Hi Folks,
I am running a SPA-3000 behind a legacy PABX on an analog line.
I have been able to set up a dial plan that sends outgoing calls out
to the appropriate VSP depending on prefix, and that part and the
incoming call handling works fine.
I am now trying to implement call pickup (dial 6*)
[EMAIL PROTECTED] wrote:
I've been working on a project for Asterisk for some time and it is
finally ready for a beta release. Any feedback is well appreciated. At
the basic core it's a Dialer for Windows. I'll be adding more features
quickly, but I wanted to keep everything simple and stable in
11 apr 2006 kl. 14.28 skrev Michael Strelnikov:
I do have that line. I also have all my phones defined by IP
address. But all providers are defined by names.
On 4/10/06, Michiel van Baak [EMAIL PROTECTED] wrote:On
22:14, Mon 10 Apr 06, Michael Strelnikov wrote:
Hi,
My * refuses
11 apr 2006 kl. 16.05 skrev Brent Torrenga:
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before
the Cisco
79[46]0's start to ring.
If we were lucky enough
What caching DNS do you recommend?On 4/12/06, Olle E Johansson [EMAIL PROTECTED] wrote:
11 apr 2006 kl. 14.28 skrev Michael Strelnikov: I do have that line. I also have all my phones defined by IP address. But all providers are defined by names. On 4/10/06, Michiel van Baak
[EMAIL PROTECTED]
Please use the asterisk-biz mailing list for all commercial
offerings. Thank you.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
___
--Bandwidth and Colocation provided by Easynews.com --
I have one of the Draytek USB adaptors.
Can someone point me in the right direction on how to get mISDN running with
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, 17 March 2006 12:17 PM
To: Asterisk Users Mailing
12 apr 2006 kl. 08.46 skrev Michael Strelnikov:
What caching DNS do you recommend?
Anyone you feel comfortable running.
/O
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
On 11 Apr 2006, at 23:41, Carey O'Shea wrote:
PA168S
There is a manual at:
http://www.centralitycomm.com/solutions/Download/documents/product/
PA168SUserguideEng.pdf
If I understand it, you can use the 'set' key (followed by 'speaker')
to navigate the settings menu.
I guess the trick is
OK, your solution is fine but I'd like a more generic solution to adapt it
to my current [EMAIL PROTECTED] setup.
Thanks anyway
Mimmus
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter J Dean
We do it slightly different, rather than multiple macros, we
do it within
If DNS does not work on your local network, Asterisk will lock up.
Out of curiosity - the async implementation you mentioned in the other
thread - will it replace gethostbyname with something smarter or just
run things in a different thread asynchronously?
Thanks,
Cristi
I just never used one. Is BIND good enough?On 4/12/06, Olle E Johansson [EMAIL PROTECTED] wrote:
12 apr 2006 kl. 08.46 skrev Michael Strelnikov: What caching DNS do you recommend?Anyone you feel comfortable running./O___--Bandwidth and Colocation
You can usually unlock the phone and then erase the config using the setting
sbutton. Push the setting button, nafigate to the bottom of the list, select
unlock. Use the keypad to enter the password which is cisco. Undwer network
configuraiton there is an erase configuraiton option.
Hope this
JT == Joseph Tanner [EMAIL PROTECTED] writes:
JT A slightly better (in my opinion) solution would be to code a pure
JT caching dns server, whose sole purpose is to look up specific
JT domains and resolve them to their ip address. It'll record the
JT result, and will check every so often (once a
Has anybody managed to get SipXPhone working with asterisk? I just
cannot get it to work. It just keeps reporting an authentication
failure even though all the details seem correct. The same settings
work fine in X-Lite. Failing that, are there any opensource or
reasonably priced SIP SDKs that
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are
Hi,
how do you record calls? Monitor app. or MixMonitor or something else?
How does your storage backend looks like?
What kind of channels do you use? Do you record IAX2 channels?
Regards,
Tamas
Wai Wu wrote:
You got to be kidding about 53 calls being recorded at sametime is an issue.
I have
Steve Kennedy wrote:
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's
Hi guys,
i've been using iax2 show netstats and i wonder if someone could explain what all these means, just
in case i have them wrong. Because i am looking for something that tells me that there is delay ,
and/or packet loss.
LOCAL -
Hello,
Have you tried to delete rtnoupdate and rtautoclear lines? I'm using realtime,
with caching for sip but without those 2 lines, and works perfectly.
Another point : verify that you have the field fullcontact in your realtime
sip table.
