Vodia PBX
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We currently are doing an off board solution which provisions all the parameters we require, however, it is off board and is not a clean process. Can the attached template be modified by integrators such as The VoIP Highway. If parameters are added to the PnP file, does this require an upgrade?? Since we are a reseller of pbxnsip, we like to customize each application to suit the user.
All you have to do is put the above file into the html directory. Then you can also make local changes there. Only if you add a PnP parameter you need to restart the system, otherwise the file drop will already do the job.
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im not understanding what you are saying? if the problem is tracking the email then it shouldnt allow to send out the email twice. and if it deletes the file from the system then how come the file is still in the mailbox? why cant we just send the attachment twice?
also even if i had to do the workaround how would i create an email list in the system?
You mean if you keep the WAV in the system then it also does not work? IMHO it should only be a problem if you select that the message is deleted after the email has been sent.
The idea about the exploder is related to your email service. If you can set up a group email account then the email server will distribute the email to the group members and the PBX has to send only one account.
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any update on this request? thx
There was probably another problem with empty user names. If you can make the gateway use "anonymous@..." it would of course solve the problem. We'll try to incorporate a fix in the 3.1.1 version. If you want a preview, let me know what OS you have.
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While auto provisioning Linksys products, a file "spa_phone.txt" is generated in the Generated/{mac} directory. If we place this file in the html directory, the phone will receive this file but it does not get any of the parameters required for provisioning. How can this be accomplished with the linksys product or better yet, how can we affect change on the parameters generated in the "generated/{mac}/spa_phone.txt" file, that would be the ideal scenario. Specifically, we are looking at passwords and dial plans.
Well, currently we are using the attached template. Ideally, if you have a improvement proposal we just add it there. We can also add PnP parameters to make the file a little bit more flexible.
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1. in list of accounts, all extension, aa, etc... are now shown as 1-00, 1-01, 8-00 etc... with the strange behaviour I can't change the extension, invalid input. But also the aliasses are done with the -, for e.g. ramond is now r-amond (9-99) (as example, ramond has 999 as extension)
What did you put in as country code? Seems the beautification tried to be tooo nice...
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G.711 needs 64 kbit/s, and RTP adds an overhead of 24 kbit if you have 20 ms packets. That is 88 kbit/s per channel, so 176 for both channels. Add some extra for the SIP packets and then 192 is pretty realistic.
If you choose different codecs the picture changes. G.711 is a "worst case" scenario.
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Anyone else seeing trunk issues using Teliax?
They have had some ups and downs for the last month. I move to another Trunk and everything is fine. That list of OK trunks started to dwindle. They explained they were just doing some updates. Today they told me they did the final upgrade on the last lax.teliax.net trunk I was using. Since then I cannot connect. They have said several Asterisks users who had to make some configuration changes.
I currently get a 481 Call Does Not Exist (Registration failed, retry after 60 seconds). I have the information correct. The selected server is correct on the dashboard provided by Teliax.
Anyone else figured out what to change? They can give me nothing more then invalid user ID.
Any ideas?
[...]
[9] 20081119145322: SIP Rx udp:74.201.8.23:5060:
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 75.149.200.210:5060;branch=z9hG4bK-b18664ba5bb745a6d3828102d36620bb;rport=5060
From: "brichter" <sip:brichter@lax.teliax.net>;tag=26524
To: "brichter" <sip:brichter@lax.teliax.net>;tag=vyegX37aDjQHc
Call-ID: ttagjj3g@pbx
CSeq: 125 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
[5] 20081119145322: Registration on trunk 9 (Teliax 1) failed. Retry in 60 secon
Hmm, not sure why FreeSWITCH replies with with a "call" when trying to register. But that looks like a bug to me. Or maybe it is something trivial like that user really does not exist.
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It is the default of "Entering the agent group". What should I set it to so that the caller hears music on hold indefinitely until and agent answers the phone, even if it takes 30 min for the agent to answer? If it matters I also have "Ringback tone" set to "No ringback tone, continue playing music".
