The other day my asterisk local SIP clients got hung when my provider had a
DNS failure. All registrations went dead (even the ones that were IP
addresses) and all sip peers went offline. I know this was know problem at
one point is there any update on this when using a FQDN for one of the peer
Mark,
I thought I would also mention that I am still having similar issues even
after updating to the latest Asterisk, Zaptel and Libpri. Although I am
using a Sangoma, we have similar symptoms with a restart fixing it. I am
starting to wonder if I must go back to a Digium card. We originally
Mark,
I thought I would chime in here on your problem. Oddly, I have having the
same issue with a PRI with similar symptoms. The odd part is that I have
never had an issue like this with a asterisk PRI setup. My setup is a PRI
with a Sangoma card with the exact same issue with 1.4.14. After a
Kevin,
After upgrading to the latest build of everything have you seen the problem
anymore?
Don't know yet, waiting for it to break ( not a good feeling as you know)
What's your hardware and software configs? Maybe we can find a similarity in
our systems.
It's a dell poweredge with
I have a remote user on a Polycom IP Phone who has set call forwarding by
accident and is away from the phone. Does anyone know of a way to remotely
un-forward the phone? I tried to reboot the phone but that didn't work and
removing the mac-phone.cfg caused problems
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I
re-touched the file and followed your suggestion.
Any way of removing the call forwarding feature via the xml configs?
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
I guess I was interested in Disabling the forwarding feature completely via
the config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Thursday, January 17, 2008 7:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
The only problem with this workaround is that on the Polycom 550 (backlit
display) the backlit goes bright every 30 seconds then back to dim. Any
work around for that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson
Sent: Friday, December
I'll second the interest in this topic... a walkthrough on this topic
especially with OpenSER would be great..
Kevin Kiely
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Monday, October 08, 2007 12:25 PM
To: asterisk-users
via Telnet and see more info but no place for the realm or Sip credentials.
Am I missing something?
Thanks in advance..
Kevin Kiely
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Friday, September 14, 2007 10:33 AM
To: Asterisk
When using app_followme, I am receiving the following warnings on the
console. We are calling the followme app with no options for additional
voice announcements. Is anyone else experiencing this issue with 1.4.11?
-- Executing [EMAIL PROTECTED]:1]
FollowMe(SIP/101206006-b72223d8,
Sent: Thursday, September 13, 2007 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Followme app_followme
Kevin Kiely wrote:
When using app_followme, I am receiving the following warnings on the
console. We are calling the followme app
Does anyone know a way in Asterisk 1.4 to select the options from the
menuselect menu from the command line?
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*menuselect-tree*
look in menuselect-tree, and...
hmm... this looks promising for trying to figure it out...
Current Directory is
/usr/local/src/asterisk-1.4.5/menuselect
-rw-r--r-- 1 root 31131 Aug 19 2006 example_menuselect-tree
daveC
Kevin Kiely wrote:
Does anyone
Does anyone have the multiparking feature enabled in asterisk 1.4? or
suggest multiple parking lots?
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Anyone using any variation of Multiparking, Parking Valet or servicing Call
Parking with Multiple Tennants?
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I should have included using a multi parking feature with asterisk 1.4?
-Original Message-
From: Kevin Kiely [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 17, 2007 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Multiple Parking Lots
-Commercial Discussion
Subject: Re: [asterisk-users] MultiParking
Look at app_valetparking here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+addons
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Kiely
Sent: Monday, July 16, 2007 16:47
To: 'Asterisk Users
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/
Indicates support for Asterisk 1.4. The documentation listed suggests an
install like so:
cd /usr/src/asterisk
cp contrib/scripts/astxs /usr/bin/
cd apps
wget http://www.bkw.org/app_valetparking.c
cd ..
astxs -install
I cant figure this out. I have seen this same example many places but the
group never gets incremented. Am I missing something?
exten = 99,1,Set(GROUP(99) = G99)
exten = 99,2,GotoIf($[${GROUP_COUNT(99)}0]?103)
exten = 99,3,dial(SIP/qoqieoeiwq)
exten = 99,103,Hangup
I tried to look at the code in Trixbox but when the option 'confirm' is
selected in the follow me properties screen, no code is generated and the
call goes dead. Is there a trick to get the code generated?
_
From: Philippe Lindheimer [mailto:[EMAIL PROTECTED]
Sent: Saturday, March
Ok, bug report submitted. 0009307
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Friday, March 16, 2007 6:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 Follow-Me Application
On 3/16/07, Kevin Kiely
I am having an issue with the follow me application in 1.4
The application description (below) indicates that if the specified
followmeid profile doesn't exist in followme.conf, execution will be
returned to the dialplan and call execution will continue at the next
priority.
That's not happening
Before you take the jump, take a look at my post earlier regarding 1.4 and
follow me
_
From: Ritesh Agrawal [mailto:[EMAIL PROTECTED]
Sent: Friday, March 16, 2007 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Follow me on multiple
Any ideas?
Did anyone experience something like that?
Thx
Yes, unfortunately, all the time. There answer is if it works with a sip
softphone client than it's not their problem. It does work with the
softphone client.
