Hi Rchard,
I have the feeling OptiPoint phones are distributed through Siemens
integrators which, here in France, are few and not inclined to quote and
distribute these products without any Siemens PBX.
How (and where) could you provision those phones ?
Do you have any support from Siemens or
Use
qualify=3000
For an acceptable lag of up to 3 seconds. That value _doesn't_ mean to
ping the peer every 3 seconds, btw. By default, It will be pinged
every 60s if ok, and every 10s if there is any problem (peer lagged,
unreachable, etc).
Julian.
On 1/4/07, Eric ManxPower Wieling [EMAIL
Hi,
We need some one who can install hylafax on our Asterisk box and
configure it with Sip DIDS.Please PM mail me if you are interested or
want any further clarification.
Thanks,
Mantra
___
--Bandwidth and Colocation provided by Easynews.com --
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco
So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go
out H323 or SIP to Cisco, where
How (and where) could you provision those phones ?
Do you have any support from Siemens or anyone ?
We have a HiPath4000 V1.0 interconnected with Asterisk
using oh323. I have flashed several OptiPoints (from
the HiPath) to SIP firmware. But again OptiPoints seem
to work well with Asterisk but
Yes, unfortunately they want to run a queue for each client, who has
different agents located etc. This is a bit like a true Skill-based
routing using Asterisk standard mechanisms. Apart for the business
problems, I wondered if there are known problems with Asterisk running so
many
You are correct, this is more or less the scenario involved - the problem
is that people want to call a personalized line AND speak to the same
subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or 40
queues, never 200 - so I was wondering if
Thanks!
I can´t rely figure out how to block for only one extension.
Eg. Extension 209 need to be blocked from making calls starting with 070
(eg. 9070).
Some clues did I get bout would it men a new form-internal-blocked dialplan?
Regards
Mattias
On 04/01/07, C F [EMAIL PROTECTED] wrote:
The
On Thu, 04 Jan 2007 11:05:38 +0100
Lenz [EMAIL PROTECTED] wrote:
You are correct, this is more or less the scenario involved - the
problem is that people want to call a personalized line AND speak to
the same subset of agents preferably.
I have never seen such a setup myself - I have seen
Hi All,
I'm about to modify my menu to allow for certain callers to bypass my
menu entirely, and just ring my phone. I don't need friends hearing my
business menu all the time.
So, my idea was to use my already enabled callerid asterisk database
and lookup the callerid info from it, if it
Hi moises,
Hi i need to done some modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
and also let me know which editor suitable it ,I'm using suse linux..
Plz help me reg. this ..
-nsthi,
Lenz wrote:
You are correct, this is more or less the scenario involved - the
problem is that people want to call a personalized line AND speak to
the same subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or 40
queues, never 200 - so I was
Lenz wrote:
You are correct, this is more or less the scenario involved - the
problem is that people want to call a personalized line AND speak to
the same subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or 40
queues, never 200 - so I was
Gregory Machin wrote:
Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an answering service and
record messages ?
Asterisk is probably abit overkill for this. Any modem that supports a
decompress audio command can usually do this.
Hi list:
This is my first post and first off all i want to wish a good
year for everone! well my problem is; i already installed asterisk on
a server and created a channel and a couple of extensions, all seems
to work just fine, y can make calls and receive them, i'm using the
x-lite
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first off all i want to wish a good
year for everone! well my problem is; i already installed asterisk on
a server and created a channel and a couple of extensions, all seems
to work just fine, y can make calls and receive
Most SIP phones handle this functionality by recognizing numbers from
speed dial or address book entries in the phone itself. I believe
that the PolyCom SIP phones do this (IP430, IP501, IP601, IP650).
I hope that this is helpful.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
On Thu, Jan 04, 2007 at 05:14:30AM -0500, Matt Gibson wrote:
Hi All,
I'm about to modify my menu to allow for certain callers to bypass my
menu entirely, and just ring my phone. I don't need friends hearing my
business menu all the time.
