Vodia PBX
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does it possible to use account parameters from a trunk connection as caller id ? cause i need that my ip pbx use sip:account@ipness.com and not sip:username@ipness.com
You can try to set the ANI for the trunk and choose the mode "No Indication" for Remote Party/Privacy Indication.
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Customer's Requirement:
If one of the Member of a Hunt Group is busy, the Hunt Group should be busy.
Situation:
Small company. Only one staff at the moment available, walking around the business area. On external calls, some extensions should ring (office, warehouse ...). This is done with setting up an agent group forwarding the call to a hunt group (not all extensions are ringing immediately, but in different stages). If a call is answered and a second call comes in, hunt group should be busy until the first call is finished and (this) next call is forwarded out of the agent group queue to the hunt group. If the hunt group wouldn't get busy in this situation, 2nd caller would ring on all the other remaining phones and could get the impression no one is available, because no one answers the ringing phone. Even if I set "caller gets MOH until call is answered" the staff gets crazy if all the other phones are ringing while talking.
Insufficient solutions:
Registering all these phones on the same extension. Problem: All extensions would ring immediately at the same time (warehouse phones should ring delayed, so in normal situation office staff pics the call first before they begin to ring).
Hmm. What if we just introduce a setting that says "maximum number of connected calls" in a agent group? In the above example, set it to "1" and then that poor guy who is on duty can work without phones ringing around him.
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Thanks for the fast reply. Sorry, this was a type-mismatch. I didn't changed the port 69, but have restarted the PBX service several times, but still no answer on telnet 69.
TFTP is UDP, telnet is TCP. Check out "netstat" from the command line to see what ports the PBX uses.
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if you think what that audiocodes parameter is let me know...that is likely the problem...because only calls thru audiocodes does it...
(this problem doesn't cause us a lot of pain...but a client would not put up with it.)
Do you get a admin email about the disconnect? That would be a clear sign that the PBX thinks the call dropped. Also, check if the AudioCodes has VAD turned on (turn it off if that should be on).
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If I set co line limits to a trunk it will not kick back to the dialplan for the next route.
example. Trunk ABC is limited to 7 calls but on the 8th call pbxnsip does nothing but send a 404 back to the phone.. when in theory it should go back to the dialplan and try the next route.
Full CO-lines are not triggering the failover... Only a full PSTN gateway can trigger that trough a SIP response (ideally, a 5xx code).
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How can we shut this off so it just dials the number and doesn't send the email saying remote call initiation. Or where is the security setting it is referring to.
It sends an email? I would not know about that nor how to turn this off.
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prior to migrating the phone to sip. The time and date showed at the top off the phone. IT doesn't any more. Anyone know how to get it back?
Which config file are you using to get the phone up and running? What firmware version?
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I'm new to pbxnsip. It is up and running on my Linux Centos 5 server. I can access the localhost. Now I want to access pbxnsip from another computer. Can anyone tell me what to do. Do I have to add something after the IP-Adres or do I have to grant permission on in Centos?
Well, you need to know the IP address of the PBX server. Check out http://wiki.pbxnsip.com/index.php/Login for more information.
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I have the same experience with CallCenter edition
was installed with v 3.1.2 and manually upgraded to 3.2.0
Don't see that email setting
Guys, this is not in 3.2. It will be in a 3.3 build, which is not "officially" available yet.
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Does anybody know how to enable SRTP?. It should work with Snom phones.
TLS works but would like to have the media encrypted too.
If you are using TLS (which is the default) then you automatically get SRTP. Check if you see a "crypto" header in the INVITE/SDP.
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Can you add a Blaser or even predictive dialer. Blaster where we can import phone numbers and send out the calls to those phone numbers and when they pick or voicemail picksup it send a wav file message about new products or any news.
There is such a software available I believe from Sangoma. Thanks to the SIP standard it can be used with the PBX.
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[5] 2009/02/10 13:08:23: UDP: recvfrom returns EAGAIN
[5] 2009/02/10 13:34:34: Last message repeated 8 times
Eight in thirty minutes is no reason for concern. Maybe the cable is a little bit instable. But the PBX will not drop a call because of that.
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It would be nice if pbxnsip had a field to specifiy if the phone is internal or external to properly build the PNP tftp files...
