On Tue, Jan 31, 2006 at 06:29:07PM -0700, Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, January 31, 2006 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Dinesh Nair wrote:
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not at least start to fall apart?
Hi Olle,
Nice to know that. In my case I'm simulating a prepaid call
from Asterisk to SER. On the Asterisk side, there are users
registered with of course different extensions. Asterisk uses SER as
the SIP trunk and SER will forward it to the PSTN gateway. Asterisk
registers to SER
On Wed, Feb 01, 2006 at 03:38:21PM +0800, Dinesh Nair wrote:
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not
On Wed, 2006-02-01 at 15:18 +0800, Dinesh Nair wrote:
On 01/31/06 15:37 trixter aka Bret McDanel said the following:
symantic differences but not a lot in terms of performance. Because the
systems are close enough its mapping stuff more than creating a virtual
machine.
the mapping stuff
On 02/01/06 15:54 Dustin Wildes said the following:
Why not bond multiple NICs together to do a load balance output? Would
provide redundancy as well.
the issue here would be the increased interrupts needed to handle the load,
not necessarily a bandwidth related issue. using device polling
I have tested an asterisk server with over 5000
concurrent calls.
The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1
gbps Ethernet connection on a cisco 3560 switch.
This works, but puts some serious stresses on the
system.
Why don't u considered using g.729 codec, this will at
least
Hi,
Has anybody come across a situation where they were unable to register with
Asterisk through a SIP stateless proxy server?
I'm getting an error:
403 Authentication user name does not match account name
As far as I can tell the requests reaching Asterisk with and without the proxy
are
Ronald Wiplinger wrote:
snip
601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf
621 and 626 are in Real-time sip_buddies
621 and 626 changes username back from name to number (name) in the
database, and never shows it in sip show peer
615 changed username Ronald office to
Even if you could, you wouldn't want to use just one system to handle
this call load. What happens when you lose a power supply or a hard
drive, or any other random failure?
I would think you would want a more robust design. While you can go the
signate way and use SGI hardware to increase your
I have [EMAIL PROTECTED] 2.1, running * 1.2.1. I am trying to put
information into the userfield with SetCDRUserField and AppendCDRUserField. However,
the field is never populated in the cdr Ive checked the csv
files and the MySQL asteriskcdrdb table. The field is defined in the
MySQL
Hi,
I'm new in this list and my experience on Asterisk quite limited, can
anyone help me. This line is shown in the asterisk console and I don't
now the meaning
-- parse_srv: SRV mapped to host proxysip.sip.somedomain.com, port 5060
Thanks folks
--
[EMAIL PROTECTED] is believed to have said:
While we are at the subject another couple of simple related question.
Are HFC-S cards active? I got one for a very low price, so that I
imagine it will be NOT the case...
No, these cards are passive. The protocol handling must be done by
the
Hello list,
as likely most of you, this morning I have compiled my fresh 1.2.4 with
H.323 support.
I have prepared a recipe of the compilation process so that maybe it can
be useful for future reference or for a quick start:
http://www.oinko.net/astrecipes/index.php?n=102
Any comment or
I guess I just assumed that that the connection to asterisk would have
to be IP since it is absolutely impossible to connect ~208 T1s directly
to a single asterisk server. You would have to use an external media
gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :)
Not
If you have an Asterisk and a SER, is there a reason for registering at all?
/Olle
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Using isdnlog command and watching results in /var/log/isdn.log I see
that my call to 338??? becomes a call to +39(my country
prefix)38???. Why does isdn miss a number (3)? Does someone think
that the problem is in?
Thanks, Francesco.
___
Aaron Clauson wrote:
Hi,
Has anybody come across a situation where they were unable to register with
Asterisk through a SIP stateless proxy server?
I'm getting an error:
403 Authentication user name does not match account name
Doesn't the error message explain the problem, really? The
On Wed, 1 Feb 2006, Aldo Bergamini wrote:
What cards do support operation of an ISDN phone set? (I imagine there
will be something similar to the FXS-FXO stuff of the analog world in
the ISDN land).
For that you need cards operated in NT-mode. The driver / protocol must
support this,
Chris A. Icide wrote:
Ronald Wiplinger wrote:
snip
601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf
621 and 626 are in Real-time sip_buddies
621 and 626 changes username back from name to number (name) in the
database, and never shows it in sip show peer
615 changed username
Still,
Everybody is using T1 / E1 interfaces in servers.
I would go for purpose build voice gateways.