So, my idea was to use my already enabled callerid
Hi,
Recent addition to Avaya 4600 Series enabled such phones to 802.1x.
http://www.avaya.co.uk/gcm/emea/en-us/products/offers/4600_series_ip_telephones.htm
As you can see, it's not so easy to guess if 802.1x support is offered along
SIP firmware.
Cheers
Eric ManxPower Wieling wrote:
Arik Raffael Funke wrote:
I am having a problem that is a miracle to me: If I dial out via
voipstunt.com the call rings for a few seconds and then gives me a
busy sign.
Start out with not using the r option to the Dial line. That will
remove the faked ringing
Hello and thanks for a reply. If I understand correctly, maxlen parameter
limits total number of people waiting in queue, and I don't want to limit this,
it just seems strange to me that asterisk lets ring all waiting people
even on busy agents)
On 2007-01-03, Ex Vitorino wrote:
Nikola,
Hi list,
Is Digium Wildcard B410P compatible with Asterisk 1.2 / 1.4 (zaptel of
course) directly out of the box, or do I need things like bristuff?
http://www.digium.com/en/products/hardware/b410p.php
Best regards,
Henrik Woffinden
___
--Bandwidth
My GXP2000 does what you are talking about. I solved the problem by
assigning lines 2-4 to other extensions which are not queue agents.
Then those lines don't ring.
hth
-t-
On Jan 2, 2007, at 5:03 PM, Nikola Ciprich wrote:
Hello,
I'm using asterisk queues, for reception phone, and I
I need to retrieve my asterisk to retrieve a values from postgresql, i am
looking for some sort of application like *mysql*() app, I found one but it
is only available on Suse, is there any way for doing this?
Regards,
O.Youssef
___
--Bandwidth and
You'll need mISDN
A small tutorial (in dutch): http://www.blicbox.be/node/22
You should be able to translate using: http://babelfish.altavista.com/
Timothy.
Henrik Woffinden wrote:
Hi list,
Is Digium Wildcard B410P compatible with Asterisk 1.2 / 1.4 (zaptel of
course) directly out of the
007/1/4, Bob Chiodini [EMAIL PROTECTED]:
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first off all i want to wish a good
year for everone! well my problem is; i already installed asterisk on
a server and created a channel and a couple of extensions, all seems
O.Kamal wrote:
I need to retrieve my asterisk to retrieve a values from postgresql, i
am looking for some sort of application like *mysql*() app, I found one
but it is only available on Suse, is there any way for doing this?
Regards,
O.Youssef
What do you need to do?
To get an SQL console
HI Gavin,
wish we could do that! :) the problem is that they want to have
personalized agents too - so that each client has its own line AND his own
agents, so that they get back to speaking to the same people all of the
time. SO we need many different queues to accomodate all those
chester c young wrote:
does anyone know if ztdummy is requires under 1.6 or are they using
Linux' rtc?
If you really want to use the rtc as a timing source, there is zaprtc. I
don't think it's digium supported and I don't even know if it works (I
don't use it).
The best bet (If you've got
On Wed, Jan 03, 2007 at 04:51:23PM -0600, John French said:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet 24
port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.
We are doing this here is RSA.
Please apologize but what does RSA stand for ?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
It sounds very promising, I will have to check it out.
Does it edit conf. files? Or is there just one lead in conference number that
flows into your app.?
I am currently using freepbx, and do not want to break anything.
--
--
Steven
http://www.glimasoutheast.org
Dan Austin [EMAIL
Please note that trunk is currently 1.5
Evens are stable, odds are development.
trunk is technically 1.5 until 1.6 is released.
--
--
Steven
http://www.glimasoutheast.org
chester c young [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
does anyone know if ztdummy is requires
Is it an external serial modem ?
any tips on how to get it to work ?
Thanks for you time ..
On 1/4/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Gregory Machin [EMAIL PROTECTED]
Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an
For 1.4, you can try the trunk patch, it may work, PLEASE READ the
bugtracker, there someone seems to have tested this in 1.4.
