andy

you are right. sorry for the typo. obviously i was refering to server2008. however, the pbxnsip service deactivates shortly after login. same effect on serfver 2008 :-)

it looks like a great job, BUT this RC stops running on our server (Windows Server 2009) a few seconds after the admin login. v3 has beenrunning super stable so far on the very same machine.

I wonder if anyone can help me with this quite special request. I would like to route certain caller-ids automatically to specific extensions. In this case the calling party would not need to know what extension to dial. His caller-id would be recognized by the system (or a customer database), so the PBX can send his call to the right extension automatically. It would be great if the blacklist feature would allow to sent listed caller-ids to specific extensions.

We operate most phones within our LAN, but I have 2-3 phones at a remote location. Those should ideally connect through our VPN to the PBXNSIP server. First question: Should this simple configuration work with PBXNSIP? So far it does not. PBXNSIP server is on IP 192.168.26.130/Subnet: 255.255.255.192 Remote phone (Snom320) is on IP 192.168.26.26/Subnet: 255.255.255.240 Server and phone are using the same DNS: 192.168.26.130 All ports are open within the VPN, but the Snom320 reports: Network Failure. Second question: Would it be better to set up a second (smaller) PBXNSIP server at the remote location and connect both servers in order to get this stable? I am not sure to do from the info I got from the forum so far.

I have the very same problem. I checked on the firewall, the http Ports. No success so far. Is there a way to listen to the traffic of the PAC on oder to find the problem?

We found out what happened. The problem was caused by one of our SIP providers that delivered instable services for a couple of days (with the trunk connecting sometimes, sometimes not). If the auto attendant transfers the call to a SIP line that's not available for a short moment it just hangs itself up. Now we selected a different SIP trunk for these calls and the problems is gone.

My domain is localhost. I didn't setup the name "_cpempty_". I just wanted to make sure that I wasn't missing something on my configuration. But the system seems to work anyway.

I have a reoccuring issue with the auto attendant when it's called by a cell phone. Once the caller has entered a number to select an option/extension, the call is not beeing transfered to the chosen option/extension, but there is an echo only of the callers audio nothing happens anymore. Only restarting the pbx process helped so far. Any idea what is causing this problem and how to create a workaround?

Since I changed to V 2.1 I find records like this in the log data: Dialplan default: Match xxx@_cpempty_ to <sip:xxx@192.168.26.140;user=phone> on trunk voxip The PBX works fine but I am still curious what "_cpempty_" stands for, as I can't remember to have seen this in older logs.

Probably found a bug with the mailbox. Please check my post under Mailbox for details. Moving back to an older built solved the problem immediately.

Changed the pbxctrl.exe back to v 3.1.0.3043 (Win32) and everything is working as it should. There seem to be some nasty bugs out there.

We have major trouble with mailbox messages since I upgraded to v 3.1.1.3110 (Win32). Messages are not beeing delivered by mail anymore, even though there was no change of settings (at least I believe so). The SMTP system is ok. CDR records are beeing delivered by PBXNSIP with no problems at all. The log shows these lines: [7] 2008/12/20 14:45:28: Attendant: Calling extension 550 [8] 2008/12/20 14:45:28: Play recordings/personal26-1.wav audio_moh/mb_beep.wav [7] 2008/12/20 14:45:28: Set packet length to 20 [5] 2008/12/20 14:45:42: Message file recordings/msg27.wav was removed, removing associated message Another strange matter: I changed the SMTP server for testing. After this change a couple of voicemails did come in, but not for long. After a short while every test message is disappearing. Why would the system delete our messages? In the email settings the option "After sending a message:" is set to "Keep message as new message"? However, the system does not keep them.

This is a known windows bug. This problem is not caused by PBXNSIP. Exactly the same thing happens with other software that had a service installed. After deinstallation the Service Manager will still show the program listed, even though all folders and files are gone from the program directory. I would not really recommend to edit the registry, except you know exactly what you are doing. Just leave everything like it is and set the dead service link to deactivated.

We have a Sonicwall TZ170 with Enhanced OS on a single IP connection. Works very reliable since 2 years.

Thanks for your support. IPv6 is deactived and I will also analyze the PCAP trace within the next couple of days. Grüsse in die Friedrichstrasse aus Berlin-Mitte!

Server 2008 on our main subnet at 192.168.26.130 and doesn not receive any package. The testing phone (Snom300) is on another subnet at 192.168.26.26. The traffic in between the subnets is not filtered by a firewall according to our networking guys. Maybe the Snom's log is of any help (see attachment). snom_log.txt

We tried to connect additional phones via VPN from a remote location, but these phones failed to register. The network has three different subnets. There are no firewall restrictions, all ports are open. Pings, remote file access and remote desktop are all working fine. Any idea what could cause that problem?