Bye,
Alban Elziere
I have two phones (111 and 112) on
Hello to all
Im looking for a billing tool for Asterisk, that works with PostgreSQL.
All the tools I found in www.asteriskbilling.com just work with MySQL :(
Do you know a nice billing tool for Asterisk with PostgreSQL?
Thanks
Joao Pereira
___
On Wed, 2006-04-12 at 07:58 +0100, Tim Panton wrote:
There is a manual at:
http://www.centralitycomm.com/solutions/Download/documents/product/
PA168SUserguideEng.pdf
Tim Panton
[EMAIL PROTECTED]
I'm now outside the network again and have run iax2 debug. Below are
the results. Notice how
Hi,We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel.
When agents are dialing, channels doesn't
Alban wrote:
Hello,
Have you tried to delete rtnoupdate and rtautoclear lines? I'm using realtime,
with caching for sip but without those 2 lines, and works perfectly.
Another point : verify that you have the field fullcontact in your realtime
sip table.
Bye,
Alban Elziere
While I
Hi,
I've tired to forward a Fax from Asterisk to Hylafax. It works so far
until I tried with a Fax machine.
I just got error shown in the log below. I'm not sure why. I've tested
it with other 6 machines and they all work fine.
Do you have any idea why?
Pim
Hylafax Session log:
Apr 12
I looked into it last year, and in Texas BRIs are only about $55/mo and include the optional calling features for which you pay extra with POTS. (Caller ID, call forwarding, etc)
The roblem I ran into was that Euro standard hardware does not work on US standard BRI lines. And I could find
Hi all,My architecture is:PSTN-E1OldPBXE1-AsteriskI've a similar problem, SIP user agents using X-Lite:Sip User Agent A calls to PSTN user BB user hangs the call
A user starts listening busy indications on the phone, and if he doesn't hangup correctly on Xlite The calls seems to be
I am using eyebeam and I am happy with it. However, it is boring just to
talk to my son in the other room.
Whenever I try to convince somebody to buy eyebeam, they are scared of
the price.
Is there a free video soft phone available, that will work with eyebeam
/ asterisk?
bye
Ronald
I'm in Houston as well. Would be very interested.
Michael
On Tue, 11 Apr 2006 21:00:49 -0500 (CDT), Aaron Daniel wrote:
I'm in Huntsville... close enough to Houston.
Aaron
On Tue, 11 Apr 2006, Lacy Moore - Aspendora wrote:
I'm in Houston.
On 4/11/06, Ryan Burke [EMAIL
Hi,
Iam using Asterisk Asterisk 1.2.5
Iam calling:
NOT OK:
phone A -ulaw - Asterik-A - gsm - Asterisk-B - g.726 - POTS phone B
NO sound from from phone A to phone B, phone B to phone A works
If iam using ulaw to connect from Asterisk-B to POTS phone B everythink is OK:
OK:
phone A -ulaw -
Hi,
Did someone experience that Asterisk OS 1.2.5 voicemail issues?
Problem description:
Some one call to the extensions 200,
After 10 sec ring then go to voicemail [EMAIL PROTECTED]
Announcement Please leave me a messages.blar blar..
When I completed to leave a message...
IF :
I press the
Bartosz,
When set up correctly the phone on your desk should ring and then when
you pick it you will be connected to the number you dialed. This is
all done via the origination command.
Did you configure the Asterisk management interface both in Asterisk
and Snap? The best approach to debugging
Mark Coccimiglio wrote:
Hey all,
It such a shame that BRI technology is such a flop in the USA. For a
small office such as mine it would be a great product. So her goes my
question What is a known asterisk working BRI card that will
operate in the USA. I need to weigh price/quality.
Hi group!
Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to
enable it, make it work? I have tried with inBandDTMF=yes in general context
of oh323.conf, but I get this message when I * is starting.
[chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver)
==
12 apr 2006 kl. 09.08 skrev Cristian Draghici:
If DNS does not work on your local network, Asterisk will lock up.
Out of curiosity - the async implementation you mentioned in the other
thread - will it replace gethostbyname with something smarter or just
run things in a different thread
[EMAIL PROTECTED] wrote:
I changed from a TE410P to a TE411P and fax carriers weren't detected
anymore !
I have tried everything (recompile zaptel+asterisk+spandsp ;
echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing
worked.