On 1/3/07, chester c young [EMAIL PROTECTED] wrote:
how is this fitting into 1.4?
- can it be compiled against 1.4 or only 1.2?
- if not, are there leanings in that
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Does anyone know if misdn and the B410P is working yet in kernel
2.6.18/19?
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL
On Thu, Jan 04, 2007 at 05:12:54PM +0200, Richard Soderblom wrote:
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Does anyone know if misdn and the B410P is working yet in kernel
2.6.18/19?
Hi
actually, for call i use ZAP Channels on a E1 and SIP Account on a VoIP
provider ...
in Zap, we use group and we have:
exten = _1.,2,Dial(Zap/r1/${EXTEN:1},50,rt)
exten = _1.,3,Hangup
r1= he change of channels at all calls channel group 1
It's possible to create a
I've been looking through everything I can find and observing the mysql
logs and I don't see password changes passing through to the DB. Is
that correct?
--
One day at a time, one second if that's what it takes
___
--Bandwidth and Colocation
Hi,
how do i have to specify the key's in misdn.conf? Does it even work in
asterisk 1.2? When i try to do an encrypted call it get's rejected because i
have 0 keys but specified key index 1?!
Regards
Andreas
_
Need more speed?
Hello
Do no forget the rtp ports 1 to 2
Regards
On 1/4/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote:
007/1/4, Bob Chiodini [EMAIL PROTECTED]:
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first off all i want to wish a good
year for everone!
Technically, there is no such thing as 1.5. svn trunk doesn't get a release
version number. trunk is simply trunk
- Steven [EMAIL PROTECTED] wrote:
Please note that trunk is currently 1.5
Evens are stable, odds are development.
trunk is technically 1.5 until 1.6 is released.
--
Hey Everyone,
So this is a problem I've been having for sometime now. I sent a few
messages to the list with no luck.
The problem is that when people dial into the Asterisk system using DID
numbers, it works the first time or 2, then I get busy signals.
A friend recommended I clear out the
Hi,
if i use System() or TrySystem() into my extensions.conf for execute a
external command, can i get and put the result of the command into a
variable ?
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Correction in my zapata.conf file I used
Hey Everyone,
So this is a problem I've been having for sometime now. I sent a few
messages to the list with no luck.
The problem is that when people dial into the Asterisk system using DID
numbers, it works the first time or 2, then I get busy signals.
On Jan 4, 2007, at 7:18 AM, Henrik Woffinden wrote:
Hi list,
Is Digium Wildcard B410P compatible with Asterisk 1.2 / 1.4 (zaptel of
course) directly out of the box, or do I need things like bristuff?
Using the current version of Zaptel/Asterisk it should out of the box
just fine.
Matthew
David Gomillion wrote:
We bought 7 switches, and 3 of them failed after one year. It took quite
a bit of doing to get the off-shored customer support to read their own
literature to cover one switch under warranty. Never could get the other
two covered...
When I got the warranty status
One thing you might want to consider is what distribution strategy to use for
the queues. Note that Asterisk Queues use FIFO methodology to handle calls and
distribute them to the agents.
Say you have one queue and round robin strategy. When 5 calls come in and only
2 agents are available, the
Thanks for the input
not seeing any evidence of disconnect supervision or callerid but
you're right, depends on provider ---
Anyone have any luck with provider MTNL?
--
Chris
Rajkumar S [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On 1/1/07, ram [EMAIL PROTECTED] wrote:
Douglas Garstang wrote
*snipped
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
*snipped
a quick run through of of app_queue.c (my copy) for anything directly
dealing with a reload
shows tmp in use for realtime
later a reference for convert to
Tell us about channels 45-47...
We have 45-47 defined in zapata.conf, but not in zaptel.conf. Probably not
the problem, but it might cause some confusion...
