I have a new customer that wants to upgrade their Asterisk installation
from 1.2.27 to 1.4.22. They use FreePBX for administration. Since
there are many syntax and command changes from those versions of
Asterisk, is there an easy way to convert the FreePBX configuration so
it will work
FOP is not compatible with Asterisk 1.6, you should look into the FOP
list as the author is looking for people to try a new version to make it
work.
On Tue, 2008-11-18 at 21:14 +1000, David Klaverstyn wrote:
Hi All,
For some reason the Asterisk Flash Operator Panel is not working
Simply put the shared mailbox on all the phones definition like:
mailbox=100,101
That way the phone will flash if there is mail on any of the boxes.
On Tue, 2008-11-04 at 13:31 -0800, Kelvin Chan wrote:
Hi list,
I'm wondering if there's a way for multiple users to share the
I have a weird problem with a client. I recently upgraded to Asterisk
1.4.22 and Zaptel 1.4.12.1 on their server and now there is a problem
when a fax call is received.
Basically when faxdetect=incoming is set in zapata.conf the call comes
in and the fax extension dials a Linksys
I have a customer that wants to use meetme but they want to have the users
record their name so it is played to the other people on the conference. Is
there an easy way to do this?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161
I am having a big problem with DTMF. I have a customer using an
Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem
is that when they dial into a conference bridge or IVR where they have
to enter a code they always get an error. Either some numbers are
duplicated or
I have a customer that is reporting that sometimes when they dial an
outside line they can hear other conversations. At this moment I am
assuming it only happens when they dial an outside number and not
between extensions.
They are using Asterisk 1.4.11, Zaptel 1.2.12.1 (just
The Linksys SPA-8000 is an 8 port FXS unit that works very well.
For that volume you should also consider either a Channelbank or maybe
a Xorcom Astribank. You can get those in 24 or 32 port versions.
On Mon, 2008-09-29 at 02:00 -0700, Vieri wrote:
--- On Mon, 9/29/08, Sam Tam
This has now happened to me on two different machines with different
hardware. Suddenly a user dials and the last port on the card will have
a loud noise and the call cannot complete.
The first machine where this happened had a TDM400P card with 4 FXO
ports and the second machine
On Wed, 2008-08-20 at 03:11 -0600, Joseph wrote:
On 08/20/08 10:09, Steve Repo wrote:
On Wed, Aug 20, 2008 at 7:44 AM, Joseph [EMAIL PROTECTED] wrote:
Does anybody know if the process of upgrading firmware on Linksys
SPA3102-NA in Linux is the same as on Sipura 3K as described on
. If the phone makes and receives a call I can see the button
reacting to that call.
Name/username HostDyn Nat ACL Port Status
Realtime
108/108192.168.1.90 D 5061 UNKNOWN
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
You need a door phone or a switch to control the buzzer. I have used
the 2N Entrycom and Helios line and they work very well. If you have an
Astribank (Xorcom) you can use the output ports as switches. There are
many brands of door phones you can choose from. You can connect them as
I have a new setup that uses a 2N Entrycom door phone that has a switch
to open an electric lock. The way this works is that when you are
speaking with someone at the door you dial a code and it releases the
lock on the door. This part works great.
My customer wants to be able
My office Asterisk box has a TDM04B card for three land lines and a GSM
gateway. I have noticed that the Zap channels get stuck a couple times
a week and I have to restart Asterisk to clear them. Here is what I see
in the console:
Connected to Asterisk 1.4.21.2 currently running on
On Fri, 2008-08-08 at 23:00 +0300, Tzafrir Cohen wrote:
On Fri, Aug 08, 2008 at 10:16:31AM -0500, Carlos Chavez wrote:
My office Asterisk box has a TDM04B card for three land lines and a GSM
gateway. I have noticed that the Zap channels get stuck a couple times
a week and I have
I installed a new machine with CentOS 5.2, Zaptel 1.4.11 and Asterisk
1.4.21.2 and an OpenVox A1200P card. This card has its own driver and
Zaptel has been patched to use it. The problem is that from the moment
I load Zaptel I get this messages on the console:
buffer re-sync occur from
I am having a problem with and Asterisk installation where two ports
connected to a TDM400 card will have a very loud noise when you try to
dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank
8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18.
The problem
Carlos Chavez wrote:
I have a customer with a small outgoing call center. Usually only
3 to
5 agents online. We are still using Agent/XXX channels in this
application on Asterisk 1.4.18. I have an autodialer that is
making the
outgoing calls and then dropping them into a Queue
The main DSL provider in Mexico is no blocking access to port 25 so the
email notification for voicemail is stuck in the server.