The only solution that worked for me was to
Or you can make it a bit simple in this way (number at the end of line
is your timestamp) :
[EMAIL PROTECTED] perl -le 'print scalar localtime 1112336460'
|It's a unixtime stamp. It's the number of seconds since the
|epoch(Jan 1, 1970).
|
|[EMAIL PROTECTED] wrote:
| How do I read (make
Hi,
Ie been looking for some time how to use asterisk to initiate SIP
connections between 2 IP phones, but afetr initiated the communication
making the RTP go directly from one telephone to the other, without
passing by asterisk. Unfortunately I found no explanations of how to do it.
Tiago Stein D`Agostini wrote:
Hi,
Ie been looking for some time how to use asterisk to initiate SIP
connections between 2 IP phones, but afetr initiated the
communication making the RTP go directly from one telephone to the
other, without passing by asterisk. Unfortunately I found no
I dunno if it's THAT bad. I had a BRI line in the (relatively) podunk
town of Kalamazoo, Michigan back in 1998. Sure, it took the phone
company a couple of weeks to provision the service, but it takes the
phone company a couple of weeks to do most anything in my experience.
The price was
Dear List,
We have Romania Bucharest DIDs available with area
code 4021 and 4031
For more information go to www.didx.org
Best Regards,
Oliver
Vermeulen
World Venture
Group Telecom
Tech
/ Admin
Corporate Address:
Str Avionului Nr 35/bl16J/3
Bucharest, 014333 Romania
I'm in Lubbock. A little closer to
Amarillo than Dallas.
Thanks,
Greg
From: Ryan Burke
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 11, 2006 7:14
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Texas User Group
Dear Group,
I'm working on a call recording solution and would like to have the ability to
initiate a 3 way call based on an incoming call.
One party will be an AGI that I have other will be an outbound call via a
second T1 interface.
Does anyone have a working configuration for an Asterisk
I'd like for our custom soft phone to be able to know what queue, and/or
what DID is calling an Agent's phone before the agent picks up. The
agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple
queues so it would be nice if they could get a pop up window telling
them who's on
Oliver Vermeulen schrieb:
...
We have ...
Hi,
I'm sure, there are a lot of providers of very interesting
and useful and helpful products and offers reading and writing
to this group - including our company.
Nevertheless, noone is offering his products here, because it is not
fair, if
Hi Joao,
some billing solutions are listed here -
http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems
IIRC, none works with PGSQL. My opinion is that considering the
importance of billing, it's better to develop a customised solution.
That way, you would have full understanding and
Yep, there is a lot of chatter about how hardware x performs with
Asterisk and while I/O is the primary mover, most designs today will
handle the modest Asterisk install easily. I've got a site where they
use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB
flash disk.
On 4/12/06, Steve Feinstein [EMAIL PROTECTED] wrote:
I'd like for our custom soft phone to be able to know what queue, and/or
what DID is calling an Agent's phone before the agent picks up. The
agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple
queues so it would be nice if
Check your features.conf file for conflicting key set. # is the default
key for blind transfer feature.
[]'s
MM
chan (Alpha Trilogies Networks) wrote:
Hi,
Did someone experience that Asterisk OS 1.2.5 voicemail issues?
Problem description:
Some one call to the extensions 200,
After 10 sec
RandyW wrote:
Yep, there is a lot of chatter about how hardware x performs with
Asterisk and while I/O is the primary mover, most designs today will
handle the modest Asterisk install easily. I've got a site where
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off
a
Hi,
understand that Asterisk supports a variety of signaling protocols like
SIP, IAX2 etc. As a ITSP, which would be the best or most appropiate
protocol to use as trunk to wholesale providers? Know that IAX2 can
conserve bandwidth, but I believe media and signaling are carried with
the same
[macro-hangupcall]exten = s,1,ResetCDR(w)exten = s,2,NoCDR()exten = s,3,Wait(5)exten = s,4,HangupHi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging.
I found that [EMAIL PROTECTED] is executing this Wait(5) and it seems to me
Mark,
Do you have the Flash Operator Panel or anything else installed?
I only had 1 phone stop registering in the first 2 weeks that I used
them and then after I installed FOP I had 3 phones stop registering in
the next couple of days.
I have now disabled FOP and have gone just over 2 days without
It sounds like what might be best is a Texas User group, since most of us are spread out across our great state. With Astircon 2006 coming to Dallas this year, we could all probably get together at that time. Mainly I would like to see a user group in Texas because I am deploying a wide spread
Yep, there is a lot of chatter about how hardware x performs with
Asterisk and while I/O is the primary mover, most designs today will
handle the modest Asterisk install easily. I've got a site where
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off
a 2GB flash disk.
