Hi All;
Asterisk version is 1.8.4
The call is going to the queue but it is not routing to the agent which is
logged in.
I am afraid that I am missing a parameter to be set or enable or disable in the
queues.conf so it is causing not to route for the interface.
I am getting the following error
Hi All;
I succeeded now to configure callpickup so if the SIP user pressed *8 then it
will pickup the call within the group.
What is the possibility to have another code (for example *7 or any thing else)
to pickup the call from another callgroup, for example: if I pressed *8 then I
can
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes to
be working togethor as one entity so a distributing for the load and for the
tasks can be acheived.
I need such kind of protocols to be used in a large call center is where alot
of E1s and alot of agents.
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes to
be working togethor as one entity so a distributing for the load and for the
tasks can be acheived.
I need such kind of protocols to be used in a large call center is where alot
of E1s and alot of agents.
I need the agent password?
From the other side, is there a method to login as agent and to be assigned to
specific extension (other than the extension that I dialed the code to login)?
Regards
Bilal
-
El 18/07/11 18:03, bilal ghayyad escribi?:
Dears;
If I need
have 2 options, add an agent to the queue or add a
registered ip phone(
or pstn line) to the queue.
in first option, your operator must enter a password to
identify as agent.
but next option does not need password.
On Mon, Jul 11, 2011 at 3:06 AM, bilal ghayyad bilmar...@yahoo.com
wrote
Hi All;
How can I find a references customers that used Asterisk as IP Telephony or
Call Center or IVR? In which link they are mentioned?
Regards
Bilal
--
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I mean:
What are the customers (big customers I mean) that they installed Asterisk in
their company to be as a reference?
Example: Toyota, GM, Hilton, Shiraton hotel, ... etc
An example of such companies, whom?
Is there a link that mention them?
Regards
Bilal
---
What do you mean by
Hi All;
Why we use the agent password when we configure the agent in the agents.conf if
the agent login by dialing the number configured in the extensions.conf?
example: exten = 28, 1, AgentLogin(1001)
I know that agent username is used to assign the agent to the queue, but when
we use the
Hi All;
How agent logout if he logged in using AgentLogin?
Also, there is not ready and not ready status? Or only agent login and logout?
Appreciate ur kindly help.
Regards
Bilal
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Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the incoming calls of the agents?
Regards
Bilal
Hi All;
The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by
selecting the add-on). But really does not work in good performance, for
example: if a call came from gnugk to asterisk and the ooh323 handled it, the
performance is bad .. some calls are drop and if it is
Just to be sure that I am working in the right direction.
To do ACD, then I have to configure queue.conf and agent.conf?
One more question: if the agent needs to be in the NotReady state, then how
this can be acheived?
Regards
Bilal
-
FreeBPX calls them Ring Groups,
Hi All;
We know that agents can login and logout from the phone handset. But if we need
the login, logout, ready and not ready to be from an application and to be
integrated with the CRM, how to acheive this?
Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which
is
Hi All;
To be able to distribute the incoming calls on a group of extensions, is there
huntgroup in Asterisk? Or what I have to use?
I need first call to be send for extension 500 and second call to be send for
extension 501 and third call to be send for extension 502 and fourth call to be
Hi All;
I am running Asterisk version 1.8.4 and I need to know if I am going to use it
as a call center, and I have up to 6 E1s and about 150 Agents running
concurrently, did anyone test if Asterisk will crash or not? How much it might
be stable? And for how long (number of days or monthes) it
Thanks a lot. OK, from where you got these files? I am trying to know the
source so I can get from it any missing file that the phone is needed.
Regards
Bilal
-
bilal ghayyad wrote:
Dears;
The Cisco 7942 worked in SIP and did not work in
skinny firmware (in skinny
Dears;
The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it
register but no voice can be heared).
But now when we need to dial any number from the Cisco IP Phone 7942, it gives
busy (the phone send the call for the asterisk just by dialing the first digit).
So, do
, will also face the same problem that the phones
will restarted if I did reload?
Regards
Bilal
--- On Mon, 6/20/11, Dan Austin dan_aus...@phoenix.com wrote:
From: Dan Austin dan_aus...@phoenix.com
Subject: RE: Cisco IP Phones and Skinny in asterisk
To: bilal ghayyad bilmar...@yahoo.com
Date
Dear Warren;
It look like u have a good experience in 791x series and in selecting SIP
formware, so I am sure you might be able to help in the following:
As u know, there are SIP firmware for Cisco phones to be used with Call Manager
and other firmware to be used with Generic SIP Server (other
Dears;
OK, I have two things now:
1) When I do reload from the asterisk CLI, then all the skinny phones are
reset. This is very bad thing, how to avoid this from happening in each reload?
