Vodia PBX
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Posts posted by Vodia PBX
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I programmed my system in order to dial directly to our customers, I need to eliminate "1" confirmation dialing, Do you know what I need to add in the following sample?
What version are you on? I remember we took the "1" out...
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Nope. We have a domain.local (alias) and a server.domain.local.
Well, then that's the problem: REGISTER sip:192.168.41.223:5060 SIP/2.0. "domain.local" != "192.168.41.223" and "server.domain.local" != "192.168.41.223". Try "localhost" as alias.
The string "localhost" is a magic name that matches any domain name: "localhost" == "192.168.41.223"!
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We are using SNOM Phones...320's and 360's
Maybe you can just use the speed dial mode for a button. Not sure if you then press hold on the same button then.
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In the 3.0 release we (will) have a ANI settings that should solve many of these problems.
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Very strange. Do you have an alias name of "localhost" for the domain?
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Yes
Well, then it will be very difficult - the PBX knows only at the end of the DTMF sequence that the code is supposed to be intercepted. I would prefer to have a option on the PAC that turns recording on and off from a user.
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Will there any any problem/limit by doing this? How many calls can the PBXNSIP IVR handle simultaneously?
Apart from the usual resource limitations (number of calls etc), the number of pending transactions is limited to "1". The PBX will send out one SOAP request and wait for the answer. That means the processing of the key input will depend on the speed of the database. But I believe unless you have a absolutely crazy database the response times should be still well below one second, which should be okay for a caller.
http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node shows how to do this with PHP, I guess you have already seen it. The mySQL part is missing, though. But there is plenty of resources available on this topic.
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It would need to be automatic since this is an afterhours emergency notification mailbox
I would solve this problem by a email-to-SMS mechanism. Most cell phones even support email, which would make things even easier - just attach the voicemail and then those support guys can listen to the VM in alomost real-time in a very convenient way.
The problem with calling the cellphones at the same time would be that practically most installations would run out of PSTN channels when the PBX starts "blasting" out calls. In order to solve this, those calls would have to serialized, and that is not so easy.
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I am a newbie to pbxnsip, so please point me in the right direction if I have posted in the incorrect location: I have a client that requires calls coming into their main switchboard number to ring on the Receptionist desk for 20 seconds, then on the office administrators desk for 10 seconds and then to loop back to reception, then to office admin again etc. They are adamant that they do not want an option of voice mail during business hours as there is "no situation that would arise that cause the phone not to be answered". I have tried to create a hunt group that has the two stages required and a final stage that calls itself again but have not been able to get it to work. Can anyone give me some pointers as to how to go about this one.
In the evenings they would like the call to go to a message that informs the caller that the office is closed and that then requests that they press a digit to leave a message (in a shared mail box) and hangs up if they do not press the required digit. I guess that in this case the night number would be a second auto attendant or an IVR triggered by appropriate service flag?
I would solve that problem by setting a very long timeout (like 999 seconds) on stage 3. There you can list all extensions that should ring "forever".
Looping is not a good idea (though it should be possible!), the PBX has a hard time detecting and fighting such loops. Imagine that all extensions are not registered, and then the stage duration is very short (like 0 seconds), and then such a loop will be an "endless" loop. After a reboot, typically all extensions are not registered for a few seconds, and then if a call comes in you have an endless loop.
The night mode should be straight-forward. Don't forget to put a 8 in front of the mailbox number, so that it does not ring the extension first.
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We have nearly 400 users on PBXnSIP each with about 4 aliases, which makes it impossible to find a particular record through the web interface, as one needs to read through each screen, then click the next range, and when you end up on 851, and you accidentally change something, you are back the the first screen and the manual screen-reading search starts again.
Please could you add a search function on the accounts screen where one can put in the the name or part of the number and then return the search results as opposed to read through each screen manually.
We solve that problem by changing the the index. Instead of "1-25 26-50 51-75" and so on we'll put the actual account there, so that you can immediately click on the right link, for example "401-425 426-456 457-co9".
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Will also be fixed in 2.1.11... It was indeed pretty "open".
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Well, the service provider obviously uses software that discards the line parameter in the URI. This is not RFC compliant, and that is because of the problem we are facting here - someone (the PBX) registeres several contacts and when the call comes it it needs to know where exactly it should go.
The workaround it to send the call to a tel:alias. But this is not very reliable and the PBX needs to perform a table scan on the trunks to locate the right trunk. If you have a lot of trunks that will take some time, and if you have a lot of calls the CPU will start choking.