Well, we shouldn't discuss here who should get fired but rather find technical solutions. And we have found it: Simple but very efficient. Without any changes to the server or the network, a completely new install and setup solved the problem. Probably some of the XML files were corrupt. But this seems to be extremely rare according to the very helpful people at PBXNSIP. Usually a simple transfer of the settings should do it. Case closed.

Our networking technician was did check the server today and he did not find any unusual behaviour or settings. It's a new MS Server 2008, but that should not make a real difference. netstat -abn shows UDP Port 5060 listening at IP 0.0.0.0. PBXNSIP shows both IPS's 127.0.0.1 and 192.168.26.130 under Admin/Status/General. The firewall is turned off. The phones are adressing 192.168.26.130 (extension@192.168.26.130:5060). Should I simply delete the complete PBXNSIP setup and install it from scratch? Maybe there is a problem with some of the XML settings taken over from the old machine.

No solution yet. We are still running our phones on a conventional ISDN phone system which we kept for backup. And luckily I am not a professional technician who gets fired on this matter, but a small companie's manager who has to take care of our network besides his regular job. But I believe in technology so it was worth the try to go with a soft PBX. Actually we've been quite happy with PBXNSIP until this incident, as it is way more flexible than a conventional small phone system. Here is a new hint from the PBX's log. 192.168.26.141 is one of our Snom phones. It's request to register seems to find it's way into the PBX somehow, otherwise there would be no log entry like this one: [8] 2008/03/30 10:07:21: SIP Rx udp:192.168.26.141:5060: REGISTER sip:192.168.26.130 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.141:5060;branch=z9hG4bK-fsc2w36u2lpq;rport From: "andreas" <sip:200@192.168.26.130>;tag=1zxwzngbuq To: "andreas" <sip:200@192.168.26.130> Call-ID: 3c267012321c-xkx751bzghuq CSeq: 2 REGISTER Max-Forwards: 70 Contact: <sip:200@192.168.26.141:5060>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:6b24e3ed-00ba-4fd5-95c1-5cd623b49ef0>";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" Contact: <https://192.168.26.141:443> User-Agent: snom320/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.26.141 Expires: 60 Content-Length: 0 There is an internal number 200 and it seems the PBX can't find it. This is the next few lines of the log: [8] 2008/03/30 10:07:21: SIP Tx udp:192.168.26.141:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.26.141:5060;branch=z9hG4bK-fsc2w36u2lpq;rport=5060 From: "andreas" <sip:200@192.168.26.130>;tag=1zxwzngbuq To: "andreas" <sip:200@192.168.26.130>;tag=55fce6be84 Call-ID: 3c267012321c-xkx751bzghuq CSeq: 2 REGISTER Content-Length: 0 So far I though the phone can't find the server at all. That does not seem to be the case. Maybe there is someone in this forum who understands the full technical meaning of this log.

I don't think that's the reason. Our replacement machine has the same IP, no firewall and no MAC learning activated. The only difference is the operating system. The broken server was running Server 2003, the replacement machine Win XP. The internal DNS is down, but we are using IP numbers anyway to adress all components. We'll - the server will be back in operation tomorrow. Still it would be interesting to find the problem for future reference, because it's not really possible in a small company to have redundant machines for all servers, as breakdown are quite rare anyway.

Our main server crashed so I had to move the PBX on a spare machine temporarily. Our licence is dongled, so there is basically no problem to do that. Result: The http interface is available, all trunks are registering, but the phones can't find the server and register with the setip extensions, even though the spare machine has the same IP and no firewall. I installed the software on the new machine first from a downloaded PBXNSIP installation package. Then I copied the complete content of my previously backed up PBX folder into the new installation. Would you see any reason why the phone would not find the PBXNSIP server? Maybe there are any settings that can't simply be transferred from one machine to another.

Upon a redirect, that is once a caller is dialing an optional number through the Direct Destination feature.

1. Once an audio file has been uploaded to the auto attendant, the field for entering the wav file's location stays empty. We have to change our announcement from time to time and it would be useful to see what file is currently loaded. 2. I would be cool if the system would convert audio files into 8 kHz Mono, 16 bit file automatically, or even would accept audio files of better quality. The sound quality really could be a little better. No everyone is using analog lines any more. Thanks and Merry Christmas to all. You made a great product so far. Besides of some minor problems it's really working great for our company .

Problem not really solved. I have the same problem. There is no ringtone on the caller's side upon a transfer from the auto attendant, even though the called phone is ringing. This causes many callers to hang up because they assume their connection got lost somehow.