Also, look at your timing sources. They don't look quite right to me. Spans
1 and 2 are both marked as primary timing sources.
Hope that
Sorry, they were the NetGear 24-port rack-mount 1U switches (7326P is the
part number, if I remember correctly). It's the brand that was mentioned
right before my reply...
On 1/4/07, Allen Casteran [EMAIL PROTECTED] wrote:
David Gomillion wrote:
We bought 7 switches, and 3 of them failed
Hi Thiru -
Hi i need to done some modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
I would not recommend trying to compile only small pieces of asterisk.
It is broken up into many sections,
Followup on this issue, it appears that using a single PRI's clock as the
master clock avoids clock drift between the PRI's and we get no more
artifacts. So, :
wanpipe1.conf:
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
wanpipe2.conf:
TE_CLOCK= MASTER
TE_REF_CLOCK= 1
zaptel.conf:
Hi Rob -
The problem is that when people dial into the Asterisk system using DID
numbers, it works the first time or 2, then I get busy signals.
A friend recommended I clear out the zapata and zaptel, start over, and
recreate my wanpipe stuff. He thought the problem was with the spans
I would like to convert a file to WAV49 for use with Asterisk using
the linux command line. Specifically I would like to upload sounds to
use for unavail.wav and busy.wav, but I'd like them to be compressed
so that space is not wasted.
I tried using SOX but havent found the correct command-line
Jan 4 17:16:54 asterisk kernel: asterisk[8455]: segfault at
0001 rip 007ac2d8 rsp 408ab0d8 error 6
What a disillusion, I hoped 1.4.0 would be much more stable than 1.2.4
that we used before.
1.2.4 lasted for 6 months before it crashed for the first time.
1.4.0 took 14
Hi,
I too have found this matching to be frustrating. I would like it to
behave as you describe.
Doug
--
Doug Meredith
506-854-7997 ext. 801
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Thursday, January 04, 2007
I completely missed the 47-47. That wasn't very smart.
With that T1, we don't have the full thing. We are sharing it with
someone else, and 4 of the channels are used elsewhere. It was setup
with em wink, so there isn't a d channel. I'm guessing I should have set
both to 44 and it probably would
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Andrewartha
Sent: 04 January 2007 05:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] caller id ring tones for Asterisk Phone
Jeronimo Romero
Richard,
We have underscores all over the place in our config files, including others in
queues.conf. I don't think that's the murder weapon.
I think, in general, queues are one of Asterisks biggest features, and also one
of it's shakiest. The reload, which is run from a script, caused a
I use a lot of included files in my dialplan and I see that #include
file.conf is now interpreted as an attempt to include a non existant
context.
--
One day at a time, one second if that's what it takes
___
--Bandwidth and Colocation provided by
Thanks Tzafrir,
I knew about the ex girlfriend logic - but does that allow to be
looked up from the callerid database instead? I'd like to have it only
be one goto for direct dial, and one for the main menu instead of
having to manually input all those numbers, and add new ones when
required.
I have an upcoming install which places the switch close to some employees
in a quiet work environment. Can anyone recommend a quiet 24 port POE
switch? The Linksys SRW224P behind me right now would be objectionable, I'm
sure.
How about the Linksys SRW208MP? I don't have one (yet), but I
Hello
I am currently having a problem, that threatens to drive me insane...
I cannot understand how Asterisk matches up a sip request with a peer.
Here is my example:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.59:5060;branch=z9hG4bK00d97b99;rport
From: 1088200336
Republic of South Africa
On 04/01/07, Olivier [EMAIL PROTECTED] wrote:
We are doing this here is RSA.
Please apologize but what does RSA stand for ?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
While I was trying to patch chan_sip.c to force a specific codec by
using a channel variable, I found out that this is already
implemented. It there even for asterisk 1.2
sip.conf:
allow=g729,gsm,ulaw
outbound call:
exten = _X.,1,Set(SIP_CODEC=ulaw)
exten = _X.,2,Dial(SIP/itsp/${EXTEN})
Hi,
I'm looking for opinions on the best value router to use for home offices.
