens. However, I don't even want this feature. I've read all through the manual and I can't figure out how to turn off this prompt. Help! Thanx.
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Why 2.0.3.1707? There is 2.0.3.1715 which is the last release and this image likely fixed that DoS issue (I remember that intermediate build had a problem).
There is a global setting called "max_udp_invite" which defaults to 10. If you have more than 10 new calls per second, you might need to increase that value.
Thank you. I pulled the latest download link from this forum which was 1707. I see on the main download page that 1715 is the latest. I was confused.
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Last week we upgraded to 2.0.3 Build 1707 (Win32). Today for no apparent reason the phones stopped working. We turned on debug and found this in the logs:
[3] 2007/09/14 10:02:31: DoS protection: Not accepting more callsWe restarted the PBXnSIP service and an hour later it did the same thing. I see there is a brief comment in the Release Notes for 2.0.3 about Denial of Service being added.
Our PBX is behind our firewall and NAT we don't see any anomolous traffic on our network. Help!
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There is a new version available that you can use. Everything about this image is in http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.0. The image can be downloaded from http://www.pbxnsip.com/download/pbxctrl-2.1.0.2084.exe (Manual Upgrade).
The manual link URL does not work... but I see it's an earliier version.
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We've been testing pbxnsip v2.0.2.1676 (Win32) installation on a Dell Server with Xeon 3.0 GHZ dual-processor and 2GB of RAM, running Windows 2003 Server R2.
We kept seeing jitter. For example, if the web interface Call Status screen was up (http://pbx/dom_calls.htm), each time it would refresh we would get a short burst of jitter on the phones. We saw similar jitter episodes on a conference call each time someone would join or leave the conference.
We were monitoring Windows Task Manager and we noticed that the per-processor CPU Usage History graph did a lot of jumping around concurrent with the jitter. Under Task manager Processes, we selected pbxctrl.exe, right-clicked and chose "Set Affinity". By default the PBXnSIP process was set to use both of the processors. We changed it to only use one processor and the jitter problem vanished!
We found a Windows utility that allows us to set the Affinity to one processor for a specific process, so we don't have to remember to set this each time we reboot.
Looks to us like PBXNSIP needs to do more testing on multi-processing machines.
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Hmmm. Agreed, we need moderator a PIN. Next version.
Did this chage get into the most recent beta release?
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We upgraded from Snom firmware 6.5.8 to 6.5.10 because the snom web site showed that as the latest release. After the upgrade we started noticing that on many phone calls lasting more than 5 minutes the phone would suddently go to loud static. This was completely repeatable. We set up a group of test phones and put them on a conf call. The phones with version 6.5.10 would all eventually go to static. The phones with the earlier firmware 6.5.8 would not.
We've downgraded all our phones back to 6.5.8 and the problem is gone.
We are running pbxnsip v2.0.2.1676 (win32)
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Good question. In an ad-hoc conference there is no moderator. I am not sure if it is a good idea to allow everyone to kick everyone else out.
Other ad-hock conference tools have a participant PIN and a moderator PIN. When the moderator hangs up, the participants have only xx seconds left before they get disconnected.
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In the latest version (2.0.3) the moderator can terminate the conference with *9. See http://wiki.pbxnsip.com/index.php/Scheduling_Conferences and http://wiki.pbxnsip.com/index.php/Conferencing.
As far as I can tell, the feature you describe only applies to scheduled conferences. How do you assign a moderator to an ad-hoc conference? How do you terminate an ad-hoc conference?
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Is there some sort of administration mode for conferences? What we need is the ability to setup an ad-hock conference, and when the moderator hangs up, the conference ends withing xx seconds. Is there a way to do that either automatically or manually?
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I'm running version 2.0.3.1715 (Win32). In the Extension Setup screen is "Watch the calls of the following extensions"
The pbxnsip wiki ( http://wiki.pbxnsip.com/index.php/Extension ) does not document this feature.
How does it work?
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Is there a way to modify the template for the email that is sent with a WAV attachment from voice mail? I thought I would find a sub-folder under \PBX that contained an HTML file template, but no such thing exists.
Version: 2.0.3.1715 (Win32)
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We have an external application that does IVR functions via SIP. It works fine when we access it directly from an AudioCodes gateway. I want to be able to make a call through PBXNSIP to my external SIP application.
I setup a new Trunk on the PBXNSIP and I configured a Dialplan to send calls to the external IVR. I can make a call from the PBX to the to the external IVR and the IVR application answers, and the audio comes through fine. However, I cannot get my application to recognize DTMF from snom telephone, or from a softphone.
