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Art King

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Everything posted by Art King

  1. 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
  2. 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.
  3. 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
  4. 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!
  5. 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?
  6. My system level setting is 4 hours. I clicked on the delete button and it disappeared.
  7. 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?
  8. 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.
  9. 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.
  10. 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?
  11. 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.
  12. 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.
  13. 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.
  14. 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 calls We 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!
  15. The manual link URL does not work... but I see it's an earliier version.
  16. 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.
  17. Did this chage get into the most recent beta release?
  18. 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)
  19. 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.
  20. 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?
  21. 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?
  22. 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?
  23. 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)
  24. 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
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