It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the phones always have higher priority traffic
than the PCs. (and not rely on the phones to do the QoS because some
2007/1/4, Carlos Rojas [EMAIL PROTECTED]:
Hello
Do no forget the rtp ports 1 to 2
Regards
On 1/4/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote:
007/1/4, Bob Chiodini [EMAIL PROTECTED] :
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first
Hi Chris -
I would like to convert a file to WAV49 for use with Asterisk using
the linux command line. Specifically I would like to upload sounds to
use for unavail.wav and busy.wav, but I'd like them to be compressed
so that space is not wasted.
I tried using SOX but havent found the correct
I have considered opening a bug report on this, but wanted to get some
feedback and make sure I am not missing something in the way of a simple
work around. What is the scenario in which this impacts your
implementation?
Ours is the desire to use the same realtime SIP database for many
Steven wrote:
It sounds very promising, I will have to check it out.
Great. Let me know how it goes for you.
Does it edit conf. files? Or is there just one lead in
conference number that flows into your app.?
Flexibility is a key design goal. The most comon installation
would be a single
On Thu, Jan 04, 2007 at 11:55:44AM -0500, Noah Miller wrote:
Hi Thiru -
Hi i need to done some modify/changes in one of the asterisk c source
code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
I would not recommend trying to
On Thu, Jan 04, 2007 at 09:26:07AM -0800, Bruce Ferrell wrote:
I use a lot of included files in my dialplan and I see that #include
file.conf is now interpreted as an attempt to include a non existant
context.
Are you running Asterisk as non-root? If so: is the included
configuration file
Mike wrote:
Hi,
I'm looking for opinions on the best value router to use for home
offices. It should work for a scenario in which there are 3 computers
and 2 SIP phones, handling QoS so that the phones always have higher
priority traffic than the PCs. (and not rely on the phones to do the
Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with
kernel 2.6.8-12-amd64-k8
make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
All,
(Happy new year!)
How big can an IAX channel grow to in size? (Realistically)
Eg, if I have a 2Mb pipe between two A*k servers, can IAX grow to use
the whole 2Mb with no issues, or do I need to create separate IAX
channels (and if so, how do you do that in the config).
Cheers,
Adrian
OK.
Have you had an opportunity to obtain the userid and password for the DEC
Chair at the FDP website? We have changes to make there on the site, i.e. new
officers.
or
The passcode on the DEC telephone, so that we can retrieve voice messages?
SAL
On Thursday 04 January 2007 11:40, [EMAIL
Hi Adrian -
(Happy new year!)
How big can an IAX channel grow to in size? (Realistically)
Eg, if I have a 2Mb pipe between two A*k servers, can IAX grow to use
the whole 2Mb with no issues, or do I need to create separate IAX
channels (and if so, how do you do that in the config).
It will
On Thu, Jan 04, 2007 at 01:42:57PM -0500, Guillermo Salas M. wrote:
Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with
kernel 2.6.8-12-amd64-k8
make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
Hi Yusuf -
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco
So calls originate from the Asterisk side (registered users on SIP or just ZAP
phones), and they go
out H323 or SIP to Cisco, where they go out PRI.
I
Hi,
I've been trying to get asterisk to use an outbound sip proxy. Putting the
outboundproxyhost directive in the [general] section of sip.conf, but it
doesn't seem to work.
My expectation would be that by setting outboundproxy and outboundproxyport
in that location, then all dial commans (or at
Yes, I knew that but it's nice that you mention it. I want QoS specifically
to prevent large downloads/kids using BitTorrent in their bedrooms locally
from interfering with the calls.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent:
I just need to retrieve a value from a field in a postgres database, and
playback this value when someone dial a specific extension.
