Displaying 20 results from an estimated 7000 matches similar to: "ViaTalk Down?"
2005 Sep 12
1
Other Voicemail systems
Since I can't seem to get anything figured out for the Comedian system, are
there any other systems out there that we can hook asterisk into?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
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2005 Aug 27
1
dtmf not being detected from viatalk
I am using viatalk as my voip provider and they use dtmf=rfc2833, but
asterisk is not seeing any of the dtmf. I am using CVShead as of
8/26/05. Nothing in the logs indicates a dtmf is being seen. If I
use my pots line it sees it fine.
Any assistance would be appreciated.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
2005 Aug 26
2
SIP Benchmarking / Stress Testing
Anyone have a good tool(s) to use for simulating a bunch of calls?
Benchmarking or stress testing?
I only need SIP protocol, and do appreciate any replies...I realize I could
google it, but I am looking for opinions as well.
Sherwood McGowan
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2005 Aug 22
1
Delete function in realtime voicemail?
since delete is a reserved word, what do you name a column in your voicemail
options table to allow setting of the delete option for realtime voicemail?
Anyone?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
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2011 Apr 05
1
asterisk-users Digest, Vol 81, Issue 12
On 04/05/2011 03:06 PM, asterisk-users-request at lists.digium.com wrote:
> Message: 12
> Date: Tue, 5 Apr 2011 13:36:21 -0500
> From: Sherwood McGowan<sherwood.mcgowan at gmail.com>
> Subject: Re: [asterisk-users] Iptables configuration to handle brute,
> force registrations?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at
2008 Jun 03
8
Any reason to *not* use AEL? (Also, MixMonitor q)
I am building a new Asterisk server here at the office, and I'm
wondering if there are any downsides to creating my dialplan with AEL.
It seems more intuitive (to me), but I'm not sure if there are any
pitfalls I need to be aware of first.
We use this for internal extensions, 8 pots lines, and our answering
service which gets about 500 incoming calls a day down our T1.
Also, one more
2011 Mar 28
2
Variable. AMI and dialplan
Hi!
Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.
2005 Sep 06
4
Sipura Devices and Asterisk?
I'm currently using the Linksys PAP2, and since there's a shortage I'm
looking for different devices. I'm mainly looking at the Sipura SPA sets
since they are the base of the pap2. Anyone else have experience using them,
and which one?
Thanks
Sherwood McGowan
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2008 May 23
2
Strange State 6 on Channel X
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make
a call into the system, the system claims to answer the call, and do
the things in the dial plan, but I just hear ringing on the phone I'm
calling in from.
I am using a Sangoma A200 4 Port Analog card.
my wanrouter version: WANPIPE Release: 3.3.6
asterisk -V: PBXtra Core fon_o_1.2.17
Any ideas?
Daniel Lockard
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
2005 Sep 02
3
DTMF and "breaking through" voice prompts
Has anyone else had problems with users being able to press key tones during
a voice prompt? I have a few users complaining that some systems will not
recognize key presses during them.
using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode.
Thanks
Sherwood McGowan
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2005 Aug 17
0
sip.conf user entry for ViaTalk
Try as I might, I can not get incoming calls from ViaTalk to match
against my user entry. I have both peer and user entries, and incoming
and outgoing calls work, but incoming calls do not move to my in-viatalk
context (they stay in the default context.) Has anyone else managed to
get this to work? My user entry looks like:
[viatalk-in]
username=1407965XXXX
context=viatalk-in
type=user
2008 Jun 13
1
AEL Help
I need help translating extensions.conf to AEL:
[default]
exten => _X.,1,Set(DID=${EXTEN:6})
exten => _X.,n,Goto(continue,1)
exten => _1X.,1,Set(DID=${EXTEN:7})
exten => _1X.,n,Goto(continue,1)
exten => continue,1,Noop(${DID})
exten => continue,n,Set(GROUP(IAX)=incoming)
exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at
2011 May 10
14
When someone helps you, at least let them know if the problem is resolved or not
I'll keep this brief because I don't want to come across like any more of an
a$$ than I absolutely have to, especially since I know I've blown my stack
before.....
Gentlemen (and Ladies, if you're out there),
If someone gives you advice on this list, and ESPECIALLY if they give you
advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if
you get your question
2005 Sep 30
2
Asterisk and RTP streams
Guys, I've been poking around trying to find a good answer for this via
voip-info, google, etc... Haven't found anything that helps, so maybe you
mates could.
A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using
Sipura SPA-2002s. Every once in a while, the customer will get one-way
audio. I've read that this is commonly caused by the outgoing RTP port not
2008 Mar 26
2
DTMF suddenly stopped working on SIP channel
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
registered in the CDR or on the console in verbose level 10, it just
times out. I haven't changed anything on my part and can't get through
to Viatalk tech support to ask them
2011 Feb 12
1
Variables losing their value????
Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be empty...I've googled and found nothing stating this kind of weirdness..
Asterisk 1.8.2.2 (upgrading
2005 Sep 06
1
Routing depending on sip response code?
Hey all,
I'm trying to create redial on busy for my users, but haven't the foggiest
on how to make asterisk route depending on the status code returned over SIP
(483, Busy Here?). . . anyone know how to do this?
Thanks
Sherwood McGowan
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2008 Jul 29
5
Callerid Woes
I am trying to setup one time caller id block on my system(activated
when an incoming call matches *811XXXXXXXXXX), and I have had little to
no luck. Could you take a look at my context/macro definition and help
me figure out what I am missing?
Here is my context for my dialplan:
include=default
plancomment=user-default
2010 Oct 14
1
MySQL and Channel Event Logging
Hey all, sorry if this has been covered, but I've not found anything after a
couple hours' worth of googling. I can see (and I'm familiar with) all the
usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any
reference to MySQL and the new CEL logging tool other than ODBC. Is this the
only method available to use MySQL with CEL at this time?
Thanks,
Sherwood