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Author Post
Thread:
dke

Replies: 1
Views: 9100

06.04.2006 18:08 Forum: Miscellaneous

It is determined by your outbound routing rules. So if you set up a rule to route all numbers to UK (say 0044....) via a particular trunk, and you specifiy the external phone number for the classic extension as 0044xxxxxxx then it will match the outbound routing rules for the trunk you send UK calls to.

DAK

Thread:
dke

Replies: 3
Views: 9341

02.04.2006 23:02 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

You may have a problem with the WAV file... only certain encodings are supported (don't know exactly what). Try setting up your recording from an extension and see if that works.

Thread: RE: ATA becomes disconnected
dke

Replies: 17
Views: 32302

24.03.2006 20:23 Forum: Terminal Equipment

No, STUN has nothing to do with voice quality. It's only purpose is to identify the external IP address/port being used if your ATA or softphone is behind a NAT firewall.

Voice quality is determined by selection of CODEC and by bandwith/congestion on the network connection between your ATA/softphone and the other party's ATA/softphone. See discussion elsewhere about audio bypass for one possible way to improve voice quality (by bypassing i-p-tel's servers).

Thread:
dke

Replies: 3
Views: 12701

24.03.2006 20:17 Forum: Terminal Equipment

firmware version is 3.1.5. Further investigation (google search) turned up several other people reporting the same problem with Sipura and STUN settings. It appears that there is a bug which causes the Sipura to think that the IP address is changed and that it has to register again immediately.

In my case, I have discovered that I can disable all STUN and NAT support in the Sipura and I still work. It seams that my NAT/Firewall (Linksys WRT54G with firmware v4.20.7) does not require it... but I did setup port forwarding to send all 5060/5061 and 16384 to 16482 ports directly on to the Sipura. This is proving the most reliable way to get my VoIP working.

DAK

Thread: RE: Classic extension
dke

Replies: 6
Views: 16035

24.03.2006 20:09 Forum: Miscellaneous

A classic extension is a regular phone number that will get routed to a trunk following whatever dial plans are setup in your outbound routing. It will cost whatever you are billed by the trunk you use. A mobile phone could be a classic extension. Classis extensions can only receive calls... they cannot make calls through i-p-tel.

A SIP extension is a userid@sipproxy-address that will be sent an INVITE following SIP protocol to create a voice connection. No costs involved. A soft phone or ATA or IP-Phone is a SIP extension. A SIP extension that is hosted by i-p-tel (user-101@pbx.i-p-tel.com) can make and receive calls. A SIP extension somewhere else (anyone@some-sip-proxy.com) can only receive calls through i-p-tel.

DAK

Thread: RE: Fax
dke

Replies: 17
Views: 53671

19.03.2006 21:37 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

Zitat:
Originally posted by supernettel
How do you set up a fax extension?
I had hoped to find that was possible, and never could find it.


Inbound routing.... My interpretation of the setting being what to do if a fax tone is detected on a given trunk. General Settings also lets you specific a destination for faxes.

But, as I posted earlier... fax tone detection does not appear to work.

DAK

Thread: RE: Fax
dke

Replies: 17
Views: 53671

18.03.2006 05:06 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I tried FAX tonight and it is not working for me either... I tried both having the PBX system detect it and e-mail me, and setting it to direct to a particular extension. Neither worked.

DAK

Thread: RE: Change CID
dke

Replies: 43
Views: 200523

16.03.2006 23:35 Forum: Providers

Try setting outbound CID for the trunk and see if that works. Also, for some carriers, I have found that the format "name" <12346> does not work. First try just passing a number (no "<>" or anything else, just a number).

DAK

Thread:
dke

Replies: 7
Views: 15882

08.03.2006 21:16 Forum: Feature Requests

Any update on when this feature will be added?

Thanks
DAK

Thread:
dke

Replies: 6
Views: 24202

07.03.2006 04:22 Forum: Bugs

We're working. At least for dialing out from PBX through my SPA-3000. I have not had a chance to try dialing into my SPA and seeing if it will connect through to the PBX.

The "dynamic" seems to be the key.

Thanks
DAK.

Thread:
dke

Replies: 9
Views: 28923

06.03.2006 21:04 Forum: News

I should add that while the RTP destination for audio is changing, the codec does not. Even though I have a preference for G729, the connection remains at G711.

DAK

Thread:
dke

Replies: 9
Views: 28923

06.03.2006 18:36 Forum: News

Audio bypass seams to be working for me at least on outbound calls (I have not tried inbound). You have to set audio bypass to yes on both the extension and the trunk that is being used.

I have tested calls through my SPA-2002 and with audio bypass on the latency does appear to be much better. Also, the syslog trace from the SPA-2002 does show a rather cryptic line suggesting that it received a new RTP destination/port during call setup.

