Sorry, I am tired and missed the virtual IP part. I am not quite sure
what that means or why you are sending traffic to the routeable IP.
Are you using a FQDN with external DNS or the IP in your client?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 10:42 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote
Pete,
I have never done it but it would seem that running a SIP client on
your SIP server may be problematic.
Also, 58.251.75.228 is certainly not on your 192.168.x.x subnet. Is
your machine dual homed?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 9:34 PM, Carlos Rojas <[EMAIL PROTECTED]> wro
Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
> I am sorry my questinos are too fundamental. I am new to Asterisk, and
> hope to catch up as fast as I can.
>
That probably includes 5 years of support but still expensive.
John Faubion wrote:
>> Although this is a "users" list, I think it is more of a list for
>> Asterisk "resellers". I'd be interested in how many of you are simply
using Asterisk as your phone system and NOT selling your services or an
On Thursday 20 March 2008 05:06:29 am Mian M Asif wrote:
> Hi eric,
> can you please tell me how can i save the value of EXTEN in a different
> variable before the Goto(s-${DIALSTATUS},1),
>
> thanks for you help,
>
> regards,
> Asif
>
>
> Message: 14
> Date: Wed, 19 Mar 2008 10:39:22 -0500
> From:
On Thu, 20 Mar 2008, Norman Franke wrote:
> On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote:
>
>> Sure some others on here may disagree, but I am also over on the trixbox
>> forums, and have often seen talk about the 2.6.9 kernel having interrupt
>> issues, and such that cause asterisk issue
Hello All,
I've been trying to get BLF working with Asterisk 1.6-LatestBeta, and My
Cisco 7970 (Latest SIP Firmware).
Has anyone successfully completed this?
I got the patch to merge in from
http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones
With a bit of hackery to tha
Al Baker wrote:
> Quote"
>
> This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
> have about 1,800 DID numbers pointed at it, "
>
> Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into
> that box, 1,800 DIDs pointing to it sems like
> one hell of a c
On Thu, Mar 20, 2008 at 09:09:05PM +0200, wassim darwish wrote:
>
> hi:
> In my zapata.conf i have 4 fxo configured channels,for fxo number 1
> to 3 i added polarity reversal property but for fxo number 4 i didnt
> add polarity reversal property but it still giving me on cosole that
> fxo numb
Steve Totaro wrote:
> On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account <[EMAIL PROTECTED]>
> wrote:
>
>> Al Baker wrote:
>> > Quote"
>> >
>> > This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
>> > have about 1,800 DID numbers pointed at it, "
>> >
>> > Are
hi:
In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i
added polarity reversal property but for fxo number 4 i didnt add polarity
reversal property but it still giving me on cosole that fxo number 4 is
polarized (because the line on fxo number 4 is not polarized).
what
Mojo with Horan & Company, LLC wrote:
> [EMAIL PROTECTED] wrote:
>
>> I am planning to write a module to find if a Special Information was
>> detected or not.
>>
>> Can anyone please help me to figure out the below fields?
>> 1. The Frequency of a frame
>> 2. Length of frame in milliseconds
>
On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account <[EMAIL PROTECTED]>
wrote:
> Al Baker wrote:
> > Quote"
> >
> > This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
> > have about 1,800 DID numbers pointed at it, "
> >
> > Are you SURE on that figure. Since you c
[EMAIL PROTECTED] wrote:
> I am planning to write a module to find if a Special Information was detected
> or not.
>
> Can anyone please help me to figure out the below fields?
> 1. The Frequency of a frame
> 2. Length of frame in milliseconds
>
Aren't all the frames in asterisk 20ms long, no
On March 20, 2008 02:33:52 pm Anselm Martin Hoffmeister wrote:
> Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
> > And what happens if at the time of the shutdown there was a
> ROTFL
> Trafrir, you made my day.
Oh god, I didn't realize that wasn't a typo until you wrote that..
Mian M Asif wrote:
> Hi eric,
> can you please tell me how can i save the value of EXTEN in a different
> variable before the Goto(s-${DIALSTATUS},1),
>
exten => s,n,Set(OLD_EXTEN=${EXTEN})
Then later, just use ${OLD_EXTEN}
___
-- Bandwidth and Coloc
hello,
you have to use following format in den extension key of the snom:
|*7
the |*7 is the extension to dial if you want to pickup the ringing
(blinking) line.
maybe you should try |*7 where 100 is your hint
extension
and *7100 is a defined pickup extensions.
best regards.
