I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten = 1102,1,Answer()
exten =
Has anyone run into this problem yet?
-Daniel
On 9/9/05, Daniel Corbe [EMAIL PROTECTED] wrote:
I'm getting a FLOOD of these types of messages on my MAC OS X box:
Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread:
Request to schedule in the past?!?!
Sep 9 14:46:37 NOTICE
I'm getting a FLOOD of these types of messages on my MAC OS X box:
Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread:
Request to schedule in the past?!?!
Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread:
Request to schedule in the past?!?!
Sep 9 14:46:37
Hello,
I'm currently researching a project that would enable us to pull the
actual signaling (SIP conversation) along with our CDRs
The best way I can tell to approach this is to set up a server on a
SPAN port which mirrors all my proxy servers' traffic.
I was curious if anyone else has ever
since it is so fundamental. Perhaps a
bug should be raised?
Regards
Cameron
- Original Message -
From: Daniel Corbe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 7:29 AM
Subject: [Asterisk
, Doug Meredith [EMAIL PROTECTED] wrote:
Daniel Corbe [EMAIL PROTECTED] wrote:
Is there any way to turn Loop Detection off or tune the params a bit?
I am having an issue with Call Forwarding on my SIP Proxy Server which
is causing me great pains.
All I can do is sympathize. The same problem
Hello,
Is there any way to turn Loop Detection off or tune the params a bit?
I am having an issue with Call Forwarding on my SIP Proxy Server which
is causing me great pains.
Here is the issue:
1) I have a SIP UA which registers with a SER proxy server.
2) I have an Asterisk TDM gateway in my
Anyone have the example.txt file that Cisco's documentation on the
7905 IP phone keeps referring to? Or can someone possibly share a
fairly complete example config file for these phones with me?
Thanks!
-Daniel
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I've got a nagios plugin making sure the * box is up, but I would like
to do more than that.
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
-Daniel
Okay
here's a quick and dirty little perl script to monitor the PRI Status
and mimic nagios plugin output.
-Daniel
On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy [EMAIL PROTECTED] wrote:
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs
Yeah, I'd be interested in porting your work so it runs under nagios.
Please post your results when you're finished.
-Daniel
On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote:
I've got a nagios plugin making
My PBX seems to have just started showing wierd codec negotiation problems.
I'm not all of a sudden getting this on certain phone numbers on my system:
Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1683
ast_set_read_format: Unable to find a path from ULAW to G729A
Feb 8 22:19:19
Hello,
I've inherited a (now) broken asterisk implementation. It seems as if
there are currently codec tanscoding issues in this box. Specifically
I am receving calls from a SIP proxy in G.729 and attempting to
transcode them to ULAW.
My asterisk installation was working up until yesterday.
Is there any way to quickly poll an asterisk server for concurrent
call count? Preferably from like a perl or PHP script.
-Daniel
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To
:)
On Thu, 3 Feb 2005 10:41:37 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
Is there any way to quickly poll an asterisk server for concurrent
call count? Preferably from like a perl or PHP script.
-Daniel
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Hrm,
All of a sudden for some reason Wait() and Playback() are returning
non-zero and its causing calls on my inbound SIP leg not to complete.
I'm not sure why
-- Executing Answer(SIP/2181-4518, ) in new stack
-- Executing Playback(SIP/2181-4518, silence/1) in new stack
-- Playing
to operate. Cisco CallManager being a 10,000
dollar application, I would like to find any open source alternatives.
Regards,
Daniel
--
Daniel Corbe, CCNP
Senior Network Engineer
Results Technologies, Inc.
952-921-2400 x104
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