Vegastream or cisco GW are able to handle multiple T1/E1 connections
easily. (make sure that in a cisco GW you get enough DSP capacity)
In this scenario the asterisk server is just used to make
Hello,
MP Can I couple this to the sound card in the Asterisk server and then have
MP it play into the MOH? If so how?
Yes, it's possible. I've tried it last week.
1. Add the following into musiconhold.conf:
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
Still,
Everybody is using T1 / E1 interfaces in servers.
I would go for purpose build voice gateways.
Vegastream or cisco GW are able to handle multiple T1/E1 connections
easily. (make sure that in a cisco GW you get enough DSP capacity)
In this scenario the asterisk server is just used to make
On Wed, February 1, 2006 7:51, Marnus van Niekerk said:
Hi,
I have a strange problem on some isdn modem channels. (* 1.0.9 /
chan_modem / 2xHFC-S cards).
Everything works fine but when the 2nd (and 3rd etc..) call comes in and
* answers and there is about a 1/2 second of sound from the
-20060201-105559-113878775
9.45) in new stack
-- Executing Playback(SIP/6900-ee4b, custom/None) in new stack
-- Executing NoOp(SIP/6900-ee4b, before queue|) in new stack
-- Executing Queue(SIP/6900-ee4b, 666|t|||0) in new stack
-- Started music on hold, class 'default', on channel 'SIP
Hi ,
I think i understand what you mean by your mail.I have done the same thing.
You must download following modules from and asterisk site
e.g.
www.digium.com
1.asterisk-1.0.3.tar
2.libpri-1.0.3.tar
3.zaptel-1.0.3.tar
Then there is a process youneed to follow which you will find in th read me
Try the following in zapata.conf
internationalprefix=900
nationalprefix=90
Which should do this for you unless your provider is not supplying the
correct indicator.
Regards,
Steve
On 1/31/06, Phil Blundell [EMAIL PROTECTED] wrote:
When a call arrives on our PRI from a UK domestic number, the
You must change in the indication.conf the country
[general]
country=it ; default location
Simone Cittadini wrote:
After reading a description of apparently the same problem by Juan J.
Sierralta more detailed than mine
tuuu tuuu instead of tuuu we've solved the problem changing
This looks like the solution.
I'll let you know how I get on.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
[EMAIL PROTECTED] wrote:
Hello,
MP Can I couple this to the sound card in the Asterisk server and then have
MP it play into the MOH? If so how?
Yes, it's possible. I've tried
How does the customer maintain the message if I have to capture it every
time he changes it?
This is not the solution.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Peter Fern wrote:
Using the classic MoH, use a custom moh player (see
Matteo Piazza ha scritto:
You must change in the indication.conf the country
[general]
country=it ; default location
After reading a description of apparently the same problem by Juan J.
Sierralta more detailed than mine
tuuu tuuu instead of tuuu we've solved the problem
Imran Ahmed wrote:
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know if IaxComm has inband dtmf mode available, if so enable
it and see if it works.
Someone can suggest
On Wed, February 1, 2006 12:07, Accursio Avona said:
Imran Ahmed wrote:
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know if IaxComm has inband dtmf mode available, if so enable
If I call a cellular phone while it's off, I can't hear the voice saying
called number is unreachable, but only if I'm passing trough a iax
channel.
SIP client --- Asterisk --- SIP gateway, works
SIP client --- Asterisk client --- Asterisk server --- SIP gateway,
doesn't work
(I can't put
Hello,
we use Handytone 386 adapter together with the Asterisk PBX.
Using normal analog phones together with the Handytone and Asterisk works fine.
We also can connect a standard fax machine to the Handytone ATA adapter.Send and
receive of faxes works fine.
When we connect a standard analog
Juan Carlos Castro y Castro wrote:
Ah -- for all intents and purposes, assume I can obtain the most kickass
PC
server hardware in the known Universe. So -- any real-life experiences
out
there?
We successfully ran four fully-loaded TDM2400Ps in a server here during
development
Hello everyone, this is my first post to the list, so hello again.
We're a small company in Romania and we're trying to set up a really small
version of call center. That is, we want to get a few land-lines from our
telco in different countys and bridge all calls to our HQ, in order to
make it
The way I understand things, there's no way for a analog line to reject a
call (give the caller an busy tone) if the line is not actually busy.
Would a digital EURO-ISDN line give me this ability?
Thanks,
Cosmin Prund
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As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?
Thanks,
Cosmin Prund
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Cosmin Prund wrote:
The way I understand things, there's no way for a analog line to reject a
call (give the caller an busy tone) if the line is not actually busy.