Okay, that setting is fine. Of course it is possible to have callers in the queue for 2 minutes (unfortunately for them). 2 minutes just sound like there is a problem on the SIP level. Is there anything in the log where the PBX explains why it disconnects the call? Are you using redirections with the value of two minutes?
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the system will allow you to send your voice mail to multiple email addresses HOWEVER, it will only send the attachment to one of them.
why is this and how do i fix it???
The problem is that we delete the file after the first email is out. It is hard to keep track of the "last" email. If we send just one email you can see who else receives the voicemail; though this seems to me the best way to solve this problem.
Workaround is to use a email-list, so that the email server "explodes" the email to the various recipients.
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New to this forum and I just purchased a cs-425 for a SIP application. The unit is currently running version 3.0.1.3023. I noticed an uprade is available to 3.1. Are there any enhancements to this upgrade that make it worthwhile at this time?
If you don't have to, don't upgrade. Release notes for 3.1 are here: http://wiki.pbxnsip.com/index.php/Release_Notes_3.1. Maybe you wait for 3.1.1; then you will probably have no upgrade problem.
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Whatever I set the gap time to the the first message plays in half the time of what it should instead of waiting the full gap time. Perhaps this is the planned behavior by the developers but neither the web interface or wiki say that it should do this on the first time playing the *1 message. After the first message is played the gap time is honored by the system at its full value.
That's a feature. You can also spefify the initial waiting time, just updated http://wiki.pbxnsip.com/index.php/Agent_Group.
Also, after ~120 seconds the call drops and hangs up on the caller. This happens on 100% of the calls into the agent group.Are those calls technically connected (on SIP level, did the PBX send "200 Ok")? Otherwise, this is just the regular ringing timeout. Calls generally cannot stay unconnected for a too long time. What is the "Event for connecting the call"?
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OK thank you. But when is your tentative timeline for the version 3.1.1?
You can have a preview version already
Currently at http://pbxnsip.com/protect/pbxctrl-3.1.1.3091.exe. -
I'm looking to record custom prompts throughout pbxnsip and was wondering if there was a script of words and phrases available?
I've done a brief search of the documentation, google, the wiki and the forums and can't find anything
If someone could point me in the right direction that be ace
Yes, there is - send a email to support and then you get it.
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Additional issue .. in the RC for some reason VM messages are not recording (that is why is need the logging working) and it seems related to this situation and not "normal" extenstions.
Caller comes into AA then presses 6 which transfers them to an extenstion with NO registrations, when the caller is connected the VM (goes right to VM as it is unregisterd .. lets call this a VM only ext) the prompts and recordings are playing and the user is leaving a VM but for some odd reason unless the caller presses # or a digit it will NOT leave the VM which is very odd as usually a call disconnect works just as well and is certanly not expected!
PLEASE check this out.
Yea, this is really strange (was also reported by other people). If you like, try http://pbxnsip.com/protect/pbxctrl-3.1.1.3091.exe this may fix the problem.
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Just to be sure that I understand correctly, it will be possible in the future to have more than one number attached with your Extension for call forking?
For a temporary fix for accessing the DISA on the system while extending the full call forking functionality, would it be possible to list have a field like DISA Authorized Numbers where the user could list their home or home office numbers in the system?
Even the temporary fix is not so easy as we need to split the cell phone entry up into a own table. So I would vote for doing it right and skipping a fix.
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Yes, the 2nd Caller on the queue just remain on the queue and just listening to music, it doesn't advanced to the agents, UNLESS the 1st Caller hang-up, then both agents will ring.
Please confirm when can we realize this solution.
Yea, that was a good one. This fix will be included in 3.1.1.
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is there anyway to do this when using a sip trunk? like a call hunt to outside numbers? like home lines/cell phones?
