After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the
problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX
protocol went downhill and many carriers (like VoicePulse) are discontinuing
support for IAX.
Is this correct? We are all heading for SIP?
I use simple port forwarding on an Linux firewall (iptables)...so that's not
the issue.
I was referring to IAX2 of course (IAX has be gone a long time I think)...
Unlimitel is running * 1.4.x (and so am I)...
I just can't understand IAX2 connections suddenly dropping (on one day)
being protocol
Robert,
We've helped clients setup monitoring scripts for this type of situation - 2
different ways. One is a ping script, the other monitors the asterisk peer
status of registration. These were temporary until they could get to the
root cause however. Since you have multiple providers going
www.linqone.com
-Original Message-
From: supp...@ocg.ca [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of OCG Technical Support
Sent: Saturday, March 07, 2009 9:59 AM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?
Robert
Damn you for solving this before he upped the bounty by a pack of tictacs!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: March 3, 2009 10:51 PM
To: Asterisk Users List
Subject: Re:
Install a Microsoft product.
(Sorry I couldn't resist when I saw the subject)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: March 4, 2009 8:48 AM
To: Asterisk Users List
Subject: Re:
Perhaps if he threw in a paperclip and some tictacs people would respond...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: March 3, 2009 7:37 PM
To: Asterisk Users List
Subject: Re:
Has anyone written a Windows Mobile app which gets the MWI info from a SIP
server, and updates the VM counter in the OS?
I'd like my PPC to show my voicemail count (and SIP MWI seems like the
easiest way)
___
-- Bandwidth and Colocation
Are you sure this is not just a standard SIP MWI message?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: February 23, 2009 8:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] strange
Did you use the same screen name / name for the 2 SIP extensions you setup
on the one phone? If so, some phones will confuse asterisk based on the SIP
header (in particular AASTRA phones). If this is an Aastra phone, this is
probably the cause...
From: asterisk-users-boun...@lists.digium.com
dialplan matching issue
Importance: High
OCG Technical Support schrieb:
We use extensions like plant201 and tunnel12 so it does work in 1.4
As a *pattern* (e.g. _plant2XX, _tunnel.)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
We use extensions like plant201 and tunnel12 so it does work in 1.4
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: February 11, 2009 10:16 PM
To: Asterisk Users List
Subject: Re:
Don't expect too much from Aastra. In our previous dealings trying to
report serious bugs (like phone lockup/crash) to Aastra, they didn't want
the details, or they simply gave us canned answers which did no good.
(Superficial tech support)
We've moved away from Aastra for new installs, but we
Have a look at smartCID at www.generationd.com
Uses a simple mySQL database, allows for call blocking flag, reverse CID
lookup, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent:
Check out the HP ProCurve Switch 2610-24-PWR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February 1, 2009 6:58 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE
Importance: High
My memory of a HP procurve (a 2626 PWR from memory) was that it was
quite noisy - have they changed?
PaulH
OCG Technical Support wrote:
Check out the HP ProCurve Switch 2610-24-PWR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
A little off topic but
I need to put a 24 port Gig PoE switch into a small office - no computer
room / rack etc. All CAT5 terminates near the owners desk (smart huh?).
I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone
I've researched so far is LOUD...
and has module just for that.
G.
On Fri, Jan 30, 2009 at 7:56 PM, OCG Technical Support supp...@ocg.ca
wrote:
No - the server generates the error:
Software error:
Hrm, can't seem to open
/var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav
For help, please send mail to th
Of Darrick
Hartman
Sent: January 31, 2009 10:02 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE gig switch
Importance: High
OCG Technical Support wrote:
A little off topic but
I need to put a 24 port Gig PoE switch into a small office - no computer
room / rack
I just tried the vmail.cgi app. Although working, there is clearly a
permissions problem preventing playing the wav files.
I run Fedora 8, and the patch files (on the wiki) are apparently broken.
Does anyone have a solution for fedora?
Thanks
From:
Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Support
Sent: January 30, 2009 9:40 PM
To: Asterisk Users List
, 2009 at 9:53 PM, OCG Technical Support supp...@ocg.ca
wrote:
Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Connect an amp to the onboard speaker (run wires out the case)...then you
can really blast the ring!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: January 28, 2009 10:53 AM
To: Asterisk Users List
Subject: Re:
I've noticed on a few installations that the very first audio played after a
call in answered (eg: Greeting), the first part of the audio is
cutoff/stuttered.
Is this because Asterisk needs some RTP to create a sync for audio - and the
first 1 second is lost? Should one play 1 sec of silence
I'm looking for some ideas of people who have setup a sidecar (eg: Aastra
560M).
Obviously it's handy for BLF (to see who's on a call)...but what else?