I suppose that I have to change the sendmail configuration so it can
send email to an alternative port but I wanted to ckeck first if there
is an option
I have a customer with a small outgoing call center. Usually only 3 to
5 agents online. We are still using Agent/XXX channels in this
application on Asterisk 1.4.18. I have an autodialer that is making the
outgoing calls and then dropping them into a Queue where all the agents
are
are on a case by case basis with ANI II codes 61-63
http://www.nanpa.com/number_resource_info/ani_ii_assignments.html
Thanks,
Steve Totaro
On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez
[EMAIL PROTECTED] wrote:
My customer has a 10mpbs fiber connection to the
Internet so
Today I had a problem where the internet connection is unstable so
calls are getting dropped all over the place. The one thing I do not
understand is that at least 30 phones on the internal network went to
No Service. Since they are on the same network segment and on the
same subnet I do
I have a system that is driving me nuts. My customer is running
Asterisk 1.4.20.1 on a CentOS 5.2 server. It is a purely SIP and IAX2
service with no cards installed and it uses ztdummy from Zaptel 1.4.11.
They use Teliax for calls to the USA and Protel for calls in Mexico.
The
My customer has a 10mpbs fiber connection to the Internet so we have
always assumed that the connection is not really a problem. We will
look into it. Thank you.
On Thu, 2008-07-10 at 17:49 -0500, John Faubion wrote:
-Original Message-
Subject: [asterisk-users] Diagnosing
Since Zaptel 1.4.11 has been released, why is the link on the Asterisk
website pointing to 1.4.10.1? Is there a problem with the newest
version or just someone forgot to update the link?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
As long as each tenant has its own context you can use the same
numbering plan. The only thing you need to keep unique are the names
for the SIP devices. If you want your tenants to be able to call each
other then you would need to set up a special prefix for each tenant.
On Thu,
Don't know about Debian but in Fedora or CentOS you need to install
mysql-devel to compile Mysql support in Asterisk-Addons
On Wed, 2008-05-21 at 14:31 -0500, JR Richardson wrote:
Hi All,
I'm poking around with 1.6, tried to compile the addon package, but it
doesn't see mysql_config
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED] escreveu:
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would
like
to busy out that line. Since
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access
I am a bit desperate trying to solve this problem. Sorry if I am
abusing the list a bit with the same king of question.
The problem I am having is very specific which is why it is very
difficult to diagnose and fix. Basically an Asterisk server is
connected via E1 PRI to an
I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall.
Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look
as if they are on the same local network because the Fortinet rewrites
the incoming IP as its own address.
The problem I have is that when
I am still having a very frustrating problem win an Avaya-Asterisk
system. I have written about this before but I am expanding the
description of the problem just in case someone can give me some
insight.
This installation is an Asterisk 1.4.19.1 server connected to an Avaya
PBX
parameters correctly.
2) Also in sip.conf, try the following on the PAP2's sections:
disallow=all
allow=alaw:10
In case that fails, try also
disallow=all
allow=alaw:20
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED
Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED] escreveu:
I am still having a very frustrating problem win an Avaya-Asterisk
system. I have written about this before but I am expanding the
description of the problem just in case someone
I have a big headache. I have an Asterisk server connected to an Avaya
PBX. Everything is working between those two. The problem is that I
have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the
Internet to the Asterisk server through a Fortinet firewall. When
calling from a
I need a refresher course on how many licenses I need to buy. I have
an Asterisk server that receives calls by SIP (G729) and then sends them
to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if
the license is per channel or per call so I do not know if I need 32 or
64
I have a customer with a Fortinet Firewall that is having stability
issues with Asterisk and SIP endpoints (PAP2T) outside his network.
The first issue I see is that Asterisk sees all phones as the IP
address of the Fortinet. Since the parameter localnet defines the
local
I have a new installation where an Asterisk server is connected to an
Avaya PBX via a PRI E1. We are having a problem that I attribute to
their firewall but I just want to make sure.
When we make a call from the Avaya to a SIP extension there is only
sound on the receiving end.
I just connected an Asterisk server to an Avaya Pbx using the
instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk
+Avaya
Everything seems to be working as I can send and receive calls. The
only detail I am having a problem with is that when an extension on the
I do not know if this will make a difference but the protocol-variant
for Mexico should be:
protocol-variant mx,10,4
You only get 10 digits from the phone company.
On Wed, 2008-02-27 at 18:03 -0800, Andres Tello Abrego wrote:
protocol-variant mx,20,4
--
Telecomunicaciones
You do not have to do anything else. When Asterisk detects a fax tone
it will disable echo cancellation on those channels so the fax can go
through. Just make sure that the Astribank is the sync source for
timing and you should be able to send and receive faxes.