Even if the reload will be done to take sip configuration !!
2) The line tone that is heared (the
the skinny.conf file for the configuration? About
the firmware on the Phone, it will stay the same?
I appreciate the kindly help please.
Regards
Bilal
---
Hi,
On 06/13/2011 01:04 PM, bilal ghayyad wrote:
Can anyone advise if using Cisco IP Phones in skinny
protocol
I'm
not receiving
all the features I should.. But MWI works and multiple call
appearance..
On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Dears;
snip
Have you thought about perhaps just flashing the phones to
use the SIP
firmware?
--
Thanks
Dears;
Actually, the needed file name to be ooh323.conf and not chan_ooh323.conf, so I
copied the file from chan_ooh323.conf to a new file name ooh323.conf and it is
working fine.
From the other side:
I discovered that when I need to do compilation, and when running make
menuselect then no
Hi;
Those are not needed for ooh323 .
In fact, chan_h323 won't build with them ATM. There's a
patch on the
review-board to make it build with it, but noone seems to
care about it.
* What do u mean by ATM? How to see this review-board that is related to this
issue?
What is the
Hi All;
Please I need a help in the ooh323.
First of all, the only way to have h323 working in asterisk 1.8.3 or 1.8.4 is
to use ooh323? There is no way to get the normal h323 channel that come with
asterisk to work fine !!
Now, let us see the ooh323 problem that I am facing:
Already I
Dears;
I am sure that you have experience with Cisco IP Phones. I need to be sure if
someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was
(if fine or it has a problem).
Are the below the only 3 needed files to be placed in the tftpboot directory:
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it UP, I have to configure
the two ports and both of those two ports to take the timing from span 1.
Why this, I do not know ! Although I am using only one E1 connected to span 1,
so why I
from my iPhone
On Jun 15, 2011, at 6:53 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it
UP, I have to
configure the two ports and both of those two ports to
take the
timing
:02 PM, bilal ghayyad wrote:
Dears;
The Asterisk version is 1.8.3.2
The Cisco IP Phone is 7942G and it is running now
skinny.
The used TFTP is tftp-server which is installed in
fedora.
I placed the following two files (but look like it was
not taken from the TFTP, as nothing
Hi All;
My ISDN was working fine, and suddenly I start getting the below:
sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!
There is a Yellow Alarm, so what it could be the problem?
My configuration as following:
system.conf
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
15 16:14:00] WARNING[2773]: sig_pri.c:985 pri_find_dchan: Span 1: No
D-channels available! Using Primary channel as D-channel anyway!
Regards
Bilal
--
bilal ghayyad wrote:
There is a Yellow Alarm, so what it could be the
problem?
Experience says you need to call your
BE?
Regards
Bilal
--
bilal ghayyad wrote:
There is a Yellow Alarm, so what it could be the
problem?
Experience says you need to call your provider.
Doug
--
_
-- Bandwidth and Colocation Provided
?
Regards
Bilal
---
bilal ghayyad wrote:
But I am afraid it is a bug because I read something
this in the below
This bug is referring to Zaptel, not dahdi.
If things were working fine, and you haven't made any
recent changes, in
my experience it's always been provider (99
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of
bilal ghayyad
Sent: Tuesday, June 14, 2011 2:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] sig_pri.c:985
pri_find_dchan:
Span 1: D-chanannel anyway!
Dears;
To patch libpri: I just place the patch file
Hi All;
Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not?
Or it is better to use it in SIP protocol?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi All;
Can anyone advise if using Cisco IP Phones
Which model(s) are you planning to use ?
in skinny protocol is fine or not? Or it is better to
use it in SIP
protocol?
--
Hi,
On 06/13/2011 01:04 PM, bilal ghayyad wrote:
Can anyone advise if using Cisco
the res_phoneprov_and_TFTP, actually I do not see
it that it is used to download the firmware and configuration files, but I see
it is used for provisioning, correct?
Regards
Bilal
-
bilal ghayyad wrote:
Any one can suggest a TFTP server to be installed in
Fedora
The one
Hi All;
I need to create the needed files for the Cisco Phones to be placed in the TFTP
server to be able to register on Asterisk.
I need a help in the following please:
1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP address
of Asterisk?
2) Regarding to the file:
Hi All;
Any one can suggest a TFTP server to be installed in Fedora (same machine that
Asterisk is installed) to be used for Cisco IP Phones to download the required
firmware and configuration files.
Thanks for the help in advance.
Regards
Bilal
--
Hi All;
This is the output of the pri show status, so I appreciate if to know if that
means the E1 is UP? What does it means that the status us (Status: In Alarm,
Down, Active)? What in the below result give an indication that it is UP?