That is totally under the control of you in the dialplan. You send busy
or congestion or answer the calls as you see fit, either
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?
Using Asterisk is a good way. If you define a phone in sip.conf and turn
on qualify=, we will measure the latency for the network between the
phone and Asterisk.
/O
On Wednesday 01 February 2006 14:12, Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network
latency? If not, how can I measure latency?
Only if ICMP Echo has the same Class of Service (DSCP, 802.1p,
priority/class in routers/shapers etc.) as VoIP traffic across the
Hi Cosmin
You should be able to get about 12 simultaneous calls on a 128k line and
about 28 on a 256k line according to asteriskguru's bandwidth calculator
http://www.asteriskguru.com/tools/bandwidth_calculator.php.
Kind Regards
Garth
BitCo Data Communications
http://www.bitco.co.za
Cosmin
Dear Sir/Madma,
I need to create dial plan to access out side world from office. Our
office PBX system need to wait few seconds after pressing 9 before
enter phone number.
How should i prepare dial plan to add delay between first and rest of
digits.
Here is my idiotic try;
exten =
There is a good utility called iaxping to test IAX latency.
Kind Regards
Garth
BitCo Data Communications
http://www.bitco.co.za
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?
Thanks,
Cosmin Prund
Andy Kuo wrote:
I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax,
Sorry for bringing so old topic, but what should I put in
extensions.conf for receiving faxes? Is
exten = 1234567,1,rxfax(/home/steveu/testfax.tif)
enough or do I have to answer the call before or
I have a strange problem with echo.
My setup includes a Grandstream HT-488 which is both an FXO and a FXS.
I noticed last evening that if I called the FXS through my asterisk box
from my cell, the resulting connection was fine for me at the cell end,
but produced dramatic and
Hi,
I am 'playing' with asttapi which looks great on a first installation but I
must be missing something regarding the source code because I haven't been
able to work with it without problems.
If you have played with this, you already know that the code to talk to
Asterisk is placed in a file
I have a request from a customer to be able to switch lines using dtmf, for
example pressing ** to switch to the second line. So if a user is on the
phone, and they hear the call-waiting beep (which I am alos not sure if it
can be implemented on asterisk directly), they would then press ** to
That is correct, The SIP phones are all on our LAN. I changed the nat's to say no, but I still get the same problem. Another thing, when I call out to the pstn from our local sip phones. The same problem happens. The outid line rings, the person picks p but no sounds.Any suggestions
Mabe I didn't understand your answer, mabe I did not ask the question
properly, so I'll try an other aproach:
Let's say I've got a * and a client calls it's number. Note this number
would not be SIP, it would be the old kind of number. My only choice would
be to use ISDN or analog. Under certain
I'm pritty sure VoIP traffic would not be favored in any way along the route
from my first * to my second *, but ICMP might be boosted a bit by my ISP to
look better :-)
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dmitry
Thanks.
I was aware of the calculator but I thoght I'd ask here too, just in case
there's something else that needs to be taken into account.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Garth van Sittert
Sent: Wednesday, February
Thanks Facundo for instruction but it didn't work. there is nothing new in your suggestion compare to my conf files nevertheless I tried itbut it didn't work. I can make call from my sip phone but can't receive any phone call. I am sure some one had had the same problem and solved it. as always
are you sure your sip phone is registering ok?
2006/2/1, abc def [EMAIL PROTECTED]:
Thanks Facundo for instruction but it didn't work. there is nothing new in
your suggestion compare to my conf files nevertheless I tried it but it
didn't work. I can make call from my sip phone but can't
Based on our benchmarking, I am VERY
skeptical of this number. Im guessing that you dont really have
RTP streams going through the NIC.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Joash Herbrink
Sent: Wednesday, February 01, 2006
12:23 AM
To: Asterisk Users
Hy,
I would want to test the FXO and FXS channels of a TDM400p with the
asterisk. For example, when I want to Know if a sip peers is correctly
register, I use ChanisAvail, but if a module FXS hasn't a phone
connected, How could I Know it?.
Note, that I don't want to Know if the channel ZAP
Francesco Peeters (Asterisk) wrote:
On Wed, February 1, 2006 12:07, Accursio Avona said:
Imran Ahmed wrote:
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know
Thanks Jerry. What I don`t understand is what are the files greet.gsm and
temp.gsm, and why are they present in one mailbox and not the other?