Steve Smith
L
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
> And what happens if at the time of the shutdown there was a
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
ROTFL
Trafrir, you made my day.
(BTW: I t
No, I meant if I leave this office, what to do when the cpu fan or power
supply breaks on our current * box :) They might just be so worried
that they'd *want* something like the 3Com V3000 :)
Steve Totaro wrote:
> Call your dealer as I am sure you would have a support contract.
>
> Haven't rea
> For such a small system there is no earthly reason for it to
> be 10 percent of that, even on a 5 year lease.
> I know that EVERYTHING is big in Texas, but that is nothing
> more than highway robbery.
I fully agreed, that's why we built her an Asterisk based system. Splitting
this up they want
Hello,
I am having some troubles with Snom phones and maybe someone can help
me.
Let me say this: BLF and pickup works great with Polycomes and
Grandstream etc... So I think my problem might not be Asterisk related
but I am not 100% sure.
The snom phones subscribe to my extensions (hint priority)
Hi,
I'm getting this error when registering with SIP server using Asterisk
1.4.10 and Freepbx...
I'm getting this error no matter what I try to setup in sip.conf :
- I'm getting confused whether options are maxexpirey=36000 or
maxexpiry=36000 ?
- Can I solve this with some settings in sip.conf
Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
> Holy Mackeral. Ignore that last message. I still do NOT know how to
> route calls with the same extension being used in two locations,
> however the issue I've resolved is getting Cisco CallManager and
> Asterisk talking together p
John Faubion wrote:
>> Although this is a "users" list, I think it is more of a list for
>> Asterisk "resellers". I'd be interested in how many of you are simply using
>> Asterisk as your phone system and NOT selling your services or an Asterisk
>> based solution?
>>
>
> I actually work
To add some further details to this thread I set up a "Monitor" command
that records just the IVR portion of an incoming call. I left the m
flag off so I could listen to the incoming audio separate from the
outgoing recording. On calls where the DTMF detection works correctly I
only hear extr
List,
Question about the Polycom 650: when dialing the digits for a phone number,
and an incoming call comes in, does the phone prevent you from completing
your outgoing call until the phone stops ringing, like a Cisco 79X0 does?
--Brent
___
-- Band
Thank you all for the great advice. Although fairly new to Asterisk, and
relearning systems administration, it has helped put some perspective on
the matter for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 19, 20
On Thu, Mar 20, 2008 at 11:10:21AM -0400, Norman Franke wrote:
> On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED]
> wrote:
>
> > Sure some others on here may disagree, but I am also over on the
> >trixbox
> >forums, and have often seen talk about the 2.6.9 kernel having
> >interrupt
> >issues
I was running Trixbox 2.2 up until about 2 months ago, and had persistent
interrupt issues. I upgraded to 2.4, with the updated kernel, and its been
complete smooth sailing ever since.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Norman Franke
Sent: 20 March 2008 0
Still grasping at straws trying to solve DTMF detection issues with one
of my asterisk servers. This particular server is now running Asterisk
1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P
cards. I have tried adjusting channel gains, turning call progress and
relaxdtmf
On Thu, 20 Mar 2008, Tzafrir Cohen wrote:
> On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW
> +1-303-722-7209 wrote:
>> Hi,
>>
>> When I build the same asterisk package that I build on i386 on
>> x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same
>>
On Mar 19, 2008, at 5:56 PM, [EMAIL PROTECTED]
wrote:
Anyone? Just a user?
I'm just a user, although I also develop things for internal use.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
___
-- Bandwidth and Colocation Provi
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED]
wrote:
Sure some others on here may disagree, but I am also over on the
trixbox
forums, and have often seen talk about the 2.6.9 kernel having
interrupt
issues, and such that cause asterisk issues. One reason I think
they moved
forward int
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to "check" to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten => _4xx,1,Dial(SIP/${EXTEN}&IAX2/
Holy Mackeral. Ignore that last message. I still do NOT know how to route
calls with the same extension being used in two locations, however the issue
I've resolved is getting Cisco CallManager and Asterisk talking together
properly.