DTMF entries interrupt the voice prompts on the IVR system as they are supposed to. The IVR script is looking for four digits, and it accepts four digits correctly. However, the IVR log, shows it received the DTMF digits of ?5555? when what I entered was "5353"
In all my testing, it seems like whatever the first DTMF digit I enter, this digit is repeated for all the digits requested. So if I enter ?1234? the debug shows ?1111?
I changed the setting on PBXNSIP to allow ?Inband DTMF Detection? but it made no difference. I get different results when calling from the snom phone vs the X-Lite softphone, but I never have gotten it to work in any case.
Below is a section of the log showing the DTMF passing across to the external IVR from an X-Lite softphone.
Troubleshooting suggestions would be appreciated!
Art
The log below is from PBXNSIP
The softphone making the call is at IP address 10.0.0.85
The PBXNSIP is at IP address 10.0.0.22
The external IVR is at 10.0.0.205
INVITE sip:321@10.0.0.22 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:5353@10.0.0.85:37272>
To: "321"<sip:321@10.0.0.22>
From: "Art King (soft)"<sip:5353@10.0.0.22>;tag=2405ad3c
Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 316
v=0
o=- 5 2 IN IP4 10.0.0.85
s=CounterPath X-Lite 3.0
c=IN IP4 10.0.0.85
t=0 0
m=audio 61172 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : X+KZtdhy XCvsEoGu 10.0.0.85 61172
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
UDP: Opening socket on port 53760
UDP: Opening socket on port 53761
Identify trunk 6
Resolve destination 50438: a udp 10.0.0.85 37272
Resolve destination 50438: udp 10.0.0.85 37272
Send Packet 100
SIP Tx udp:10.0.0.85:37272:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport=37272
From: "Art King (soft)" <sip:5353@10.0.0.22>;tag=2405ad3c
To: "321" <sip:321@10.0.0.22>;tag=3151257388
Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM.
CSeq: 1 INVITE
Content-Length: 0
Sending RTP to 10.0.0.85:61172
Resolve destination 50439: url sip:5353@10.0.0.122
Resolve destination 50439: udp 10.0.0.122 5060
Send Packet NOTIFY
Dialplan: Match 321@10.0.0.22 to <sip:321@10.0.0.205;user=phone> on trunk ExtIVR
Play audio_moh/noise.wav
UDP: Opening socket on port 51638
UDP: Opening socket on port 51639
Resolve destination 50440: url sip:10.0.0.205
Resolve destination 50440: udp 10.0.0.205 5060
Send Packet INVITE
SIP Tx udp:10.0.0.205:5060:
INVITE sip:321@10.0.0.205;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.22:5060;branch=z9hG4bK-47c500d1bc1c4e59da4c9db28bd275a3;rport
From: "ExtIVR" <sip:10.0.0.205>;tag=46519
To: <sip:321@10.0.0.205;user=phone>
Call-ID: 29cca7cd@pbx
CSeq: 29522 INVITE
Max-Forwards: 70
Contact: <sip:10.0.0.22:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.0.3.1715
Content-Type: application/sdp
Content-Length: 286
v=0
o=- 26774 26774 IN IP4 10.0.0.22
s=-
c=IN IP4 10.0.0.22
t=0 0
m=audio 51638 RTP/AVP 0 8 18 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=sendrecv
Resolve destination 50441: a udp 10.0.0.85 37272
Resolve destination 50441: udp 10.0.0.85 37272
Send Packet 183
SIP Tx udp:10.0.0.85:37272:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport=37272
From: "Art King (soft)" <sip:5353@10.0.0.22>;tag=2405ad3c
To: "321" <sip:321@10.0.0.22>;tag=3151257388
Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM.
CSeq: 1 INVITE
Contact: <sip:5353@10.0.0.22:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.0.3.1715
Content-Type: application/sdp
Content-Length: 186
v=0
o=- 10805 10805 IN IP4 10.0.0.22
s=-
c=IN IP4 10.0.0.22
t=0 0
m=audio 53760 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=sendrecv
How to integrate PBXnSIP with NET SatisFAXtion
in Fax Setup
Posted
FaxBack Inc's NET SatisFAXtion version 8.5 IP fax server integrates very well with PBXnSIP. There is an excellent "How To" in the FaxBack knowlege base here:
http://kb.faxback.com/HOWTO+-+pbxnsip+Integration