Vodia PBX
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I have an issue were an extension is a member of a shared mailbox, which is working fine, but if a message is left in the users personal mailbox 1st and then another is left in the shared mailbox, once the user has listened to the 1st voicemail message, the lamp is tuned off and does not indicate that there are still messages in the other mailbox, this is the same if the group voicemail was 1st, then the lamp does not show that there are still messages in the personal mailbox. The messages are still there though and in the case of a group mailbox the lamp remains lit on the other extensions.
The customer is using SNOM handsets.
Any one come accross this before?
The problem is simply that most (all?) phones have only one storage place for MWI and the last message is stored. The workaround is to register two extensions (all phones can register the same 2nd extension), then the phone will receive the MWI on two different accounts (one private and one shared).
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I have having issues with Scheduled Conferences...
I am using version 2.1.5, and we have an ad-hoc conference working without any problems...
I can setup the scheduled conference, but when a user logins and schedules a conference, no email is ever sent...
The email works for all users, and the domain... Every users gets emails for their voice mail messages, missed calls, etc... and it all works...
Is there anything else that has to be setup so that the scheduleded conference can send out the email?
Any chance to make a backup and move to 3.1.2? Anything in the logs? How did you enter the email address or the email addresses?
Second issue, our conference is an extension, is there anyway to have the email, once working, send out the phone number and extension? I can add a Tel Alias for the main Trunk, then it includes this in the mail and not the extension, if I leave the Tel Alias off it sends the Extension and not the phone number, I would like to do both...Eh, I nix understand... You mean how the conference room is represented to the outside world?
Last question, main reason to want to use scheduled conference is the recording feature, is there any way to start a recording on an ad-hoc conference...Nope. It needs to be a scheduled conference because this is the only way to pull the WAV down the web interface.
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Many thanks for your reply, I understand, it is a large topic. So to make it simple let me rephrase what I need.
All phones are LinkSys SPA962/SPA942
PBX is running under WinBox
Idealy, I would like PBX to generate whatever it need in order to have all SPA phones to reade and display Company directory.
Oh I did not know they support LDAP... That would be another way of reason the address book! Any links to that?
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Has any thought or consideration been given to the idea of supporting SNMP trapping? This would create far less load on the PBX responding to GETs (not much I suspect) but would allow a far more proactive management by exception policy.
Yea, I would say emails are more popular for immediate alerts than traps. In then end, the SNMP tools do nothing else than sending out emails. SNMP is great for monitoring resources 24x7, and there are a lot of great tools available. But for events, IMHO emails are easier.
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In post http://forum.pbxnsip.com/index.php?showtopic=1166 the reference is SSL is supported in V 3.0.xxxxx
Using 3.0.0.2993 (Linux) and cannot make this work. Is this a hidden XML setting or are we overlooking the obvious?
[8] 2009/01/15 00:20:11: SMTP: Received 220 2.0.0 Ready to start TLS
[8] 2009/01/15 00:20:12: SMTP: Received
[8] 2009/01/15 00:20:12: Last message repeated 2 times
[5] 2009/01/15 00:20:12: SMTP: Connection refused on 74.43.63.17:25
[8] 2009/01/15 00:20:17: SMTP: Connect to 74.43.63.17:25
Maybe a simple problem with the TLS negotiation. Anyway, you can use the global setting "smtp_starttls" to enable or disable TLS. Set it to "false" if you want to disable it.
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ever since we upgraded to versino.3120 any grandstreams phone in our enviornment are unable to send dtmf in voicemail menu...DTMF works when calling other phone numbers.
There is probably a problem with the DTMF codec negotiation. Try the latest Grandstream firmware or try setting the codec on the phone to 101, that's the codec the PBX will propose.
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How can I setup a redundant/failover installation of PBXnSIP. I have 2 servers in 2 different locations that are connected via a run of point to point fiber with just media converters at each end. I want 1 server to be the main server but should it ever die I need to be able to use the backup. Is this possible and if so is there a document that provides instruction.
At the moment you need to use the Windows Server features for failover. Those features include file system replication, and also service recovery. Don't ask me details, I just heared it is possible to set everything up. And it costs more than you pay for two PBX licenses together. Seems like the OS vendor knows how valuable this feature may be!
For the next major release, we want to make this job easier and include some basic settings that also automatically bring up a secondary service.