We checked this feature - in principle that's possible. The reason why we did not have this yet is that the trunk requires a failover context, and for a calls we currently have only one. It is not a huge problem; it is that it just needs to be done and we want to make sure we don't create any side effects...
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Can you help me with the logic in only allowing 3 trunks on the CS425 as this is an outstanding solution for places with multiple offices, in our case 1 with a 100 user PRO version and then smaller offices with CS425 solutions and for site to site dialing it really does not make sense to limit to three trunks for that .. can you maybe add a S2S trunk that is unlimited or at least could be licensed for more than 3?
The intention is that this box is not being used as a PBX for a large organization and usually three trunks should be really enough (PSTN, ITSP, HQ trunk). However, I believe that if you send a friendly email to sales they won't be too bitchy about it and give you a upgrade.
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Okay on the transformations .. we have the dialplan set as NAPNA 3 digit and when entering an extension as:
** Inbound DNIS **
In v3.0 we had setup extensions as 123 with a tel:40612345678 alias and after conversion to 3.1 it added the 011 to all of these so in the ext it showed 123 011 40 61234567 or something similar?
v3.1 - when we have an extension same as the areacode it tried adding +1 to it?
v3.1 - 123 4061234567 it changes to 123 (406)123-4567 on the screus-en and then in the user alias db entry it stores
445.xml:<row><display>(406)123-4567</display><domain>2</domain><name>+140612345678</name><user>35</user></row>
We took care of the issue with the inbound by adding that to the incoming number as it was coming in as 10 digits so we prepended in the inbound trunk +1 to those .. should that be required just to a version upgrade?
Before: !([0-9]{10})!\1!t!500 .. for 2.x 3.0
After: !([0-9]{10})!+1\1!t!500 for 3.1
** Logging **
Also, we are trying enable logging which full debug and it just ignores me and does not do so, no file no log entries in the status->log
** Outbound **
Why does it still transform the OUTBOUND ANI to a full number
Did you use "1" as country code? Then those transformations IMHO make sense. Okay, the upgrade procedure is a little bit bumpy, but then we have a clean solution.
The whole magic is disabled when there is no country code. But IMHO it makes sense to specify the country code and let the PBX convert the different telephone number formats into a normalized format.
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can you upload the current snom dialplan files? I cant for the life of me get it to work.
I have to press the CHECK MARK every time..
Version 3.1 will have it's seperate PnP dialplan in the domain. Maybe that solves this problem. The attached dialplan is the current state.
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Are there any plans to integrate the snom presence status with pbxnsip?
and What file do i edit for control the PnP softkeys?
Presence is a topic that sounds nice on the datasheet, but I never saw real people in real life using something else than MSN, Yahoo or Skype. Believe it or not, but the PBX has a presence agent. Nobody knows about it or even uses it.
The problem is that publishing presence from a phone is just too inconvient. Once you have a PC, it is just so much easier and colorful to use MSN & Co.
Just my two cents!
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When i send a text message from the web page to a snom phone, how do i delete it from the screen?
I believe you have to send an empty message...
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do you have to enable a button for the CMC during the call (snom)?
Yes. There is a settings for this in the phone.
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right which is normally does. However this is the only scenerio it doesnt work.
set your vm limit to 5 messages and set the voicemail to email you and mark as read. now leave 6 messages the voicemail no matter what give indication that there is 1 message still. call the voicemail and do nothing and it updates the phone.
same think scenerio works with polycom, xlite, snom so it has to be the PBX not updating the phone that the message was maked as read and thus not a new message.
Yea, the problem is that the PBX first sends the MWI, then sends the email and then marks the message as "saved". We will change the sequence so that the PBX will first send the email, mark the message as saved and then send the MWI. That should fix the problem.
Currently at 
Hunt Group - Multiple Service Flags
in Hunt Group Setup
Posted
It should be working. Did you also put a second destination there? Every Service Flag Account must correspond to a Night Service Number.