Anyone want to share interesting things they've done with a sidecar?
___
-- Bandwidth
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: January 18, 2009 3:11 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Using a sidecar? Ideas?
Importance: High
OCG Technical Support schrieb:
I'm looking
Sent: January 18, 2009 9:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Using a sidecar? Ideas?
Importance: High
Here is a list of stuff that I can remember:
BLF
Group login/logout
Day/Night mode
Call Record
Speed dial
On Sun, Jan 18, 2009 at 2:43 PM, OCG Technical Support supp
If you want to email me your fixed script I'll put it up on the web site...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 15, 2009 7:08 PM
To: Asterisk Users List
Subject: Re:
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 14, 2009 5:31 PM
To: Asterisk Users List
I can do a great Colonel Klink and pretty good Shulz imitation...in case you
want me to record some prompts.
:)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: January 7, 2009 11:31 AM
I see a variety of DECT 6 phones available CHEAP at costco. Is there a way
to convert these to SIP?
I recall someone talking about a Siemens devices that works with all DECT
phones, making them SIP (I think)
___
-- Bandwidth and
: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
Importance: High
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 11:14 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk
I'm using the Wm6 built in client. (Enabled via CAB file to add-back files
removed from ROM)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta
Sent: December 3, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host
mean Windows Live Messenger?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 9:15 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client.
Overall things are working well. However, I regularly get the following
message:
[Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote
host can't match request NOTIFY to call
'[EMAIL PROTECTED]'.
I've got a user with Linksys ATA's for their analog phones. At random times
during calls, the other party hears DTMF tones during the call.
Is there a way to solve this?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Take a look at smartCID at www.generationd.com
This tool will set callerid based on number in a database. If not listed
there, it will search 411 for reverse lookup etc.
It will also let you flag calls for blocking, etc..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Could you send a link to the post you referenced? I'd like to get sendmail
working with rogers too...
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: November 8, 2008 8:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users]
This was so interesting I had to move it to its own thread!
Is anyone using this script? How does it perform compared to the older
WonderShaper script?
-M-
==
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-Original
Post it on the wiki! Im sure Ill need it someday
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of César García
Sent: October 29, 2008 6:54 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] XML Cisco config file
Well guys I got it, I started up again making the xml
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc...
Are you looking for lost RTP between * and internal phones or * and external
provider?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24,
that
picks up calls from the analog lines and routes them to the appropriate
phone, although it will eventually be linked to a larger system once all
the minor bugs are resolved.
OCG Technical Support wrote:
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc
I have a new Fedora 9 firewall I am setting up in front of an Asterisk 1.4
box. I ported over all of my iptables rules..but now have a strange
problem: SOMETIMES, the audio is only 1-way (i.e. and RTP path problem).
Can someone offer a tip here? Since I have conntrack_sip loaded on the
I have a 7921 wireless phone working with Asterisk, and I want to tighten
the wide open port range of my IPTABLES now.
I tried allowing only SCCP port (2000) in/out and found that my audio was
gone. A quick look at my iptables message shows source port 15886 and dest
port 25968 used:
FORWARD
I'm setting up a 7921 and now want to add a second line to the phone. In my
SCCP.conf file I have:
autologin = 235,299
However, on reloading SCCP the phone fails to login to the second line with
this error:
[Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register:
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
here?
Thanks
MD
on Cisco 7921
On 13:15, Fri 12 Sep 08, OCG Technical Support wrote:
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something
I would have said the short answer is IAX
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: September 12, 2008 7:31 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Which internet phone protocol best to choose
The short
This is some pretty basic stuff... (someone will probably send you a RTFM)
Start with the sample dialplan (make samples I think)...trace the dialplan
along to understand how it works
Check the wiki and then post anything that you need help with
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I little more digging and I confirmed that cell phone VM and FAX waiting
icons are in fact controlled by a proprietary SMS message format. Here's
what I found:
Message Waiting Indication Group: Store Message
This Group allows an indication to be provided to the user about the status
of
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing # voicemails waiting. Can Asterisk do the same somehow?
MD
Technical Support [EMAIL PROTECTED]
wrote:
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing # voicemails waiting. Can Asterisk do the same
On Sun, Jun 22, 2008 at 11:23 AM, OCG Technical Support [EMAIL PROTECTED]
wrote:
We have a number of clients who have replaced their cell carrier
voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a
message
showing
-0400, OCG Technical Support wrote:
I've tried a few approaches to making the multimedia keys on my kbd play
nice with myth, but all have lead to dead ends.
One such dead end is to post this question to the Asteris Users mailing
list, I guess :-(
Wrong list?
--
Tzafrir Cohen
icq
I've tried a few approaches to making the multimedia keys on my kbd play
nice with myth, but all have lead to dead ends.