CC*CLI pri show span 1
Primary D-channel: 16
Status: In
Hi All;
From the CLI, if I typed pri then I can find the command and the relative
commands for it .. does this mean that the libpri is installed well? How can I
be sure that Asterisk took the libpri and it is functioning?
Now, regarding to the PRI configurations:
The provider is using: ISDN
@lists.digium.com
Subject: Re: [asterisk-users] chan_zap
On Mon, May 23, 2011 at 10:36:09AM -0700, bilal
ghayyad wrote:
Hi All;
Suddenly the zaptel channel look like stop
working and it is giving me
this error when I do zap restart:
[May 24 19:30:21] ERROR[2772
Hi All;
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
Regards
Bilal
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hello All;
If I need the Asterisk to do automatic dialing for a list of numbers and when
the destination answer, then to play the proper sound message, is it possible?
How?
About sending SMS, can asterisk do this?
Regards
Bilal
--
Thanks Alex for ur help and advise.
In case we decided to do a script for this reporting, we will depend on the
logs or we need to use the AGI?
In case we will do a dashboard to display how many agents are login and how
many calls in the queue and how many calls in specific skill group?
Hi All;
It look like there are some free (open source) tools that are used for Asterisk
reporting special for call center (to see number of agents logged in, number of
calls now, .. etc), and to be used as dashboard.
Can someone direct me for something really is suitable and stable?
Regards
for the
correct usage. Also read ALL the
UPGRADE*.txt files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of
bilal ghayyad
Sent: Friday, May 06, 2011 12:49 PM
To: asterisk-users@lists.digium.com
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of
bilal ghayyad
Sent: Friday, May 06, 2011 12:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Configuring
Voicemail
Hi All;
Already in the voicemail.conf file, I added the extension 500 and kindly find
below my voicemail configuration:
[Internal]
0 = 1234,Gama Operator,opera...@gama.com
500 = 1234,Operator,opera...@gama.com
501 = 1234,Employer Name,employer_em...@gama.com
502 = 1234,Employer
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange the
sip username and password. What is the possibility that this will be capture by
the hacker and how to avoid this problem?
Regards
Bilal
--
Hi All;
I need to configure the SIP account so if first IP address failed then to send
for the second IP address. How to do this?
While configuring the sip account, at the host parameter, can I give two IP
addresses separated by comma? Or what should I do to have such redundancy?
Regards
Hi All;
I am using Asterisk 1.8, how I can protect my self from hackers in case they
was able to see my sip.conf file? I need the password to be encrypted, how?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
Hi All;
I am using Asterisk for Call Center (so agents login, logout, ready, not ready,
... etc). To be able to have a good call center reporting, on what I have to
depend? On the CDR of Asterisk or there is another way?
Is there a good open source tool to be used for Asterisk call center
Hi All;
I installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet
ports. I gave IP address 192.168.0.3 for one Ethernet port.
I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there
(in the command line) I can type a commands.
I have an Polycom IP
OK Dears;
Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the
equivalent?
I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if
someone can advise me:
Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, )
in new stack
[Apr
Kindly find below my notes preceded by ( * ).
Good morning,
from the last question i assume you're looking for a
SIP-based
configureation.
On 03-30-2011 00:16, bilal ghayyad wrote:
1) How I can assign for each button an extension?
you can configure them as lines (at least in my
Hi All;
I have an E1 card with two ports for ISDN PRI.
Do I need to install DAHDI in addition to LIBPRI?
For placing outside calls (outgoing) via the PRI, then in the extension.exe
file, I will use the Dial function? But how can I determine that I need to use
the PRI channels and not the
Hello;
I need to use Cisco IP Phones with Asterisk and I have some questions to know
how to use it if someone can advise:
1) How I can assign for each button an extension?
2) How I can assign for specific button a feature to be used (like call forward
or call pickup .. etc)?
3) As you know
Hi All;
Any one advise for open source prepaid billing other than A2Billing that can
work with Asterisk and it is rich by features (for large business)?
Regards
Bilal
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Hi All;
I would like to have two Asterisk machines to have redundancy between them, so
if first machine failed then we can depend on the second machine.
Because of this, I would like to know (if someone can advise me):
1) If I did modification on the configuration, how this will be applied to
ghayyad bilmar...@yahoo.com
Date: Monday, February 28, 2011, 5:35 AM
hi
using database as realtime functions solves your first problem,
for second try by using dns
best
On Mon, Feb 28, 2011 at 1:54 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I would like to have two Asterisk machines
Hi All;
My server and its slots written in it the following so I need to know which
card to order it (I need a card supporting 2 E1s):
PCIE_G2_X4
PCIE_G2_X8
Actually I do not know what is meaning by G2.
