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Summary:ASTERISK-16391: Fetching Call with Grandstream-BLF (**${EXTEN})
Reporter:Florian Schroen (florianschroen)Labels:
Date Opened:2010-07-19 08:29:27Date Closed:2010-07-20 08:11:01
Priority:MinorRegression?No
Status:Closed/CompleteComponents:General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) fulldebug.5
Description:It is not possible to fetch a call via the Busy-Lamp-Field of a Grandstream-Phone.

I have 3 phones 116, 123, 202

when i call 123 from 116, the busy-lamp is blinking on the 202. now I try to fetch the call with pressing the blinking busy-lamp-button on phone 202.

now phone sends "**123" and the call should be fetched. but instead of this I'll get an error 603.

the same configuration on phone and asterisk is working fine if i use an asterisk 1.6.1.X

so i had a look at the sip-headers and there's the problem. the 2 stars in front of the extension are not shown. (see log above)

****** ADDITIONAL INFORMATION ******

##> begin of log <##

sip*CLI> core show version
Asterisk 1.6.1.20 built by root @ sip.knowledge.lan on a x86_64 running Linux on 2010-07-19 12:37:59 UTC

<--- SIP read from UDP://192.168.150.91:5060 --->
INVITE sip:**123@sip.knowledge.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.150.91:5060;branch=z9hG4bKff8a68521aa38f11
From: "Florian Schroen" <sip:202@sip.knowledge.lan;user=phone>;tag=2221a7909c25f40c
To: <sip:**123@sip.knowledge.lan;user=phone>
Contact: <sip:202@192.168.150.91:5060;transport=udp;user=phone>
Supported: replaces, timer, path
X-Grandstream-PBX: true
P-Early-Media: Supported
Call-ID: ffbbd44792375fc9@192.168.150.91
CSeq: 1806 INVITE
User-Agent: Grandstream GXP2010 1.2.3.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 407

v=0
o=202 8000 8000 IN IP4 192.168.150.91
s=SIP Call
c=IN IP4 192.168.150.91
t=0 0
m=audio 5074 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
[...]

### call is fetched correctly ###



sip*CLI> core show version
Asterisk 1.6.2.9 built by root @ sip.knowledge.lan on a x86_64 running Linux on 2010-07-06 17:53:05 UTC

<--- SIP read from UDP:192.168.150.91:5060 --->
INVITE sip:123@sip.knowledge.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.150.91:5060;branch=z9hG4bK3f8bd40008d612c2
From: "Florian Schroen" <sip:202@sip.knowledge.lan;user=phone>;tag=5edbc26991415ba2
To: <sip:123@sip.knowledge.lan;user=phone>
Contact: <sip:202@192.168.150.91:5060;transport=udp;user=phone>
Replaces: pickup-4e91311ff3be8002@192.168.150.91
Supported: replaces, timer, path
X-Grandstream-PBX: true
P-Early-Media: Supported
Call-ID: fd8fc46fcb7f93f3@192.168.150.91
CSeq: 25032 INVITE
User-Agent: Grandstream GXP2010 1.2.3.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 407

v=0
o=202 8000 8000 IN IP4 192.168.150.91
s=SIP Call
c=IN IP4 192.168.150.91
t=0 0
m=audio 5070 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
[...]

### Error 603 at the phone ###

##> end of log <##
Comments:By: Paul Belanger (pabelanger) 2010-07-19 09:55:05

Do not upload your debug log as notes, upload them to the tracker directly (see below).
--
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Florian Schroen (florianschroen) 2010-07-19 10:54:28

uploaded debug-log.
By: Paul Belanger (pabelanger) 2010-07-19 14:17:44

A sample dialplan would be helpful too.

Edit: Nevermind.


By: Paul Belanger (pabelanger) 2010-07-20 08:11:01

Closing, since a duplicate.
---
Thanks for the bug report. This particular bug has already been reported into our bug tracking system, but please feel free to report any further bugs you find. Thanks!