If you can wait a little, you can just make sure that you run a backup job every night and have a second server ready to go when the first hardware fails. Then the failover would have to be done manually (or by a hand-crafted script).
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5539 275.606445 {SIP:20, UDP:19, IPv4:18} 10.1.1.99 PBX01 SDP SDP:Request: INVITE sip:2113@2114@example.com SIP/2.0;
The double @ looks very suspicious to me...
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Running 3.1.2 on Windows box
1- Does PBXnSIP support Company Directory in some ways? and does it generate content that can be acessible company wide? If yes How?
If not
2- Does LDAP could be a good solution? and How
My goal is to get everyone access to the complete company directory on every phone, either by grabbing some file from PBX or true provisioning (ould be better)
That is a difficult topic.
First, the question is: Who wants to read the directory?
If it is the phone, then we have a couple of methods. Polycom gets the address book provisioned after the reboot, snom pulls it down in its XML format (via HTTP), and there are new ideas about using XCAP for a more interoperable way of having an address book.
If it is the PBX that should pull down the directory (especially from a ActiveDirectory), that is a completely different topic.
And then we have a new topic coming up. LinkedIn, Xing, Salesforce, and I-don't-know what social networks are storing a lot of very interesting data (even with pictures!) that users would love to see on the display when a call comes in. A standard? No way. Everybody doing it in a different way. That is not making our life easier...
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Well, I don't see a lot of traffic, but no NOTIFY... Either that is the problem or the logging wasn't turned on the that. It would be good to see also the SUBSCRIBE for the resource.
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Not a problem, what level of logs will do? certenally not "0"
You need to log the SIP packets, SUBSCRIBE/NOTIFY. They are usually logged on level 7, unless you monitor a specifc IP address.
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Is there any easy way to tell the maximum number of simultaneous calls for a day, or since the PBX has been restarted? Just wondering since your hosting version is licensed by max simultaneous calls, and having an indication from the software would allow me to plan when a license upgrade is necessary.
Totally agree. There is not right now, but next version will have it.
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I am running 3.1.2.3120 win32 and for the first time ever I received an email stating
CPU meter shows 78, which is over the threshold of 75. Therefore, a new call has been rejected
I logged into the system and looked at the log file. The log file did not show anything for almost 2 minutes prior to this entry, and the previous entry was a call with nothing remarkable about it. The system is a 3 ghz pentium4 (single core)running Windows Server 2003 with 2 gigs of ram. It has a clean build, and the only additional software that was loaded on it was Retrospect (backup client software) and the management utility for the raid controller. I am trying to figure out why there was a spike. Based upon the call log, there were only 2 simultaneous calls going at the time of the spike, and other than the spike, I had run 24 hours with an average of only 5% processor utilization. Should I be looking for a backround program causing the problem, or is there a posibility I have something set wrong. Any suggestions about tracking it down are appreciated.
This incident caused tome to look back through my history of daily CPU utilization emails. I do see a large spike every 2 or 3 days, however it does not seem to be on a schedule, and this time must have been the first when a call happened to come in during it.
So far the only exceptional problem we can think of is that the logging to the file system slows down the process. E.g. when the file system is slow for whatever reason, the logging would block the RTP-specific thread. The workaround is to turn file system logging only on when you are actively debugging something.
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This does the trick. Only one little down, Lamp is not changing color either if logedin or out
any idea on where to look?
Hmm. Maybe you can post the XML documents that the PBX sends to the phone when the state changes (the body of the NOFIFY messages). Then we can see if the phone should believe that the resource is "active".
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We have a client running version 3.0.0.2998, the client site had power failure and when the pbx stared up again most of the configuration was lost. Could the power loss have caused this or not?
All configuration is stored in the file system. Usually a power failure does not wipe out the file system. Check if there is still data in the working directory and copy it somewhere.
Another reason to make back up back up back up. If you really have to set up everything again, and everyone is happy again, then make a back up! Next time you don't have to do this again...
Maybe someone who is a Windows guru can post a trick on how to SIP the working directory every night at three o'clock and move it to some safe place in the network. That would make many people's life a lot easier!
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This could have been the solution BUT, Web interface does not allow to have duplication code, If I try the web interface is telling me "Dupilcate"
Can we change the feature code someplace else?
Ehhhh.... Okay, maybe temporarily turn Javascript off. Or you can also use a handcraftet URL like this: http://yourpbx/dom_feature_codes.htm?editd...gent_logout=*75.
