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I experience a very strange problem with our PBX (Asterisk 1.8, FreePBX, Grandstream GXP1200 phones)

If I call an internal number (1020 in this case) the asterisk server redirects me to the last called number of the phone.

I enabled and checked the debug log and it looks like the 302 message comes from the phone, but in the phone configuration I don't see the number or any redirecting feature anywhere.

The log looks like:

[2013-04-04 13:16:46] VERBOSE[3278] app_dial.c:     -- Called SIP/3543*
[2013-04-04 13:16:46] VERBOSE[3278] chan_sip.c: 
<--- Transmitting (no NAT) to 10.10.10.240:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.240:5062;branch=z9hG4bK197ebd0e30221ad9;received=10.10.10.240
From: "Michael *" <sip:1070@*.org;user=phone>;tag=1afa14dde19d2814
To: <sip:1020@*.org;user=phone>;tag=as24cb89e7
Call-ID: 62405cbce0fcf32e@10.10.10.240
CSeq: 39761 INVITE
Server: FPBX-2.10.0rc1(1.8.15)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1020@10.10.11.201:5060>
Remote-Party-ID: "Philipp *" <sip:1020@*.org>;party=called;privacy=off;screen=no
Content-Length: 0

the answer from the phone:

<------------>
[2013-04-04 13:16:46] VERBOSE[3278] app_dial.c:     -- Connected line update to SIP/1070-0000001c prevented.
[2013-04-04 13:16:46] VERBOSE[2654] chan_sip.c: 
<--- SIP read from UDP:10.10.10.247:5062 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.10.11.201:5060;branch=z9hG4bK3f571876
From: "IT Department" <sip:1070@10.10.11.201>;tag=as21ffd3bb
To: <sip:3543*@10.10.10.247:5062;transport=udp;user=phone>;tag=16149ed6566c95c6
Call-ID: 37138635018102a42e632a2261de52aa@10.10.11.201:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXP1200 1.2.5.3
Contact: <sip:001888*@*.org>
Diversion: <sip:35436*@10.10.10.247:5062;transport=udp;user=phone>;reason=unconditional
Content-Length: 0

The dialed number in this case is a teleconferencing server by AT&T in US.

I checked all configuration files if the number is somewhere but a grep over the directory and the number returns nothing.

Hope the message is not too specific to Grandstream phones.

Maybe there is a possibility to disable call forwarding complete from asterisk side?

Thanks in advance

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1 Answer 1

up vote 0 down vote accepted

Ok I found the solution after digging in the manuals.

there is no menu or something for the Grandstream GXP series for call-forwarding, but if GXP call features is activated in the acount, the unconditional call forwarding can be disabled my using the same GXP call feature codes:

*71 - follwoed by the number sets the unconditional call forward *72 disables the unconditional call forward

for me i had to press line 2 first, because it only happened on the second voip line

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