-Original Message-
From: Bartosz Wegrzyn - maillists
Is anyone having problems and Broadvoice with incoming DTMF not being
recognized from a caller originating on the PSTN connection to Broadvoice?
Broadvoice tech support confirmed this issue as a result of their carrier
connections and suggested a work around in the dial plan(SIPDtmf). This
does
This parking patch looks like a good idea. I applied the patch but it
doesn't seem to work. The patch install was successful and I modified my
features.conf like the features.conf.sample suggested. I don't see any
mention of the k or K in the 'show application dial'. Any ideas? Did I miss
a
I tried unsuccessfully to get this to work. I am using AAH 2.7 which has
asterisk 1.2.5.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
This information seems to indicate there is a problem with the 1850 and the
onboard nic.
http://connection-telecom.com/support.html
-Original Message-
From: Ed Greenberg [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 16, 2006 1:45 PM
To: Asterisk Users Mailing List -
Has anyone used the Polycom expansion module with multiple lines?
My application is for 20 lines and read there was a limit of 7 at one point.
Thanks
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I am considering a Polycom expansion
module for the IP601 for a DSS/BLF application. I had read that there was a limitation as to
the number of lines that could be monitored with the hint
command.
Can anyone tell me if they are using this
with multiple lines, I need to monitor 20
Be careful here... Our local PSAP is handled by the fire department. I had
one of our guy's make a test call and we were told that this test must be
coordinated and scheduled in advance with the chief. They want no test
calls. It would probable be safest to check before making the call as they
Is there a way to patch an existing Asterisk 1.2.5 version with the
follow me application?
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 25, 2006 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
I am trying to limit IAX connectivity to a server with the permit/deny
combination. In this example to allow ip 123.123.123.123 but it's not
working. If I remove the mask on the deny parameter it allows all
hosts. With the deny statement like below it blocks all connections
even using a mask or
Does anyone have any suggestions as how to enter a CDR Account code
during a call?
I know it can be done in the extension logic before the answering the
call, but I wanted to optionally enter an account code on certain calls
without prompting on every call before or after the call?
App_valetparking is a great (and necessary) addition to asterisk. Does
app_valetparking.c work with the current release of asterisk? I tried
to install it on Asterisk 1.0.9 and I get errors following the
instruction in the wiki?
app_valetparking.c:678: dereferencing pointer to incomplete type
App_valetparking is a great (and necessary) addition to asterisk. Does
app_valetparking.c work with the current release of asterisk? I tried
to install it on Asterisk 1.0.9 and I get errors following the
instruction in the wiki?
app_valetparking.c:678: dereferencing pointer to incomplete type
Does anyone know how to join two .wav audio files via the command line
in Linux for playback with Asterisk?
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We are getting HDLC errors on a PRI with a Dell PowerEdge SC420. I
suspect it may be an interrupt issue.
Can anyone recommend a low cost name brand server that will not share
the interrupts or have the issues that the Dell PowerEdge SC420.
Thanks
A while back I used a Meridian Mail system that had a goto function to
go to a specific message in your inbox/folder. I found this feature
useful as I tend to keep a fair amount of messages in my boxes and it's
helpful to advance to a more recent message like ( for example in the
advanced menu
I have an interest in using vendor strings in my DHCP scope to assign
different IP's for my Polycom and Cisco phones. Has anyone used this
approach and may have some examples of the dhcp.conf with the strings?
Thanks
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I am having a problem communicating with my asterisk box behind a Cisco
router. I am running NAT on the inside and wanted to port forward to
the asterisk IP but it is not working. I must be missing something..
This is the NAT statement I am using:
ip nat inside source static udp 10.2.1.50 4569
I had the same problem on the HEAD version and went to STABLE to resolve
it.
-Original Message-
From: Andrew C. Brown [mailto:[EMAIL PROTECTED]
Sent: Friday, April 15, 2005 6:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Empty voicemail attachments?
I have Asterisk
Exactly the point. Not sure if there is a limitation but it is generally
provisioned the same as with most of there other offering which support
call waiting. I don't think they want to advertise that multiple
channels can be used, I think they monitor the monthly minutes and make
an assessment
T1 PRI
-Original Message-
From: Scott Wolfe [mailto:[EMAIL PROTECTED]
Sent: Monday, April 04, 2005 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Concurrent calls: best provider?
This brings up the question. What is the best service for
Not that I need to stick up for Broadvoice and yes, they are not very
good at returning emails, but, for me, I have used Broadvoice on several
asterisk systems at different locations and haven't had any problems.
It works great and is very flexible.
-Original Message-
From: Daryll Strauss
Try this:
dtmfmode=inband
register =
[number]:[EMAIL PROTECTED]
[broadvoice]
type=peer
fromuser=[number]
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice
reinvite=no
canreinvite=no
pedantic=yes
qualify=yes
disallow=all
allow=alaw
better service
than voice pulse?
Chris Ford
CMF International Technologies LLC.
[EMAIL PROTECTED]
- Original Message -
From: Kevin Kiely
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, January 25, 2005 7:51 PM
Subject
Does anyone use VoicePulse Inbound service and receive Caller ID Name?
I receive caller ID number but no name.
Thanks,
Kevin
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