I decided to take the simple approach, and use the myth setup menu for
keyboard mappings. Now, I have myth (0.20) waiting for a key with Press a
key, but when I press the
I'm looking at building up a standard asterisk system fanless/no moving
parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it
is SLOW...25mb/sec read 8mb/sec write.
Has anyone tried a slow disk like this on asterisk? Will this delay voice
prompts or screw up ast/linux in
Change the order of resolution (hosts first, then DNS) and add relevant
entries to your hosts table. That makes asterisk happy w/o an internet
connection.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: June 9, 2008 9:09 PM
To:
Has anyone create the necessary config/kbd file to allow the DiNovo mini to
work well with myth? (Mapped all of the multimedia buttons etc)
=MD=
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
PROTECTED] On Behalf Of OCG Technical
Support
Sent: June 6, 2008 9:03 PM
To: Asterisk Users List
Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth
Has anyone create the necessary config/kbd file to allow the DiNovo mini to
work well with myth? (Mapped all of the multimedia buttons etc
Anyone tried Asterisk with Fedora 9 (recent release)?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
We also have a script available (on www.generationd.com) which allows a user
to reply to an emailed voicemail, which then deletes the associated VM file
on the asterisk box.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent:
A client has asked that our asterisk installation leverage their large
investment in their existing data center infrastructure. We're thinking
about putting the voicemail messages onto a Samba share (on their file
servers). Any pros/cons to this? Does network/samba latency create
choppiness?
Like many users I get my voicemails emailed to me, AND left on the asterisk
server, so that I can retrieve them by phone or by email. However, I was
frustrated that after I deleted a message in outlook that I still had to
delete it from asterisk manually.
So, I wrote a script that runs on the
After lots of interest I've stopped emailing people the script and have made
it available for download from www.generationd.com Look in the Downloads |
Asterisk section.
Be sure to read the readme AND the top of the script for instructions...
___
--
Another useful script for those interested
On the www.generationd.com web site you will now find the asteriskcontrol
script file. This script can automatically restart Asterisk (gracefully)
following a change in external IP address - for dynamic IP hosts. As well,
it can update the
We did a custom Goldmine customer lookup popup based on Asterisk CID...but
that's about it.
What are you trying to do?
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: March-24-08 9:05 AM
To: Asterisk Users List
Subject: [asterisk-users] Gold Mine CRM +
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I had set serveremail option
to:
[EMAIL PROTECTED]
and under ast 1.2.x messages arrived at user mailboxes from
[EMAIL PROTECTED] . However, since upgrading emails
PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:41 AM
To: Asterisk Users List
Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I
to tell sendmail to trust the asterisk account or
voicemail from address
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in
1.4.18
RESOLVED! For others fiting a similar problem look at
/etc/mail/service.switch
This is the only way to force sendmail to not do a DNS lookup (first)...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users
I (like many others probably have) added the sender of the invite to my spam
filter. That avoids the many replies - and also blocks future email from
someone stupid enough to spam multiple entire list with an invite!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have an unusual and recurring problem since I upgraded to Asterisk 1.4.
Sometimes, mid-way through a call, I hear 6 shorts beeps and then the
inbound voice quality degrades massively. It sounds like the other user is
a robot...etc.
I'm guessing something (aastra 480 or Asterisk 1.4) is warning
We are installing Aastra phones (480's and 57i's) into a fairly simple
asterisk setup. Although call park pickup work fine using xfer to 700 (to
park), dial 701 (to pickup), we are unable to make the park/pickup softkey
feature work on the aastra's.
Although we've programmed the softkeys per
Are the 7921G phones convertable to SIP too?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sigma Networks
Sent: March-01-08 11:56 AM
To: Asterisk Users List
Subject: [asterisk-users] Cisco 79xx users/consultants, 7970G color in
particular share
.
On Mon, Feb 25, 2008 at 9:41 PM, OCG Technical Support
[EMAIL PROTECTED] wrote:
I'm still struggling to pickup calls. I now have a single context
(entryocginternal) where I have include = parkedcalls.
The log below shows me calling from one internal extension
to another
It looks like I have a conflict! (See results of diaplan show below). How
can I force the parkedcalls context to be matched first? (I include
parkedcalls before I define the _X. priority).
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
'_X.' =
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
OCG Technical Support
Sent: Tuesday, February 26, 2008 10:02 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
It looks like I have a conflict! (See results of diaplan
show below
I'm still struggling to pickup calls. I now have a single context
(entryocginternal) where I have include = parkedcalls.
The log below shows me calling from one internal extension to another, then
picking up, then parking the call.
-- SIP/239-0915d5c8 is ringing
-- SIP/239-0915d5c8
79 matches
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