Maybe you can tell the service provider they need to fix this problem. The RFC was released in 2001 and it is time to support this.
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Does anyone know of a way to do a cell phone notification to multiple different cell phones? I have a customer that would like to be able to have multiple techs notified if a message is left on their after hours emercency mailbox. I have tried to do a huntgroup of cell phones, however that did not work.
What you could do is copy the message to different mailboxes. Check out http://wiki.pbxnsip.com/index.php/Mailbox, there you can see how to copy a message. Though this is not done automatically, someone must manually do the copy.
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I'm having the same problem and I'm going nuts trying to fix it.
What phone? Did you go through the Wiki? If there is something missing on the Wiki we should update it, so that it works right from the beginning.
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Please let me know if that's doable.
Maybe we should make this a feature. Now that we have a file-system based spool directory and a thread that can take forever to prepare the email it would be possible.
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How does one turn off the "beep" tone that is heard when initiating the call recording feature?
You mean the DTMF done coming from the phone?
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I have experienced the same thing - it's a bug.
ACK. Another fix for 2.1.11.
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Yes, Linux in a quadcore server. What worried me is the "failed" part.
Anyway, is there any way to avoid that shifting around procesors and introduced jitter? (of course taking advantage of multicore for a higher number of calls capacity)
Well that is the purpose of the affinity... Worst case is that the Linux distribution is not supporting it, then there is not way to make sure that the core does not shift processes around.
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We Have a ValCom paging Adapter 9970 and then going to a ValCom 9964 Antifeed back eliminator for our paging system.
This is connected to the PBXnSIP server via an Audio Codes Analog to digital converter. In the phone system the page is executed by dialing extension 39 and then it rings through to the ValCom and then over the paging Speakers. We had a few issues early on, like call waiting and all that coming over the paging system as well as an irritating busy signal occasionally....but that has been fixed for a few months now. I do however have another issue that is mostly user error but I want to know my options on how I can minimize the effects. I have a short cut programmed for the paging button....the person pushes it, hears it ring once, then a short tone and they speak and hang up....1 second later it broad casts.. the issue is when they hit the button and hit it again...It puts the extension on hold, and plays hold music over the PA...is there a way to disable that feature just for an extensio? or is there a better way to set up the page button so this cannot happen?
On the PBX, it is not possible right now... Any chance to do that on the phone? What phone are you using?
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When *90 is removed (INTERCOM) from the Domain Feature Codes Settings -
*00 (the next one in the list) - ceases to function.
If I place any other * code in the INTERCOM section (i.e. *01) - *00 works again.
Please let me know if you can reproduce and fix.
Whow, you are right! Will be fixed in 2.1.11.
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Could be... As long as it works we are happy already. The PBX uses a library to access the dongle, and that library is probably a little bit too pragmatic.
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Does pbxnsip support G 723.1 codec?
No. Better use G.729A - just 2 kbit more, much better quality and much less CPU intensive. BTW the RTP header overhead is already in the 12-24 kbit/s range, compressing to 5.3 kbit/s is completely pointless IMHO.
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Can we have a feature code and password to block the outgoing calls of the user? Each user will dial the feature code and enter the password to lock/unlock the phone.
Just set the standard dial plan in the domain to something restrictive. Then when callers want to place an outbound call, they need to use the "calling-card" account and use their extension and PIN code before they can place a call.
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Come across a really anoying problem.
Seems like since I enabled the SIP Replacement field (to enable users to plug snom on internet and make calls) if someone from outside phones in, and then hang up, the phone on pbxnsip still keeps ringing. If I dont pick it up, it goes to voicemail, and I get lots of silent voicemails.
The only way to get fix it, is to remove the SIP Replacement field, reboot server (Windows).... put info back, reboot. It then works fine for few haours, maybe a day. Then same problem.
Help would be appreciated.
That sounds like a serious routing problem. The really really best way is to have a fixed IP address and transparently route it to the PBX (no NAT), then you will be able to have a stable operation. Next on the preference list would be DMZ NAT with a SIP replacement (tricky, but possible to have it working in a stable way). Stuff like STUN etc is so crazy instable that it is a waste of time and support money.
Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses, if you did not do that yet.
Issue with outbound from (caller-id)
in Trunk Setup
Posted
In 2.1.10 you can do that by using the tel:alias or a per-account basis, this has higher priority than the DID number setting in the trunk. This solution is okay unless you have two extensions that should both show the same DID which is not the same as the DID for the trunk (that will be fixed in 3.0).