JohnsonOps
Technical User
We have H.323 Trunking setup to an Asterisk endpoint providing VoIP into our IP Office 406v2 and it is working fine. After upgrading to 4.0 (5) the following problems started to occur:
[ul]
[li]When a call comes in over an analog line and is forwarded out via forwarding on the extension to the VoIP line, the call doesn't go thru and in the traces a [blue]Cause Code = 21[/blue] is shown. This same issue occurs if the call comes in over the H.323 VoIP Line as well.[/li]
[li]If a call from an extension to an extension with call forwarding on to a number that goes over the VoIP line rings the remote number but when picked up it drops the call with [blue]Cause Code = 16[/blue] (this is normal call clearing code)[/li]
[/ul]
We were able to make a receive calls over the H.323 trunk with an IP phone or DS phone, just when transfering are the problems coming up.
Has anyone out there using H.323 Trunking setup and tested their trunks under 4.0 yet?
I would like to use SIP trunking but it sounds like it could consume significant resources, if it needs a VCM for each SIP connection and then another to the IP phone. Would be good if the SIP call could go direct to the SIP phone or H.323 phone.
[ul]
[li]When a call comes in over an analog line and is forwarded out via forwarding on the extension to the VoIP line, the call doesn't go thru and in the traces a [blue]Cause Code = 21[/blue] is shown. This same issue occurs if the call comes in over the H.323 VoIP Line as well.[/li]
[li]If a call from an extension to an extension with call forwarding on to a number that goes over the VoIP line rings the remote number but when picked up it drops the call with [blue]Cause Code = 16[/blue] (this is normal call clearing code)[/li]
[/ul]
We were able to make a receive calls over the H.323 trunk with an IP phone or DS phone, just when transfering are the problems coming up.
Has anyone out there using H.323 Trunking setup and tested their trunks under 4.0 yet?
I would like to use SIP trunking but it sounds like it could consume significant resources, if it needs a VCM for each SIP connection and then another to the IP phone. Would be good if the SIP call could go direct to the SIP phone or H.323 phone.