Nono. The PBX must find that out automatically, and it is possible. The routing table of the PBX make that possible.
I believe in this current case there must be something wrong with the setup. We are trying to get a login and find out what the problem is.
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i had a server fail, and am in the process of manually moving the files i want from the old server to the new. I've found wav files, but really have no idea how the pbx is associating these with the individual accounts. Can someone tell me, besideds just dropping the wav files into the recording folder, or dated sub folder, how to link them to the same account as before?
The filenames are in the XML files in the messages directory. You can "grep" for the names if you want to know where they are referenced.
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what does this mean: UDP: recvfrom returns EAGAIN
That means the PBX wanted to get a UDP packet from the OS, but that packet was not available any more. It can happen, but it should be rare.
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[2] 2009/02/10 17:13:22: SIP Rx udp:82.146.119.38:6060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-9d0d48c8b3ed85de66e34b297e320015;rport
From: <sip:tilleul@ipness.net:6060>;tag=31120
To: <sip:069665262@ipness.net:6060;user=phone>;tag=GR52RWG346-34
Call-ID: d399edac@pbx
CSeq: 1885 INVITE
Contact: "Verso CM" <sip:82.146.119.38:6060>
Allow-Events: refer
User-Agent: pbxnsip-PBX/3.2.0.3144
Content-Length: 0
Hard to say why the provider sends forbidden back. Are you sure you paid your bills?
Interestingly, they seem to run pbxnsip as well. They even use the same version as you do!
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Yes, all my parameters are set correctly. As Norman mentioned, a #9 to accept the message works. Just hanging up causes the scenario described earlier.
May be a stupid question: Do you actually see the BYE message showing up on the PBX? Do you see it in the log file? Maybe the PBX is not aware the call disconnected yet.
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Just curious how others test to make sure the 911 dial plan and emails and all work like they should?
(I don't want to be calling 911)
The last time I saw a VAR dialling 911 and talking to the officer saying "Hi, we're just installing a new PBX and testing if it works. Have a nice day." No firetrucks showing up in front of the building.
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Use cell_dis. Other parameters are driven by this.
"cell_dis" means "Display" and it can contain strings like "(978)746-2777". The cell phone number in "cell" is being used for making calls and it would typically contain "+19787462777" (if you use country code 1).
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I just got back from the PBXnSIP training and it was mentioned that the 3.x branch supports video in a limited mannor. I was trying the Counterpath XPro client but am not having luck establishing call or adding video to existing call.
From my other vendor trainings this is a design problem of the PBX. When using a B2BUA architecture the phone system needs to know what each option is in the SIP and RTP streams. If it does not know it drops the info from the relayed stream.
Try establishing an audio call first then switch the video on (while still talking). Essentially the PBX believes it is T.38. Fax and video are actually similar.
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I too am having the same issue. I've checked permissions on the "recordings" folder and everything is fine. I actually see the .msg files that the log reports as being deleted. I've checked the mailbox settings and I'm not confirgured to send messages via email. When you call into the mailbox, the system reports no new messages.
I still can't figure it out. It is hard to believe that would be a CentOS-specific problem. Last resort would be to make a strace of the interaction between the PBX and the kernel.
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is it in the pbx.xml or the GUI?
This will be in the GUI.
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I am having some issues getting this to work properly. After some investigation, I found out that when I send an ANI that does not match the account number (username), the PBXnSIP server that the trunk is registered with returns a 401 error, however if I set the ANI to match the account number, then the outbound call works properly. The initial registration, and receiving calls is never an issue.
Did you try to list all possible ANI in the extension? I believe that the PBX will then try to stick to the ANI provided from the user-agent. For example:
ANI: 9787462777 9787462778 9787462779
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I am having an issue with Voice mail. Everything seems to work properly. The Log shows M messge next to calls with voice mail, but no voice mail ever saved or delivered.
when I enable logging
i see the following message
[5] 2009/02/06 22:42:49: Message file recordings/msg42.wav was removed, removing associated message
After leaving the message, does a new .msg file show up in the recordings directory? Maybe it is write protected?
6 Minute Sample margins of error
in General Setup
Posted
No, the blue line is the number of call legs. Usually one call equals two call legs. The red line is the number of calls, and in that graph the maximum was three calls.