Sure thing! Bug #18302 has been opened
(https://issues.asterisk.org/view.php?id=18302).
Brett Woollum
br...@woollum.com
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Try changing this line:
exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks))
To:
exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks))
Sent from my iPhone
On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote:
Here is a very very basic config. But, not
What is the error message?
Sent from my iPhone
On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote:
Hi Brett,
It did not work.
I will try other ideas.
SIP or Dial plan problem?
registeration?
On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote
a great job of showing what the table should look
like for the Voicemail ODBC storage, for example. But not for the Realtime
sip_users table.
I'm currently using the table listed here:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
Is there any official documentation on this?
Brett
didn't see any
resolution posted.
Brett Woollum
br...@woollum.com
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 12, 2010 4:58:24 AM GMT -08
2.50.0.52
Reg. Contact : sip:4...@10.20.1.225:5064
Qualify Freq : 12 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :
This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for
sip_peers.
Brett Woollum
br
More information: When I have rtcachefriends = yes in sip.conf, everything
seems fine. With rtcachefriends = no I see this behavior.
I'd rather not cache. I'm aiming for as near real-time as possible.
Any thoughts?
Brett Woollum
br...@woollum.com
- Original Message -
From
the NOTIFY's for MWI.
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 12, 2010 7:36:22 AM GMT -08:00 US
= no
(or maybe a little hahaha)!
Brett Woollum
Sent from my iPhone
On Nov 12, 2010, at 11:17 AM, Sherwood McGowan sherwood.mcgo...@gmail.com
wrote:
Inline response :D
On Fri, Nov 12, 2010 at 12:54 PM, Brett Woollum br...@woollum.com wrote:
Hi Sherwood,
Thanks for the reply.
Most definitely mate
That was it! I had a value (412 and 413) set for each phone. This overwrote the
caller ID that I was setting in the dialplan. Removing the contents of the
fromuser field cleared this issue.
Thanks Olle!
Brett Woollum
br...@woollum.com
- Original Message -
From: Olle E
Hi Carlos.
Yes I did have fromuser set, which was the problem. I removed this for each
extension and that solved the issue.
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: Asterisk Users Mailing List - Non
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum br...@woollum.com
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT
,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 into
sip.conf directly, this code works (ie: the CallerID(num) I set makes it out to
the destination phone properly).
Brett Woollum
br...@woollum.com
is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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