Rich Adamson wrote:
Yep, there is a lot of chatter about how hardware x performs with
Asterisk and while I/O is the primary mover, most designs today will
handle the modest Asterisk install easily. I've got a site where
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot
off
I think this belongs to the development mail-list.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Wednesday, April 12, 2006 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Mark,
I could not get SipXPhone working
either. We've been using this SDK and really like it: http://www.worksoutsoftware.com/
The pricing is seems decent as well.
Thanks,
Greg
From: Mark Hayward [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 12, 2006
3:21 AM
Just good old monitor with no mixing onto the scsi drive.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tamas
Sent: Wednesday, April 12, 2006 4:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center
Hi,
I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup
the phone. If I want to transfer to another extension. Then I dial the #
key the system will play the onhold music. After I dial the extension
number. The system stop play onhold music and play ringtone. Is it
Kristian Kielhofner wrote:
Rich Adamson wrote:
Yep, there is a lot of chatter about how hardware x performs with
Asterisk and while I/O is the primary mover, most designs today will
handle the modest Asterisk install easily. I've got a site where
they use 6 lines and 15 users on a 500Mhz
Another solution would be to use a dedicated recording server sniffing
RTP and signalling packets in the media path using software such as
http://www.oreka.org. Oreka automatically mixes both legs of an RTP
conversation to disk and GSM encodes the result in a separate thread
so that capture always
Thanks!, I will definitely take a look at that. We were hoping not to
have to do AGI in the client, but if we have to, we have to. It'll
probably be useful for other things down the road.
-Steve Feinstein
GatherWorks Inc.
BJ Weschke wrote:
On 4/12/06, Steve Feinstein [EMAIL PROTECTED]
If you don't want to worry about * handling the full recording of all
traffic, you can potentially do this on a separate server on the RTP
path using http://www.oreka.org.
Cheers
Henri
On 10/04/06, Dov Bigio [EMAIL PROTECTED] wrote:
Hi,
I am using Asterisk for a call center on a Dual Xeon
Rich Adamson wrote:
While talking with one of the sangoma folks very recently, he was
rather emphatic the pci bus was designed to share interrupts. I was
a little concerned as a test server had the wanpipe driver sharing an
interrupt with libata and uhc1_hcd. His comment was that's the
Wai Wu wrote:
You got to be kidding about 53 calls being recorded at sametime is an
issue. I have done at least twice as many on my dual xeon 3.4Ghz system
and had no problem as clients like to record every call that goes
through the system.
Nope. We took our system to MCI's development
That may be the best idea. Unfortunately we're such a huge state that
it's going to be pretty hard to get everyone in the same room unless
there's some big event going on. Astricon may be a good time to get
together in person though.
As for the site, a simple wiki may be best, and if
Hi
I've got a
dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out
using the acopy2 test utility.
I'm having
trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where
Does anyone know if it's possible to set the codecs for a number via an
Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a
command that can set the same thing so that it can be done without having to
change sip.conf.
Essentially I want the user to
Matt Roth wrote:
These statements seem contradictory. I know of no way (short of a
custom patch) to tell Monitor() to mix the in and out legs prior to
writing them to disk. On the other hand, MixMonitor() does just that
and I believe it also buffers the writes in a way that circumvents the
i've been using iax2 show netstats and i wonder if someone could explain
what all these means, just in case i have them wrong. Because i am looking
for something that tells me that there is delay , and/or packet loss.
LOCAL -
Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn?RegardsRobOn 12/04/06, Kevin P. Fleming
[EMAIL PROTECTED] wrote:[EMAIL PROTECTED]
wrote: I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything
Rob Lith wrote:
Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it
detect the fax cgn?
Yes, that was the point of my message; with that setting, the software
tone detector will be used, just as it was before the OP's VPM got
installed.
Kevin P. Fleming wrote:
Matt Roth wrote:
These statements seem contradictory. I know of no way (short of a
custom patch) to tell Monitor() to mix the in and out legs prior to
writing them to disk. On the other hand, MixMonitor() does just that
and I believe it also buffers the writes
Hey Henri,
Long time no talk. How far have you been able to scale oreka up to?
How many simultaneous calls have you been able to record and under
what hardware config?
Thanks,
Waldo
On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote:
Another solution would be to use a dedicated
Tamas wrote:
Kevin, does MixMonitor have buffering? How big is the buffer? Is it
possible to change the size? I guess, we are talking about buffering
voice samples and writing only a bulk of them to disk (e.g. in every 50
packets - 1second).
It buffers the data in memory, there is no fixed
I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects,but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a number via an
Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a
command that can set the same thing so that it can be done without having to
change sip.conf.
