I am configuring a OmniPCX Enterprise (appliance server w/ MG small GD MADA3) with a SIP Trunk from vitelity.com (which works). I am able to dial in and out of the sip trunk provider. I have a DID that is using DID translation through the SIP trunk group that is on the SIP gateway. I am now trying to setup one of the new OmniTouch 4135 SIP conference phones. Problem is that when you create the sip device user on the LAN i cannot dial internal IP phones because they need translated.
Is it possible to have a SIP gateway that has DID translation for the external SIP trunk and also a SIP gateway that is for the local SIP devices? I cannot find a way to configure this properly for all scenario's to work.
* SIP device must be able to call local IP touch phones and vice versa
* SIP device must be able to call public network through SIP trunk
* SIP trunk must be able call into OXE using DID's translated to local extensions
* 4645 is used for automated attendant.
The other small issue is the carrier doesnt like when the Q931 signal variant is ABC-F, only ISDN all countries, which in theory will work for the 4135 however you just get the extension on the screen, not the name and number.
When DID transcoding is enabled the SIP device just says NOT FOUND when trying to dial a local IP user.
I am doing this all with G711 right now with i160512.
Very frustrating to spend all day to get this working properly. Thanks for any suggestions in the management.
SIP Gateway / Does it not work properly with a SIP trunk and 4135 w/ DID translation
The easy way is to buy sip extension licenses and make the 4135 a sip extension. In that case it do'nt use a gateway.
The other solution is that you setup the internal sip gw for your sip device (4135) and you create a seccond sip trunk with a external gateway for your outgoing calls with ddi translation. This the way to separate your traffic.
The other solution is that you setup the internal sip gw for your sip device (4135) and you create a seccond sip trunk with a external gateway for your outgoing calls with ddi translation. This the way to separate your traffic.
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streamfx
Actually as per the release notes for 9.1 it shows:
As of Release 9.0, all standard SIP sets must be declared with the SIP Extension type. Standard SIP sets are no longer supported with the former "SIP Device" mode (formerly called "External set"). Only items other than standard sets (Nokia sets in Dual mode, Alcatel 4135 conference module, fax, video, etc.) should remain with the former "SIP Device" mode.
Creating as a sip extension did work. But my myteamwork still cannot call out the SIP trunk.
As of Release 9.0, all standard SIP sets must be declared with the SIP Extension type. Standard SIP sets are no longer supported with the former "SIP Device" mode (formerly called "External set"). Only items other than standard sets (Nokia sets in Dual mode, Alcatel 4135 conference module, fax, video, etc.) should remain with the former "SIP Device" mode.
Creating as a sip extension did work. But my myteamwork still cannot call out the SIP trunk.
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streamfx
MrAnMo wrote:The easy way is to buy sip extension licenses and make the 4135 a sip extension. In that case it do'nt use a gateway.
The other solution is that you setup the internal sip gw for your sip device (4135) and you create a seccond sip trunk with a external gateway for your outgoing calls with ddi translation. This the way to separate your traffic.
In this secenario again how would you handle incoming ddi calls from the external gateway since they hit the sip gateway?
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streamfx
streamfx wrote:FIXED IT! Go figure after all those hours of troubleshooting my Sip Ext gateway for my trunks has a different IP/DNS name when calls come in as the calls going out. Therefore when an a call came in it never applied to the right gateway. Duh!
Good to hear. I´m going to have the same issue in a few weeks from now. I´m working on a new installation with a external sip gw and 4135 sets so it´s nice to hear that is really works.
