Vodia PBX
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I think you can copy and rename the aa_busy_callback.wav and use that one.
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ACK. Needs to be fixed before we can release this.
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Well, so far the answer is dialog state. There are a couple of phones that support this standard and that kind of works, for example subscribing to the dialog state of CO-lines.
But it is not such a good answer if you want to have something on your PC. We are thinking about AJAX, or maybe Flash or just running a native Windows application.
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The call list is just another table in the PBX. So in theory, you could read that table using SOAP. But I am not sure if that is a pragmatic way to go.
Alternatives:
1. Register a dummy user-agent and subscribe for the dialog information.
2. 2.1 will introduce "buttons", which are instant messages that carry call information. You still need to use SIP for that, but it is much easier than dialog information.
3. You can also subscribe for dialog information using HTTP. Then the PBX will push that out via a HTTP request.
Everything not very appealing...
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Well, that has been addressed in 2.1. In the meantime you will have to live with escape characters and complete extended regular expressions...
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Set the outbound proxy of that phone to the PBX (e.g. the IP address) and set the domain name of the registration to the name of the 2nd domain.
If you want to push it, don't use the name localhost in any domain.
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Most people just trunk into Asterisk. Just use the gateway mode and you should be all set.
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Last week we upgraded to 2.0.3 Build 1707 (Win32).
Why 2.0.3.1707? There is 2.0.3.1715 which is the last release and this image likely fixed that DoS issue (I remember that intermediate build had a problem).
There is a global setting called "max_udp_invite" which defaults to 10. If you have more than 10 new calls per second, you might need to increase that value.
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Well, the firmware files that you get for the Polycom phones. It is a little bit difficult to get, you need to talk to your reseller if you want to go the official way. But I think it is also possible to find it on the Internet.
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I guess the problem is that the calls from the Exchange trunk have to run through a dial plan. Maybe you can add "Call Extension" to the dial plan?
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Well, the 2098 version should still change the SSRC. But for an obvious reason we don't "support" that any more...
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The Explicit Remote Party ID now is just a straight, simple DID number. I you leave it empty, it automatically takes the old content "$f $a PREFIX$u DID". That means if you define the first alias it will present it.
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Ehem, that should work on most cases already? You can even back up if the extension is ringing already. Try pressing the star code in the various locations.
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Well you should be able to have a pattern that matches the 0 and then redirect you to another account... Would that solve the problem?
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BTW there is also the option for AudioCodes cards that have a SIP API.
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Well, you need to put the various *.ld files and the sip.ver files into the tftp directory, but *not* all these other template files (maybe the SoundPointIPWelcome.wav if you like).
"Worked for me"...
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Okay, here we go with 2.1.0.2100: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2100.exe.
Notes: There were issues with one way audio associated with the new RTP-pass through mode. The last bug was when the ROC in SRTP kicked in again, resulting in a kind of one-way audio. The version also now includes a JavaScript password checker that keeps users from entering too simple passwords.
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Yepp, that is possible. You must have the recording feature enabled, though (see http://wiki.pbxnsip.com/index.php/Extension#Redirection, well a little bit hidden link). Then you can turn on recording for different kind of calls, see http://wiki.pbxnsip.com/index.php/Recording.
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Well, it is allowed. The IETF's opinion about music on hold is that the music comes from another server, and that server uses a different SSRC, timestamp and packet number.
But it seems to be a challenge for compability of some devices. Maybe we have to make it possible that these parameters do not change and avoid the problems.
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Well, usually you should get a call waiting if the caller is also part of the paging group. There are some phones that audomatically put a call on hold once that they receive an INVITE with an auto-answer request (which is IMHO very problematic, think about you most important customer on the phone and someone paging).
If you have such a phone, try setting the "lines" parameter in the registration and set it to "1". Then the PBX will not call that extension.
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This will be in 2.1, including the firmware provisioning (16 MB and more). It should be working in the latest RC (make sure that you don't have your own pnp.xml file).
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If you have just one domain, just call it "localhost" and the PBX will happily accept any domain name in SIP packets.
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Ouch. That was an "old friend". The problem is that the internal table for mixing audio is quite short. Thanks for the tip!
We made a quick hot-fix image at http://www.pbxnsip.com/download/pbxctrl-2.1.0.2098.exe. Please do a quick verification (we could not do it here yet, so make sure you can move back easily).
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Okay, what about this:
The PBX keeps a list of the agents. Everytime that an agent picks up the phone that agents moves back to the list:
Initial: 1 2 3 4 5
Call 1: ring 1, ring 2, ring 3, 3 picks up -> 1 2 4 5 3
Call 2: ring 1, ring 2, ring 4, 2 picks up -> 1 4 5 3 2
Call 3: ring 1, ring 4, 1 picks up -> 4 5 3 2 1
And so on? That should be easy to implement.
Active Calls
in SOAP
Posted
1. First, you need to subscribe to dialog state. Through http that is difficult, but if you can do a SIP SUBSCRIBE that should be easy. Just put a contact like "Contact: <http://192.168.1.2/dialog_response.htm>" there.
Through HTTP is a little bit difficult at this time, you would have to do this trough SOAP. However, we will quickly add a type field to the web request, so that a regular http post can do the job (in the registrations tab, there is something at the bottom that adds a registration).
If you just want to test it, just create a static registration without a type, and edit the XML record in the file system. After that, you need to restart the service though to read the change.
2. The notifications then are walking out as http post requests to the provided location. They contain the same body as the SIP NOTIFY in a SIP SUBSCRIBE dialog would do.