OK I tried to buy directly from the below link but I found it is mentioned that
it is x1
have asked a nearly unanswerable question.
Some prefer
one, some prefer the other. Both cards are quality items. I
can say that
I only have experience with Sangoma T1/E1 cards, but our
Digium FXO/FXS
card works well, too.
On 01/14/2011 12:42 PM, bilal ghayyad wrote:
Hi All;
We would
Hi All;
We would like to build a call center having 2 E1, but we would like to know
which card to select:
Sangoma or Digium?
And card type to be PCI express or PCI 5.0V or PCI 3.3V ?
Any advise or special recommendations for the call center?
Regards
Bilal
--
Hi All;
How to configure the buttons in the Cisco IP Phones to be used for different
functionalities like Call Forward, Call Pickup, ... etc?
For example, if I need to assign one of the buttons existed at Cisco IP Phone
to be used for CallFrw, how to do this in Asterisk?
Regards
Bilal
Dear List;
For each call (in specific case), I need to do a record and save in a spearated
file, so I am thinking the best thing is to save based on the time.
Monitor(wav,Record1,m)
So, how can I make the file name to be based on the current time (which is
changed always, or based on the some
Hi All;
A2Billing is working fine for Asterisk, but in case I need to use Asterisk and
Gnugk and I need to manage the accounts and the billing from one Database and
one billing system, so I need a prepaid billing that can work with both.
Which prepaid billing (open source) can be used to work
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
Regards
Bilal
--
_
-- Bandwidth and Colocation
?
Regards
Bilal
--- On Mon, 12/6/10, Steve Totaro stot...@asteriskhelpdesk.com wrote:
From: Steve Totaro stot...@asteriskhelpdesk.com
Subject: Re: TCP port, VPN and resolving the cutting voice problem
To: bilal ghayyad bilmar...@yahoo.com
Cc: asterisk-users@lists.digium.com, eng_mohd_ta...@hotmail.com
Webmin and NTOP.
Just be aware that as soon as you activate the firewall,
everything is
blocked, so if you are going to use it as a firewall, get
as many rules in
place as you can think of.
Thanks,
Steve T
On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Dear
Thanks all for ur participation and kindly advise.
As I noticed that jitterbuffer could help if the ping does not have request
time out but the voice is also cutting .. but in that case, I have to set the
jitterbuffer at the IP Phones and Asterisk boxes.
I have a polycom phone for example, and
Dear;
I understood that Vyatta is the solution for the QoS, but I am not able to know
if I can use a Vyatta hardware router to be DSL router and I set my QoS in it
to resolve the voice problem. Is it possible?
Thanks for the help.
Regards
Bilal
Thanks all for ur participation
Hi All;
Can I run the IAX on TCP port instead of UDP port?
If I ran IAX in TCP port, and in case my network was having a lot of users
doing browse on the internet and downloading, so in that case and if the IAX
used TCP port, so the voice will be better than using UDP (because in TCP the
lost
delay?
But again, what about running IAX in TCP port, this is possible?
Any other solution to resolve the cutting in the voice while others doing
download and browsing?
Regards
Bilal
On 30 Nov 2010, at 09:28, bilal ghayyad wrote:
If I ran IAX in TCP port, and in case my network was
having
Hi All;
Did anyone try to implement (installation and configuration and running) for
more than one asterisk instance (two or three instances), where each asterisk
instance to work on a difference IP than the other where the server already has
more than one IP address.
We need to implement
confirmation regarding this .. no one give any details in this.
Regards
Bilal
-
On 10-09-26 01:00 PM, bilal ghayyad wrote:
First of all, I am looking to have the H323 Gatekeeper
service available at Asterisk, and really I do not know if
1.4 or 1.6 or 1.8 started implementing H323
Hi All;
Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is
working fine and it is same as 1.4?
Appreciate ur kindly help.
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
Hi All;
First of all, I am looking to have the H323 Gatekeeper service available at
Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing
H323 gatekeeper functionality or not?
Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any
implementation for
Hi All;
We were using a link before to be able to browse the different asterisk
versions and download the needed one as tar.gz file, but really I am not able
to find this link again.
Anyone can advise me for that link where I can browse the different deliveries
(1.2, 1.4, 1.6, 1.8 versions)
Hi All;
I have my friend that use his mobile (Nimbuz) to connect for the Asterisk and
his account was working fine. Suddenly it stop working (not able to register).