On 1/4/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
O.Kamal wrote:
I need to retrieve my asterisk to retrieve a values from postgresql, i
am looking for some
linkys is definitively one of the most noisy switch!
it must be placed far away from people :-)
Noah Miller wrote:
I have an upcoming install which places the switch close to some
employees
in a quiet work environment. Can anyone recommend a quiet 24 port POE
switch? The Linksys SRW224P
Noah Miller wrote:
I have an upcoming install which places the switch close to some
employees
in a quiet work environment. Can anyone recommend a quiet 24 port POE
switch? The Linksys SRW224P behind me right now would be
objectionable, I'm
sure.
How about the Linksys SRW208MP? I don't
For those following this discussion, below is my original message to
the list and a reply from Sangoma Tech Support.
-t-
On Jan 3, 2007, at 8:43 AM, Todd H wrote:
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my
Open-wrt on supported router and some custom scripting works very well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Thursday, January 04, 2007 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Having QoS on your router is valuable to prevent some large download
from buggering your calls though.
Isn't QoS only useful to prevent large uploads, as download rely on ISP
router prioritizing Voice over Data ?
Cheers
___
--Bandwidth and
Am Donnerstag, den 04.01.2007, 12:27 -0500 schrieb Matt Gibson:
Thanks Tzafrir,
I knew about the ex girlfriend logic - but does that allow to be
looked up from the callerid database instead? I'd like to have it only
be one goto for direct dial, and one for the main menu instead of
having to
Well Moises, if you do, please drop me a line and I will gladly test it.
I was mentioning digium because AFAIK, the guys at digium are in touch with
the programmers and contributors so I thought maybe they would have an
insight on whats going to happen with unicall on 1.4, I mean, somebody at
the
Hi All, as good?
But to tell the resolution of this problem:
I change board TE110P and is all good now,is functioning almost perfectly.
:)
When I occupy channel 17 of the board and the call is without audio is as
if it did not exist TX and RX.
I find that this is problem in the Siemens HiPath
Start by writing some agi in perl or php. Either of those should allow
you to access pg database servers. If the agi is too slow for your needs
move it over to compiled c.
O.Kamal wrote:
I just need to retrieve a value from a field in a postgres database,
and playback this value when someone
[SOLVED]
On Thu, 2007-01-04 at 21:33 +0200, Tzafrir Cohen wrote:
On Thu, Jan 04, 2007 at 01:42:57PM -0500, Guillermo Salas M. wrote:
Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with
kernel 2.6.8-12-amd64-k8
[..]
gcc 3.3 and gcc 3.4 (which are availble on Sarge)
1.4 has been released, and it's still crashing. I guess it hasn't been
resolved yet.
On 11/10/06, Mik Cheez [EMAIL PROTECTED] wrote:
Mani,
I've gotten the same result both dialing from a gtalk client to SIP, as
well as an SIP call to gtalk. You can run a jabber debug before the
call is
It used to be a problem to have very big iax2 trunks (e.g. 100 channels).
This should be resolved in asterisk 1.4, in older versions you can just
work around it by making several smaller trunks.
Zoa
Noah Miller wrote:
Hi Adrian -
(Happy new year!)
How big can an IAX channel grow to in
Question:
So for people using E1 with R2 or PRI as signaling, what are my
options in asterisk 1.4 and 1.2?
On 1/4/07, Anton Krall [EMAIL PROTECTED] wrote:
Well Moises, if you do, please drop me a line and I will gladly test it.
I was mentioning digium because AFAIK, the guys at digium are in
We are testing a Linksys SRW208P (lower power capacity than the SRW208MP).
It DOES have an internal fan as you (and many of your colleagues) will
notice the moment you apply power.
It is a surprisingly noisy little beast.
Ah, thanks for the tip, Drew. I was going to get one of these, but I
Hi
I am new on Asterisk.I want to route call to quintum Gateway.How can i
configure asterisk and quintum.
Plz help me.
Thx
_Reaz
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Hello Yusuf
yusuf wrote:
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco
So calls originate from the Asterisk side (registered users on SIP or
just ZAP phones), and they go out
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