And why, probably for the same reason x, is it that when I record my
unavailable message in my mailbox, and call back to try it, the default
asterisk
On Wednesday 01 February 2006 10:54, Michael Collins wrote:
I have [EMAIL PROTECTED] 2.1, running * 1.2.1. I am trying to put
information into the
userfield with SetCDRUserField and AppendCDRUserField. However, the field
is never populated in the cdr - I've checked the csv files and the
On Wed, February 1, 2006 15:04, Accursio Avona said:
Francesco Peeters (Asterisk) wrote:
SNIP
AFAIK there's no DTMF option in IAX2...
IAX always sends DTMF inline, eliminating the confusion often found with
SIP.
http://www.voip-info.org/wiki-IAX
If so, wy the IVR does not hear the dtmf
Why does my dial command exit non-zero when the calling party hangs up?
I am using a t1 with the following configuration:
/*from the zaptel.conf */
pan=1,1,0,esf,b8zs
em=1-24
=
/*from zapata.conf*/
[channels]
language=en
signalling=em_w
; change signalling to featd when
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] wrote:
I have a Polycom IP501 phone and have set it up to download the config from
an FTP server, it did this once and now is in an endless loop of trying to
contact the FTP server, failing, then rebooting.
When I watch the FTP
not sure but this is the output from the pbx:sip show registryHost Username Refresh State local_sip:5060 stargate3 105 Registered local_sip:5060 stargate2 105 Registered
local_sip:5060 stargate1 105 Registered from sip phone I can any other phone (cisco with sccp or iax protocol) but I
Hi all
I am a new user of asterisk and want to implement three way
calling but not getting any information about how to configure
asterisk conf file, Dial Plan etc..
whether a Digium card is essential or not?
Is Three way calling is same as MeetMe or separate feature in
asterisk?
Can You please
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Larsson
Sent: Wednesday, February 01, 2006 2:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps
-
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at
80%
utilization and not at least start to fall apart?
additionally, 5000 simultaneous SIP calls at 20ms intervals will send,
5,000
Leo Ann Boon wrote:
I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default
value is 00. I thought the value should be 010200. I know many
people have problems compiling chan_bluetooth because of this
inconsistency. Anyone has the last word on this?
That is probably a
Hi, We are a service provider using Asterisk for our softswitch. We offer SIP connections via IP phones as well as PRI and POTS replacements for our customers. However, i
am having problems with incoming calls from a Cisco IAD2431 and its dialing context. When a call comes from the PBX through the
Mark Johnson wrote:
Anyone have any idea what's causing this or how to debug it? I'm
pretty sure the root cause is with chan_sccp.so, but not sure how to
prove it.
I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from
12-17-2005. Now, once or twice a week, I get this on the
I guess I just assumed that that the connection to asterisk would
have
to be IP since it is absolutely impossible to connect ~208 T1s
directly
to a single asterisk server. You would have to use an external media
gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :)
Not
Francesco Peeters (Asterisk) wrote:
Are you sure it *is* sending DTMF in the first place? (Just trying to find
a logical place to start here...)
I do not use meetme, but when I use idefisk, my (*) server recognizes the
DTMF.
Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE
Thanks for your input, everyone, but I still think it is on Teliax's end...
I will present our collective thoughts to their tech.
Kevin,
I am using IAX. When I turn on IAX debug, I get:
--SNIP CLI OUTPUT--
-- Executing Dial(SIP/Brent_ring-bcf7, IAX2/teliax/18005558355) in
new
stack
I am using agentcallbacklogin for queues
I have a desire to modify the call behavior to the agents extension if
the call is from a queue (opposed to from a PRI or another extension).
Is there a channel variable that can be read that would indicate the
source channel waiting to be bridged is an
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, February 01, 2006 7:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] query about Three way calling
Hi all
I am a new user of
Juan Carlos Castro y Castro wrote:
Sweet! So -- what are the specs of that server you tested on? Specifically,
the power supply wattage? My intended use here is all FXS, so I suppose a
limit of 2 would be reasonable.
I believe it was an HP ML350, with only a single CPU and hard drive.
By
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Cosmin Prund
Sent: Wednesday, February 01, 2006 6:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] (newby) Is PING a good indicator of
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Wednesday, February 01, 2006 5:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] (newby) Is PING a good indicator
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its dialing context. When a call comes from the
PBX through
Olle,
Should I use defaultip and username when configuring gateways that do
not register to *? Could this be why my gateways are not sending calls
to the appropriate contexts?
Here is my sip.conf configuration for my gateway:
;---
; sip.conf
;---
Just figured I'd offer my experience with power restrictions and the
2400P...