From my mobile (Nokia) I was able to register using my username and password,
so I tried to register using his (my friend)
--
www.ilovetovoip.com
On 2010-09-23 7:22 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
I have my friend that use his mobile (Nimbuz) to connect
for the Asterisk
and his account was working fine. Suddenly it stop working
(not able to
register).
From my mobile (Nokia) I was able
Hi All;
I would like to use Asterisk for a call center, but really does not know if
Asterisk support the following in a good way:
1) Ability to do an inteligent routing, so to route the call to the proper
skill group based on the caller information?
2) If I can create skill groups and then
Hi All;
To resist the problems that occure because of the poor internet bandwidth, I
got to know that cRTP and VAD are helping to use less bandwidth.
From where I can determine in Asterisk to use cRTP and to use VAD?
What other suggested settings (as cRTP and VAD) that can help in this
Hi All;
I am configuring IAX endpoint, I just need to understand why I have to set
requirecalltoken = no to be able to register because the following message is
displayed:
[Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call
rejected, CallToken Support required. If
Hi All;
My Asterisk version is: 1.4.19.1
I am not able to pickup a call ringing at SIP Phone (exten 802), the call was
transferred from Zap FXO channel and I am trying to pickup it from SIP Phone
(exten 800), but it fails with the below error:
[Feb 2 21:14:25] NOTICE[2703]:
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of bilal ghayyad
Sent: 2010 m. vasario 7 d. 01:20
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] A2Billing and other prepaid
Billing like ASTCC,
who is better?
Hi All;
I used A2Billing, basically it is nice and fine
Hi All;
I used A2Billing, basically it is nice and fine, but management possibilities
is not that rich, so a lot of staff are need to be repeated that let the admin
facing a problem of the needed time to do the task.
Anyone advise for another open source prepaid billing that is rich by the
Hi All;
My Asterisk version is 1.4.19.1 and I am using the Pickup application, it works
when I try to pickup the call that was originated from extension (for example,
when 801 call 802, then the phone of extension 800 can pickup the call at 802).
But it does not work when someone call from
Hi All;
I have a Plocyom 320 model, it supports 2 extensions (line 1 and line 2), when
configuring line 1, then I have to determine the username and password and IP
address of the server to register on it. And same thing when configuring the
line 2.
How can I receive (and make call) using the
between
callers by using the
up/down keys.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of bilal ghayyad
Sent: Wednesday, January 20, 2010 11:52 AM
To: asterisk-users@lists.digium.com
Subject
Hi All;
When using the digest authentication method, so I have to create the realm
domain with its username and passwords to be used for SIP digest
authentication, correct?
Now, how to create this domain? Should be reachable (can be ping) from a remote
device?
In other words, to create this
, don't enable
compression and security
You want something different than sip.conf?
AFAIK it's all in there. And what you name realm should
possibly be a
context in asterisk language.
Or did I get you wrong?
Eckhard
bilal ghayyad wrote:
Hello List;
Anyone can advise how realm
Hello List;
Anyone can advise how realm authentication method is working? I mean, where to
create the SIP username and password and where to create the realm that will be
used for the authentication method for registration?
Any help?
Regards
Bilal
Dear Xavier;
Actually I beleive you put me in the right channel, but for me realm is
something new to be used. I did not try it at all before. I read some about it,
but still I am not familiar with it
If you can help me in the realm, I will appreciate this:
1) What is the relation between the
Hi All;
I am wondering of this H323 channel in asterisk, whatever I ask, I do not get
help :) - So, how to get help, I do not know.
To be able run Asterisk and gnugk on the same machine at same IP address, I
need to know how to configure the port ranges of the (Q931, H245, T120, RTP)
for the
Hello All;
Anyone can advise for the good phone (Polycom, Linksys, ... etc) that is a
stable and support the codecs: g723, g729, and speex?
Actually I would like to have the speex codec because it have the ability to
compress to very high compression so we can work with the low bandwidth (for
Hi All;
I am trying to have the possibility to pass traffic from SIP to H323, and I am
using the asterisk version 1.4.26.2 with h323 (so I have h323.so channel), the
h323 listens at port 1722 TCP and on the same machine I have gnugk running and
listens on 1721 TCP.
When placing a call from
Hi All;
I just need to know the openion about Grandstream phone, actually I tried Budge
Tone 201 and I chocked that there is a noise in the handset
(zzz) always, but in the speaker the sound is good and
no noise.
Anyone has idea about Grandstream, and if they have a
Hi All;
We are doing a configuration to link with another Simens switch via E1, they
gave us these paramters to be setted, but we are facing an error at the trunks
related to Asyn problem, look like something related to synchorinzation.
The simens paramters are:
Line Coding: HDB3
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