Dial 800 numbers, get message about Disconnected

[Randall Garner]

Since we upgraded to 4.5.0.1090 Epsilon Geminids last week, we found we are not able to make outbound calls to 800 numbers. When we attempt this, we get a message stating "Disconnected".

 

We could dial the 800 numbers before the upgrade. I've check our Dial Plan and it doesn't appear to have changed. I have attached to this ticket.

 

Please advise,

 

Randall

[Vodia support]

Hi, can you post a siptrace? also when you dial the 800# do you add the 1 in front of the number? in my lab I have the country code set to 1 so it automatically adds the 1 when the invite goes out to the provider, or you can add 1* on the replacement of the dial plan.

[Randall Garner]

This morning I tried an 866 number two times - one with and one without the "1". Both test calls failed with message on the LCD - "Disconnected: Not Found" followed by the phone number dialed.

 

Attached is both call logs from the test calls.

 

Please advise. Thanks for your help,

 

Randall

With_1.txt

WithOut_1.txt

[Vodia support]

- Use custom headers

- Set “To” as “Same as Request URI”

- Change the “From” if the previous step did not work try something like - {from-display} <sip:{trunk-ani}@{domain};user=phone>

 

 

[8] 2012/09/11 07:20:04: SIP Tx udp:24.96.139.170:5060:

INVITE sip:xxxxxxx@24.96.139.170;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.36.1.18:5060;branch=z9hG4bK-af21e9d9aaa08f8b96212f4a1ef56c1b;rport

From: <sip:xxxxxxxxxxx@24.96.139.170>;tag=65071

To: <sip:xxxxxxx@24.96.139.170;user=phone>

Call-ID: 00f4c302@pbx

CSeq: 29796 INVITE

Max-Forwards: 70

Contact: <sip:xxxxxxxxxxx@10.36.1.18:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4025

Remote-Party-ID: "Copy Room" <sip:xxxxxxxxxxx@pbx.xxxxxxx.com;user=phone>;party=calling;screen=yes

Content-Type: application/sdp

Content-Length: 272

 

 

----------------------------------------------

[5] 2012/09/11 07:23:42: SIP Tx udp:24.96.139.170:5060:

INVITE sip:XXXXXXX@24.96.139.170;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.36.1.18:5060;branch=z9hG4bK-b72281dd87ef126545592358a2a107f9;rport

From: "Copy Room" <sip:XXXXXXXXXXX@pbx.xxxxxxxx.com;user=phone>;tag=8937

To: <sip:XXXXXXX@pbx.gulfworld.com;user=phone>

Call-ID: 2a515518@pbx

CSeq: 21322 INVITE

Max-Forwards: 70

Contact: <sip:XXXXXXXXXXX@10.36.1.18:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Remote-Party-ID: "Copy Room" <sip:XXXXXXXXX@pbx.xxxxxxx.com;user=phone>

Content-Type: application/sdp

Content-Length: 272