Customer had their IT people swap out their phones and equipment on their half PRI and pots lines. Before the change out, they were able to use *72 just fine to forward their pots line to an answering service. After the equipment change out, they called us with some issues calling out. After making a few workaround changes on the Adtran, it worked just fine. They were unable to call 911 and had to make an entry for that too. Now, when they try to dial *72 to invoke call forwarding, they get an error message that the number cannot be completed as dialed. Tech tried directly off of the punch down block and are still getting an error message that the call cannot be completed as dialed. Is there something I am missing that will allow the star codes to pass through to the switch? Attached is the current configuration.
I wanted to provide the solution that resolved our issue. Come to find out, we just need to add a line
isdn switch-type 5ess
to be compatible with the unit the phone equipment vendor had installed. Once we did that star codes are working as they should. Thank you to all that assisted in their comments.
On voice grouped-trunk T01 I would suggest adding the following
accept *X$
Thanks shambler for your reply. I added that and we still get the same message that you call cannot be completed as dialed.
Have you tried adding SPRE codes?
can you attach updated config with additional accept statement
Also gather following debugs
debug int fxs
debug voice verbose
debug sip stack messages
When I try to add the SPRE code is says "%SPRE code already mapped to Call last dialed number".
You will find the current config attached as well as the debug logs. On the debug logs another call came in right as we were testing so you may have to sort through that other data. Logs begin where she was trying *73 to deactivate Call Forward Always.
the *73 originating from FXS 0/5 egresses ok out the T01 trunk to voip.lighttube.net
If that generates an error that is coming from the carrier level.
no *72 events detected in debug
If there was a *72 event from the customer did that originate from the PRI connection? if so I also noticed no *X$ ISDN trunk T02
We did not have the customer do *72 wince she does not want to activate the CFA. We test with *73 which will bring up the announcement that the CFA has been deactivated.
no SIP debug on it but yea appears to be carrier.
Perhaps no call forward variable provisioned on individual line 9314552005
From your last debug gathering it showed call being answered, so the recording is coming from the carrier level
09:15:48.871 SIP.STACK MSG | Rx: UDP src=209.209.172.231:5060 dst=10.18.216.10:5060 |
09:15:48.872 SIP.STACK MSG | SIP/2.0 200 OK |
09:15:48.873 SIP.STACK MSG | From: "Marvel MainLine" <sip:9314552005@voip.lighttube.net:5060;transport=UDP>;tag=2a1fe08-7f000001-13c4-1552a-7aa937aa-1552a |
09:15:48.873 SIP.STACK MSG | To: <sip:*73@voip.lighttube.net:5060>;tag=1299664383-1486134948829 |
09:15:48.874 SIP.STACK MSG | Call-ID: 2a47310-7f000001-13c4-1552a-55f469ce-1552a@voip.lighttube.net |
09:15:48.875 SIP.STACK MSG | CSeq: 2 INVITE |
09:15:48.875 SIP.STACK MSG | Via: SIP/2.0/UDP 10.18.216.10:5060;branch=z9hG4bK-1552a-534af03-2fb9dde0 |
09:15:48.876 SIP.STACK MSG | Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE |
09:15:48.877 SIP.STACK MSG | Supported: |
09:15:48.877 SIP.STACK MSG | Accept: application/media_control+xml,application/sdp |
09:15:48.878 SIP.STACK MSG | Contact: <sip:*73@209.209.172.231:5060;transport=udp> |
09:15:48.879 SIP.STACK MSG | Call-Info: <sip:172.16.6.141>;appearance-index=1 |
09:15:48.879 SIP.STACK MSG | Content-Type: application/sdp |
09:15:48.880 SIP.STACK MSG | Content-Length: 150 |
09:15:48.880 SIP.STACK MSG | |
09:15:48.881 SIP.STACK MSG | v=0 |
09:15:48.882 SIP.STACK MSG | o=BroadWorks 55660592 1 IN IP4 209.209.172.231 |
09:15:48.882 SIP.STACK MSG | s=- |
09:15:48.883 SIP.STACK MSG | c=IN IP4 209.209.172.231 |
09:15:48.883 SIP.STACK MSG | t=0 0 |
09:15:48.884 SIP.STACK MSG | m=audio 64674 RTP/AVP 0 |
09:15:48.885 SIP.STACK MSG | a=rtpmap:0 PCMU/8000 |
09:15:48.886 SIP.STACK MSG | a=ptime:20 |
I wanted to provide the solution that resolved our issue. Come to find out, we just need to add a line
isdn switch-type 5ess
to be compatible with the unit the phone equipment vendor had installed. Once we did that star codes are working as they should. Thank you to all that assisted in their comments.