Ahhh a variable. I was looking for a command. Thanks, I'll try it out.
-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 12, 2006 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting Codecs on
Simply check out the READMEs in asterisk/doc/ in your source directory.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julio Arruda
Sent: Wednesday, April 12, 2006 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Brought over from -users, Please reply to the -dev list.
I agree, lets move the discusstion over to that list as it has to be discussed
there. After we reach an accord on how it should be done we will open up a
issue on Mantis.
I see this as being two distinctive parts that would need to be
So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that
Hi Guys,
I want to playback a sound file stored in mysql database in my perl
scriptpls can anyone help with an idea? response would be
greatly appreciated
Rgds
_
Express yourself instantly with MSN Messenger!
Except that mixmonitor still has a bug in it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, April 12, 2006 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center
Look at using EAGI.
Hi Guys,
I want to playback a sound file stored in mysql database in
my perl scriptpls can anyone help with an idea?
response would be greatly appreciated
Rgds
___
--Bandwidth and Colocation provided by
On 4/12/06, Rob Terhaar [EMAIL PROTECTED] wrote:
So has anyone had any experience working with the polycom 501 or 301 and
vlans?
We run dell managed switches here, so we don't have the luxury of running
CDP to force the VOIP vlan. I haven't been able to get the polycom phones to
talk on a
On 4/12/06, Wai Wu [EMAIL PROTECTED] wrote:
Except that mixmonitor still has a bug in it.
Had. Corrected yesterday.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --
Regarding this article (1) I have one question to make. What can I do to
record the call if the agent makes a transfer using the flash button
instead of transfer button or using blindxfer or atxfer defined in
features. conf
If the agent makes the transfer with flash, the comunication between the
Wai Wu wrote:
Except that mixmonitor still has a bug in it.
What kind of bug? Issue number?
FYI: yesterday one issue has been fixed :D
http://bugs.digium.com/view.php?id=6457
Did you mean that type of bug? If something else, please let us know...
T.
-Original Message-
From:
On 4/12/06, Marco Mouta [EMAIL PROTECTED] wrote:
[macro-hangupcall]
exten = s,1,ResetCDR(w)
exten = s,2,NoCDR()
exten = s,3,Wait(5)
exten = s,4,Hangup
Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite
seems also to get delay but no crash on hanging.
I found
Hi list!
Has anyone ever tried the following installation :
I want to replace our legacy PBX with Asterisk but... I still need the legacy
PBX as a 'channel bank' for fax (I need E1 not T1)
I will put a dual port PRI card in the Asterisk box, and for incoming and
outgoing faxes I want to use
Matt Roth wrote:
These statements seem contradictory. I know of no way (short of a
custom patch) to tell Monitor() to mix the in and out legs prior to
writing them to disk. On the other hand, MixMonitor() does just that
and I believe it also buffers the writes in a way that circumvents
Yes. That's is the one. It is resolved now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tamas
Sent: Wednesday, April 12, 2006 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running
I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk
working fine for sip clients, and can call the 7960's just fine, but
I can't seem to dial out on them.
As soon as I enter the first digit, the phone attempts to dial it without
waiting for the rest. I've changed timeout
Ronald Lewis wrote:
I was alerted the other day by of all people, my mom, that she wasn't
hearing a ring when she dialed my number. Puzzled, I tried calling
myself. The call connects, but there's dead silence until voicemail
picks up. Calling internally, extensions worked perfectly. So, I
I'm unable to get the Dial option 'g' to work with callback agents. The plan is
to use it so that I can redirect a customer to a menu so they can rate the call
they just had with the agent. However, when the agent hangs up the call does
not continue in the dialplan.
I login with the agent.
.want to playback a raw binary file without writing into an
intermediate file which would increase latency
From: Alexander Lopez [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List -
We use Neos from neosmt.com to connect to our interoffice jabber server
and I noticed recently that it can do video and audio via a h.323
gatekeeper. Haven't tried it out yet but you might.
Ronald Wiplinger wrote:
I am using eyebeam and I am happy with it. However, it is boring just to
talk
Anyone out there have a functional DUNDi configuration using SIP for the
inter-Asterisk transport? I've gotten it to work with IAX2, but if I
change it to SIP it does not pass the call over even though it knows
where to send it. Thanks.
The contents of this email message and any attachments are
Have you tried something like:exten = 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})exten = 2,n,Queue(QUEUENAME)On 4/12/06,
Steve Feinstein [EMAIL PROTECTED] wrote:
Thanks!, I will definitely take a look at that.We were hoping not tohave to do AGI in the client, but if we have to, we have
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