On 3/25/2010 8:13 AM, David Gibbons wrote:
> Hi All
>
> I'm involved in discussions with my carrier right now and am wondering if
> anyone has interconnected Asterisk to Metasphere via SIP?
>
Yes, we're served by a Metaswitch usng SIP. Works fine.
-Daryl
--
_
That's what I needed to know. Thanks!
John covici wrote:
> But asterisk will not compile till you install the correct version of
> zaptel.
>
> on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
> > Correct. zaptel-1.2.12 is currently installed. I plan to inst
or you are using an older one.
>
> on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
> > I'm having a problem building Asterisk 1.2.22. It fails in
> > codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
> >
> > Here&
I'm having a problem building Asterisk 1.2.22. It fails in
codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
Here's the error. Can anyone help me with this?
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
Bill Michaelson wrote:
> Is it just me, or is the AGI interface at cnam.got-name.com failing
> for others? Anyone know how to contact them without sending postal
> mail or telegram?
I don't know how to contact them, but I am having the same problem.
__
I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not
displaying the Caller-ID number. The Caller-ID name is displayed, but
not the number. Instead, the phones always display the value that's set
in the fromuser= parameter in sip.conf. If fromuser= is not set, then
the litera
Steven wrote:
There are two I can think of.
Hoodahek and asterdex (or asteridex)
We used hoodahek at first, but now use asterdex(sp?)
It has a web interface to enter the new names into.
We use it to fixup, corp. cell phones and used to use it for our leagcy PBX
extensions.
I use some cust
I've been using the following macro to ring SIP and IAX devices for a
few seconds, and then add on a cell phone if there is no answer on the
SIP or IAX device. Periodic problems began a few versions ago and now
the problem happens every time with 1.2.9 and 1.2.9.1.
The problem is that when a
The @ip-address is actually a documented cisco "fix" to another problem.
I'd have to look it up, cause I don't remember exactly what it was, but
it's been on the list somewhere, and I think EVERYONE that's used 8.2 has
the same problem with the firmware. I would suggest using 7.4 or 7.5.
It's not just you. Same thing happens here. I went back to 1.0.7.
Stefan Gofferje wrote:
Hi folks,
I used to have some constructions like
exten => number/callerid,1,Goto(somewhere)
After updating to 1.0.8 those does not work any more.
Any hints?
Regards,
Stefan
___
Brent Franks wrote:
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb la
I've been reading drafts of this book for at least nine months and can
assure you that its content is very different than what is available on
the Wiki. The book is an excellent introduction to VOIP in general, and
offers sufficient information for the novice to configure a basic
Asterisk syste
I'm having a problem with intermittent lookup of Caller ID Name info
using LookupCIDName.
The same problem occurs when doing:
asterisk -rx "database show cidname"
No data is returned on every fourth or fifth query. No errors are being
logged.
I'm currently running CVS-HEAD-07/07/04-17:04:31 an
The default password is "Admin".
Adam Goryachev wrote:
On Tue, 2004-04-06 at 14:36, Christian Hoffmeyer wrote:
Thank you for all of the beta testing. New and improved graphics in this
release along with
drag and drop transfers and hold for all technologies.
There's a screenshot on the link belo
I contracted with Digium for this enhancement and am waiting for it to be
completed.
Tilghman Lesher wrote:
On 2004 Apr 02, at 12:04, Brian Capouch wrote:
I don't want to re-invent the wheel if someone has already
hacked a way to do this.
One of my customers has a number of stores, and he wants
What other problems are you having with RH9?
Jared Smith wrote:
Unfortunately, I've found several problems with Asterisk running on
RedHat 9. (Most of my problems only happened under high call volume.)
For that reason, we've rolled back to RedHat 8 on all of our servers.
It's worked great for
This is a known problem. I have the same situation with RH9 as you do.
I don't know if the problem has been added to the new bug tracking
system. We should check.
My workaround is to run the AGI scripts on a RH7 box and forward calls
using IAX.
Scott Stingel wrote:
Hi-
Asterisk (CVS 7/30/0
Brad's recent list of enhancements look good, but I haven't looked
at the code yet. If the code looks good, I hope it will be committed
to the project CVS.
Here's a partial list of enhancements that I would like to see in
Comedian Mail. I am probably interested in helping to fund the
enhancement
911 trunks are usually delivered to public-safety answering points (PSAP) on
analog reverse-battery facilities. (The PSAP provides battery toward the CO).
ANI is provided using MF tones. The PSAP equipment must take the ANI and use it
to submit a database query to lookup the caller's address (ALI -
I'm trying to debug a problem with robbed bit signalling on a T1
coming into an Asterisk box on a T100P card. Specifically, I need
to look at the signalling timing. Is there a way to turn on this
kind of debugging in Asterisk, similar to what 'pri debug' does?
___
DTMF, but
> then I don't think you would get ring events. Ring events for a PRI are
> in the D channel where E&M are in the robbed bit.
>
> On Sat, 2003-07-05 at 13:57, Daryl Jones wrote:
> > I am trying to make an in/out trunk group comprised of 4 DS0's using
>
I am trying to make an in/out trunk group comprised of 4 DS0's using
E&M Wink signalling. The first four channels of a DS1 on a T100P
are being used for the group. Outbound calls work fine, but inbound
calls fail. The other 20 DS0 channels are used for a PRI. Does the
configuration shown below l
Yes, but I have been able to mitigate it by setting the following
parameters. I have the problem with ATA's that are behind firewalls
and not, but mostly with the ones that are behind firewalls.
CfgInterval:1800
SIPRegInterval:100
On Thu, 3 Jul 2003, Kim C. Callis wrote:
> Is it just me or do
I started using Festival for the first time today and am having a problem
with zombies left behind after every time that it speaks. I'm using
Festival 1.4.3 with today's CVS of Asterisk. Everything seems to work.
The only obvious problem is that a defunct process is left behind every
call to Fest
Should Asterisk run under it's own user id, or the web server user id,
or root, or what?
___
Asterisk-Users mailing list
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I'm running a pretty successful Asterisk system and recently moved our
PRI to a T100P board. The PRI was previously connected to a Cisco 2600
that was serving as a voice gateway. We are having a frequent problem with
inbound and outbound calls being disconnected shortly after they are
answered sin
This doesn't work for me. Voicemail says the extension number but
does not play the user's name. (Asterisk CVS-04/30/03-22:57:49)
On Tue, 17 Jun 2003, Benjamin Miller wrote:
> When in voicemail they need to go into the "record name" section and
> record their name. Then it will play their name
Is there a way to configure voicemail to do reminder paging? I would like
to configure some voicemail boxes to send an e-mail message to a pager
every 10 minutes until the message is retrieved.
___
Asterisk-Users mailing list
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I experienced the exact same symptoms but didn't have the confidence
to post my experience to this list because of my lack of experience with
Asterisk. I restored the June 1 version from CVS and the problem went away.
There's definitely a problem in code since June 1.
On Sat, 7 Jun 2003, John To
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