Sorry folks AGAIN.
So if anybody has ideas on how to have exten
On Thu, Mar 20, 2008 at 09:31:03AM -0500, John Faubion wrote:
> > I reboot every evening :) Drew, what's the uptime on your
> > asterisk process on that box that's been up for 193 days?
>
> I too restart the asterisk process every night as part of the cron process.
> Many people here seem to be
> Although this is a "users" list, I think it is more of a list for
> Asterisk "resellers". I'd be interested in how many of you are simply
> using Asterisk as your phone system and NOT selling your services or
> an Asterisk based solution?
I actually work as a software engineer for a big tele
> >
> > Perhaps in a similar thread, is it possible to somehow SET the state
> > of a hint from the dialplan? Perhaps a bit like:
> > Set(${ChanIsAvail(hint,234)}=Busy)
> > or perhaps have a pseudo-device facility where you can add
> it to the
> > end of the hint list to "hint-the-hint".
On 3/20/08, Steve Davies <[EMAIL PROTECTED]> wrote:
> On 20/03/2008, Johansson Olle E <[EMAIL PROTECTED]> wrote:
> > 20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
> > > hello,
> > >
> > > i am trying to set up a asterisk server (version 1.2.26 by now) with
> > > realtime configuration but t
> I reboot every evening :) Drew, what's the uptime on your
> asterisk process on that box that's been up for 193 days?
I too restart the asterisk process every night as part of the cron process.
Many people here seem to be under the impression that restarting the
application every day is a bad
On 20/03/2008, Johansson Olle E <[EMAIL PROTECTED]> wrote:
> 20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
> > hello,
> >
> > i am trying to set up a asterisk server (version 1.2.26 by now) with
> > realtime configuration but the user shouldnt register directly to the
> > server, instead i have
Must be having a "DOH!" week.
Problem turned out to be the Fedora core firewall that was turned on.
Sorry folks.
On Wed, Mar 19, 2008 at 3:01 PM, Aaron Fransen <[EMAIL PROTECTED]>
wrote:
>
> Finally got my Cisco Call Manager link going; what it turned out to be was
> having the same extension o
MeetMe() has the K option that kills the conference,
how do I do that in app_conference() as there no kill the conference option?
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIB
Appologies for top-posting. This is the most interesting thread in a
long time. Alex, yours is the most well considered opinion I've seen in
a long while. I exactlt reflects my own, moerw limited experience.
Thank you for chiming in.
Two weeks ago on the VOIP Users Conference weekly call we had a
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
> hello,
>
> i am trying to set up a asterisk server (version 1.2.26 by now) with
> realtime configuration but the user shouldnt register directly to the
> server, instead i have set up a ser registration proxy. Everything
> works
> fine so far, but i
- Original Message -
From: "Lee, John (Sydney)" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, March 19, 2008 11:48 PM
Subject: [asterisk-users] Newbie IVR: How to read() before playback()
isfinished?
>I am working on a menu to accept input from a caller like as follows:
>
> Exten => 100,1,A
On 3/20/08, Stefan Schmidt <[EMAIL PROTECTED]> wrote:
> hello,
>
> i am trying to set up a asterisk server (version 1.2.26 by now) with
> realtime configuration but the user shouldnt register directly to the
> server, instead i have set up a ser registration proxy. Everything works
> fine so fa
Andreas Sikkema wrote:
> I've literally got _thousands_ of users and Asterisk is rock
> solid for us.
>
>
I think most of the instabilities are from the use of queues and
mixmonitor/chanspy.
I don't use either and have no real issues. I still restart the
Asterisk service once a week though,
On Thu, Mar 20, 2008 at 06:45:14AM -0400, Alex Balashov wrote:
> Tzafrir Cohen wrote:
>
> > Yeah, right. And we have no SIP compatibility issues at all. It is also
> > funny that you reflect the quality of old PRI card of one company and
> > yet ignore all the past mishaps of SIP devices.
>
> Oh,
> Although this is a "users" list, I think it is more of a list
> for Asterisk "resellers". I'd be interested in how many of you
> are simply using Asterisk as your phone system and NOT selling
> your services or an Asterisk based solution?