When I purchased a Dell 2800 series tower I found that there were no
Molex power connectors in the box so I thought I was out of luck if I
wanted to use *any* FXS card (TDM400 or TDM2400) in this box. I spoke
with
Anyone? This has been killing me for days thankssdgesa gaeharth [EMAIL PROTECTED] wrote: That is correct, The SIP phones are all on our LAN. I changed the nat's to say no, but I still get the same problem. Another thing, when I call out to the pstn from our local sip phones. The
Hi,
Im wondering if anyone has experienced an issue with
the XLite softphone and asterisk accepting dtmf? I can listen to my
voicemail perfectly from my hardphone. However when I dial the voicemail number
from my XLite softphone and enter the password at the voicemail prompt, an
error
Mark Johnson ha scritto:
Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked:
Avoided deadlock for '0xbf002d10', 10 retries!
Yes, the chan_sccp could lock the asterisk channel.
To fix it I need a sccp debug 10 log of the call that is locking the channel
Sergio
DOH! Thank you for saying this, I forgot that I changed it to inband on some
FAQ I read, that is what actually broke it, changing it back to the RFC fixed
the problem. Thank you!
= = = Original message = = =
Are your DTMF settings in sip.conf correct? On my grandstream 486 ATA's I had
to
Thank you for the excellent suggestion, I'll give that a try. I can see now in
the manual that there are some recommendations for FTP servers, but I'm going
to first try the * server to see what happens.
= = = Original message = = =
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED]
Garth van Sittert wrote:
There is a good utility called iaxping to test IAX latency.
Kind Regards
Garth
Another one is check_asterisk.pl plugin from the Nagios monitoring system.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but
Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the
dial string, but still no luck. I hooked up and listened on the line when the
call went out, and never heard any DTMF's. I'm sure this must be something
simple, I just can't seem to figure out for the life of me
Hi -
That is correct, The SIP phones are all on our LAN. I changed the nat's to
say no, but I still get the same problem. Another thing, when I call out to
the pstn from our local sip phones. The same problem happens. The outid line
rings, the person picks p but no sounds.
Any
Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot in the
dark, a brainstorm on my part is all)
Here's what the logfile shows. Any ideas? And is
there a way to fix the deadlock without restarting Asterisk?
Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked:
Brent,I had this same problem with Teliax (atleast it sounds the same). I had wanted to use g729 over IAX, so I set that on the Teliax website, but it would not connect. After weeks of asking for resolution, I just gave up and used g726 which was working. Then, about a month later, I moved my *
Just figured I'd offer my experience with power restrictions and the
2400P...
When I purchased a Dell 2800 series tower I found that there were no
Molex power connectors in the box so I thought I was out of luck if I
wanted to use *any* FXS card (TDM400 or TDM2400) in this box. I spoke
same problem here, made a workaround with an agi
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its
hallo,
maybe somebody could help me, i try to bring my asterisk server to
native bridge two iax2 channels, on my old asterisk server (Asterisk
CVS-v1-0-03/23/05-10:07:13) it is working,
since i have installed the latest cvs code, asterisk stayes always in
the middle of my iax clients? i
Hi,
We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones.
Most things are running fine ;-)
But, when you are calling and you want to Transfer, you need
to press first on the 'more' button (4th), then you have the
label 'Trnsfr' to Transfer.
these are the lables on the softkeys when
Hi.
On Asterisk for Xlite extension U need to set dtmf=inband
execute: sip reload
and that should be working
On Wed, 2006-02-01 at 17:02, Aisling wrote:
Hi,
Iâm wondering if anyone has experienced an issue with the XLite
softphone and asterisk accepting dtmf? I can listen to my voicemail
You might try inband dtmf tones
Main Menu Advanced System Settings DTMF settings DTMF Force Send In Band: Yes
On 2/1/06, Aisling [EMAIL PROTECTED] wrote:
Hi,
I'm wondering if anyone has experienced an issue with the XLite softphone
and asterisk accepting dtmf? I can listen to my
Dear all,
Anyone has experience using group and group_count to limit
outgoing calls in AGI/CAGI?
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
EXEC Gotoif $[${GROUP_COUNT([EMAIL PROTECTED])}
1]?BLOCK
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
But it doesnt work as
Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Wednesday, February 01, 2006 5:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] (newby) Is PING a
Sum Ding Wong wrote:
Olle,
Should I use defaultip and username when configuring gateways that do
not register to *? Could this be why my gateways are not sending calls
to the appropriate contexts?
No, you either configure hostname and fromuser in a peer, and dial by
dial(SIP/peername)
or
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