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Today we have a Snom 320 phone that refused to accept PNP auto with Buttons. remove the MAC address and you could manually set the phone up fine, place the MAC in the registration and shortly after booting you could not PING nor access the WEB of the phone. checked DHCP for odd reservation, moved phone to another switch, something was ODD. Trying to do as I preach by asking simple yes-no questions lead me to the PBX folder, assuming something was amiss I searched all XML files for the MAC address in question. Seems the MAC address of the phone that fails is in another extension config XML file in the extensions folder, I think the AVAIL field. We deleted and rebuilt that extension and VOILA' the phone would now autoprovision correctly.
I believe this is because the plug and play mechanism requires a password. Either change the TFTP password generation or set the password on the phone (see http://wiki.pbxnsip.com/index.php/Snom).
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i have a Polycom 550 with PBXnSIp which is working great configured through PNP.
however i consult for a second company and would like to be able to register my extension from second server on the same phone.
now i am aware that only one phone can be configured per extension and one system per phone using PNP but even when i try to manually add settings through the GIU or through the phone itself it never seems to be able to register.
does anyone have any experience with this to save me the trouble of having to spend hours on it?
Indeed, that would be tricky.
What I would do is take the files polycom_phone.xml from the generated directory, move it into the tftp directory and edit it there so that you have your second registration. Possibly you also need to check the other file, polycom_sip.xml for stuff like outbound proxy.
I believe editing the 2nd line in the web interface of the phone will not work. You'll probably loose the information on the next reboot.
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so the ivr node would be able to redirect a * prompt to any destination?
Yes. Though ERE are not very simple (you need to escape the * as it is a special symbol, \*).
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Ok great. One last question, we have looked at the Snom M3's are there any other good cordless options out there that PBXnSIP support?
Siemens Gigaset IP DECT devices are an alternative. Also, Polycom and Aastra have DECT devices - I believe those more targeted at large enterprises that require handover from one DECT base station to another.
If you use FXS ATA, you have a huge choice of cordless devices; however that audio quality may be a problem because of the additional A/D step and stupid things like hangup detection.
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Runnig 3.1.2 on Windows Box - CallCenter licence
Now that agent can log in and out I am trying to use a Key on our SAP962/SPA942 Agent phone to display status, either loged in or loged out just like a service flag will do.
for service flag we are usign
"fnc=sd+blf+cp;nme=Reception;sub=30@192.168.0.1;usr=30@192.168.0.1;vid=1"
if service flag is "clear" Lamp is Green
if service flag is "Set" lamp is Red
I tried the same wuth agent presence but since the it is usign "*64/*65" I need 2 phone buttons and it is not working well.
Any idea on how to create a beaviour or a function that will accomplish the same visual result
If agent is logged "in" lamp will go Green
If Agent is logges "out" lamp will go Red
Looking at the XML file I can view agent status, but is is not very elegent!
The code for the login and logout may be the same. If that is the case, then calling that code will toggle the state. Maybe that helps?
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We are getting ready to deploy PBXnSIP in out office. I just need to get a couple of questions answered so I can report back and we can finally get off the old pbx we have now.
1. I want to connect our Audiocodes gateway directly to the PBX. So the PBX server will have 2 nics. 1 that is directly connected to the gateway and the other nic goes to the network. Is this possible?
2. If the network were to go down and the PBX could not see any clients can we redirect calls to a 7 digit number such as a cell phone. I would want to do this on the domain level not on the extension.
1. Yes. The PBX can deal with any number of NIC (and IP addresses) - as long as the routing tables are set up correctly on OS level.
2. That can be done only on extension level. There is a feature called "Call forward when not registered" that you can use. Doing that on the whole domain has only limited use, as it is difficult to say when the whole domain is "offline". For hunt and agent groups you have to work with the final stage (hunt group) and all agents logged out (agent group) or just a time redirection.
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i am using 3.1.1.3110
Backup & upgrade...
what is most recommended pstn gateway?
in Trunk Setup
Posted
We have good experience with AudioCodes, Mediatrix, Parlay, Patton, Vegastream as seperate gateways. Also there is the AVM FritzBox if you want BRI. If you want to use PC cards, there is Sangoma and Dialogic that offer SIP-based cards. We try to keep http://wiki.pbxnsip.com/index.php/Interoperability_Pages up to date.