I'm responsible (development, maintenance, support) for
On Wed, 19 Mar 2008 16:38:23 -0500, "Bill Andersen" <[EMAIL PROTECTED]>
wrote:
> Although this is a "users" list, I think it is more of a list
> for Asterisk "resellers". I'd be interested in how many of you
> are simply using Asterisk as your phone system and NOT selling
> your services or an As
Tzafrir Cohen wrote:
> Yeah, right. And we have no SIP compatibility issues at all. It is also
> funny that you reflect the quality of old PRI card of one company and
> yet ignore all the past mishaps of SIP devices.
Oh, no, I didn't mean to imply that. There are plenty of SIP interop
problems
Gordon Henderson wrote:
> On Wed, 19 Mar 2008, Norman Franke wrote:
>
>
>> As for why a company would purchase hard phones, several reasons. First, we
>> are replacing many hard phones with computers. We have a custom application
>> and have been moving folks main numbers to use the computer.
On 3/20/08, Tobias Ahlander <[EMAIL PROTECTED]> wrote:
> >Date: Wed, 19 Mar 2008 11:31:57 +0200
> >From: "Atis Lezdins" <[EMAIL PROTECTED]>
> >Subject: Re: [asterisk-users] Handling 3 different call ending causes
> >To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> >Message-I
In article <[EMAIL PROTECTED]>,
Lee, John (Sydney) <[EMAIL PROTECTED]> wrote:
> I am working on a menu to accept input from a caller like as follows:
>
> Exten => 100,1,Answer()
> Exten => 100,n,Playback(LONG-MESSAGE)
> Exten => 100,n,Read(OPTION,,2)
> ...
>
> When I tested it, I noticed if I sta
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] How to configur
I am using asterisk 1.4.18 (server A ) and have it store records in
mysql database . One of my client uses predictive dialer ( asterisk
1.2.26 based and server B ) which makes many calls . B registers with
A over sip and there is no nat involved If i re-invite rtp from
server B to my carrier ( s
Hello,
You have to set up a hint extension pointing to the Sip user like exten
=> 777,hint,SIP/username
That extension is used in the Snom as "extension".
if you use the following format of this option field you should be able
to pickup: |*9
777 is the hint extension
127.0.0.1 is your server ip
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything works
fine so far, but i can´t use the hint feature. Its possible to subscribe
to
On Thu, Mar 20, 2008 at 01:27:47AM -0400, Al Baker wrote:
> From a lot of experience - you are not being anywhere near paranoid
> enough !!
> Think dual RAID controllers, Dual power supplies off of, at a Minimum,
> separate isolated circuits, with Hefty UPS that is in-line so it filters
Excellent topic and points brought up by all!
On Thu, Mar 20, 2008 at 8:43 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> Think of Asterisk not as a PBX but as a PBX toolkit. Various people in
That's always been the way I saw asterisk. I wondered why people
sometimes try to interface it with lega
On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW
+1-303-722-7209 wrote:
> Hi,
>
> When I build the same asterisk package that I build on i386 on
> x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same
> set of installed packages. What should I be loo
On Thu, Mar 20, 2008 at 12:59:08AM -0400, Alex Balashov wrote:
> At the risk of inflaming a lot of passions, including those of
> hard-working developers, I must say that where Asterisk may be
> production-worthy, the entire constellation of things (like Zaptel) of
> which its PSTN hardware int
>Date: Wed, 19 Mar 2008 11:31:57 +0200
>From: "Atis Lezdins" <[EMAIL PROTECTED]>
>Subject: Re: [asterisk-users] Handling 3 different call ending causes
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
>Message-ID:
><[EMAIL PROTECTED]>
>Content-Type: text/plain; charset=ISO-8
On Thu, Mar 20, 2008 at 01:09:36AM -0400, Al Baker wrote:
> Not sure if this is the best place to ask this or not...but since it was
> mentioned..
> Is "SwitchVox" a alternative to * ?
> Were they a competitor to *, and DIGIUM bought them and so DIGIUM
> has 2 Totally Different PBX software pack
On Wed, 19 Mar 2008 10:10:21 + (GMT), Gordon Henderson
<[EMAIL PROTECTED]> wrote:
>> I got free installation for Featureline Compact
>> on 3 yr contract.
>> So it saved me £££s!
>
>Intersting... But shouldn't you be using VoIP for your calls anyway...
>Then just one basic BT line, and a busin
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