NAT setup may also affect this. My SPA-2002 is behind a Linksys WRT45G router. On the SPA-2002 all stun/nat settings are disabled. On the WRT54G ports 5060/1 and RTP ports 16384-16482 are forwarded to the SPA-2002. I have had numerous problems trying to enable stun/nat settings on the SPA-2002 and have found that it works fine without them.

DAK

Thread: RE: Reaching an extension from outside
dke

Replies: 30
Views: 86996

Working 04.03.2006 03:59 Forum: Miscellaneous

See thread in "Bugs" section. Fixing my problem with the SPA-3000 has also fixed this problem... you need to enter a SIP Server of 217.195.32.11 on the trunk. Trunk has to be named <userid>-something then you can SIP Dial into SIP/<userid>-something@pbx.i-p-tel.com

Thanks to Pascal for identifying the solution.

DAK

Thread: Working
dke

Replies: 10
Views: 28184

Working 04.03.2006 03:55 Forum: Miscellaneous

See thread in "Bugs" section. Fixing my problem with the SPA-3000 has also fixed this problem... you need to enter a SIP Server of 217.195.32.11 on the trunk. Trunk has to be named <userid>-something then you can SIP Dial into SIP/<userid>-something@pbx.i-p-tel.com

Thanks to Pascal for identifying the solution.

DAK

Thread:
dke

Replies: 6
Views: 24202

registers have stopped 03.03.2006 16:01 Forum: Bugs

Thank you, REGISTERs have now stopped.

Yes, the PBX will accept REGISTERS from my SPA-3000 as extensions. That is working. However, I want to use the SPA-3000 as an outbound trunk and route local calls through it. How can I do that? There is no way to set up an outbound route with an "extension" as the destination, nor is there a way to setup a trunk with an extension as the desitination.

The SPA-3000 has both an FXS and FXO port. Asterisk can use the FXS port as an extension and the FXO port as a trunk. The VoIP interface for the FXO port can "register" with a PBX which allows for PSTN to VoIP gateway function (this is working with i.p-tel). I need VoIP to PSTN gateway also, which requires a way to route outbound calls to it. This is what I am trying to setup in i.p-tel by creating a trunk for the FXO port.

Thanks
DAK

Thread:
dke

Replies: 6
Views: 24202

Infinite loop registering to SPA3000 Voip-PSTN Gateway 03.03.2006 00:13 Forum: Bugs

I'm trying to setup a Sipura SPA3000 to use as a Voip to PSTN gateway for local calls. I think I have found a bug in your trunk registration process.

I setup a trunk to connect to the VoIP gateway on the SPA3000. pbx.i-p-tel.com sends a REGISTER request to the SPA3000, but this is not supported and a '501 Not Implemented" message is returned. Attempts to make any call through this route get a "all circuits are busy now"

Meanwhile the PBX continues to attempt to REGISTER, and The Sipura continues to reply "not implemented". EVEN IF I DELETE THE TRUNK, the PBX continues to attempt to register. I tried to add another trunk with different userid... now it too is in an infinite loop attempting to register with my SPA3000. looks to me like the trunk initialization is looping and in this state is not even able to delete itself.

It should be possible to setup a trunk that does not need to REGISTER. Authentication can take place after the INVITE.

Thread: Trunk provided voicemail
dke

Replies: 0
Views: 6298

Trunk provided voicemail 02.03.2006 17:55 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

A question... what happens if a trunk has it's own voicemail system and a message is left there? A SIP NOTIFY message will get sent from the trunk to i-p-tel PBX indicating that there are messages waiting. But what will i-p-tel do with that message?

I've not tried this but I'm guessing that the message will be ignored. Can anyone confirm?

The more interesting question is what should/could i-p-tel do if it receives that message? Maybe an extension could be setup with "trunk provided voicemail" option so that if i-p-tel receive a message waiting NOTIFY from a trunk, it would pass that on to a designated extension?

DAK

Thread: RE: IAX Protocol support for trunks
dke

Replies: 11
Views: 49273

02.03.2006 02:49 Forum: Feature Requests

Well, so far Voipjet.com is the only provider I know of that is IAX only and does not support SIP. So I guess that makes it low priority.

DAK

Thread:
dke

Replies: 4
Views: 23820

27.02.2006 04:24 Forum: Bugs

Voice Message Waiting indicator is now working for me... Sipura 2002 receives the NOTIFY message and recognizes that there are messages in voice mail box. The Sipura is then sending the correct signal to my phone which flashes the message waiting light. Similarly, once I delete the messages the light goes out.

I have not tested new/old messages to see what happens with this. But I'm happy now that new messages are being correctly NOTIFY'd.

By-the-way... the "play next" setting is still not working. I still have to press 6 after deleting a message for it to play next message.

Thanks
DAK

Thread:
dke

Replies: 4
Views: 23820

27.02.2006 04:19 Forum: Bugs

You might also check out... http://www.voip-info.org/wiki/view/Asterisk+phone+snom

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