In your dialplan
We have an Asterisk server with a small outgoing call center. We use
AMD and it usually works very well on Zap channels (E1 PRI). We added a
couple of SIP trunks to reduce long distance costs but now AMD gets
stuck when the call goes out through the SIP channels. Here is an
example call
I would recommend you use Iaxmodem / Hylafax / Avantfax for your needs.
We use this with several customers and it works very well. This way you
do not have to patch Asterisk with spanDSP. You can set up as many
virtual fax machines as your machine will handle.
On Wed, 2008-02-13 at
On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote:
Carlos, I have some spare time today in case you want me to check it.
Is this your first time with Alestra?
Thank you for the offer.
Yes this is the first time I use Alestra for R2. I have another
customer that uses
I am trying to set up Astunicall 1.4.16 with a link from Alestra in
Mexico City. I have done everything I usually do for other links in
Mexico but this one simply will not send or receive calls. I just get
Protocol error.
Anyone has any experience with R2 and Alestra?
--
I am having a problem with DTMF when sending calls through Teliax
(SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working. The problem always happens when a user is
trying to call a conference system. They simply cannot get into the
conference
On Mon, 2008-01-28 at 17:03 -0700, James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Followed the instructions at
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
I dead end at patching the channels Makefile. There have been some
changes since these instructions were
On Thu, 2008-01-03 at 18:00 +, Russell Brown wrote:
Quoth Phil Knighton [EMAIL PROTECTED]
I've incorporated the kind responses from other list members, such as
setting call limits but to no avail! I've checked the function key
settings on the Snom, and adjusted it to match the
=asreceived
signalling=pri_cpe
pridialplan=unknown
faxdetect=both
channel=1-15,17-31
Any ideas on how to solve this problem?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
signature.asc
Description
On Mon, 2007-12-17 at 11:45 -0700, Robert Norton - SophMedia LLC wrote:
Are the agents “ignoring” the calls while their ringing?
--
Robert Norton
SophMedia LLC Operations Manager
Cell: 480-234-4312 Office: 480-626-5449 (x300)
P.O. Box 7755 Tempe, AZ 85281
http://www.XStreamHost.com
callerid=asreceived
signalling=pri_cpe
pridialplan=unknown
faxdetect=both
channel=1-15,17-31
Any ideas on how to solve this problem?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Tue, 2007-12-11 at 00:14 -0800, bilal ghayyad wrote:
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is
Is there an easy way to limit the number of participants on a Meetme
room? Lets say we only want 10 people to be able to enter a particular
meetme conference, how can I prevent number 11 from entering this
conference? We will not have a pin to enter.
--
Telecomunicaciones Abiertas de
On Thu, 2007-12-06 at 10:37 -0500, Mike wrote:
Hi,
The Asterisk Wiki (page: http://www.voip-info.org/wiki/view/Asterisk
+func+cdr) mentions I can set any custom CDR field I want. Here is
the example it gives:
; Update our accountcode field and then save some random music facts too
Your problem seems to be that the card is in T1 mode and you need it to
be in E1 mode. Check the jumpers on the card and change them to the E1
position. Or you can send the module a parameter to put the card in E1
mode.
On Mon, 2007-12-03 at 13:14 -0200, Roger C. Beraldi Martins wrote:
I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on
a CentOS 5 server. The server has a single TE110 card connected to a
provider called Alestra in Monterrey, Mexico. Since we installed
everything we have been having problems dialing certain numbers, those
numbers
On Tue, 2007-11-13 at 15:26 +1100, Ryan Newington wrote:
Hi Vivek,
Thanks for the link. I had a look through and couldn’t find anything
that worked. There are no NAT problems as this is all taking place on
my internal network. The rtp.conf is used to configure the ports.
There are no
On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
On Mon, 2007-11-05 at 09:40 -0800, James Moore wrote:
I'm trying to use the MySQL CDR records.
According to dialplan show, the line in the dialplan is:
11. Set(CDR(userfield)=${billing_code}) [pbx_ael]
It looks like the value is being set when I watch the console during the call:
I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico.
This is really the first server I have used with PRI in Mexico as we
normally use MFC/R2. Everything seems to be working except that some
numbers always seem to be busy when you dial them. All these numbers
belong to
On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote:
On a slightly different matter:
http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
1.4.1 .
Yes, I noticed that too and was wondering if it is just because they
have not updated the site or if there is a
On Wed, 2007-10-31 at 09:41 -0600, Anthony Francis wrote:
This also happens if zaptel fails to load. Check your messages file.
John covici wrote:
Well, this happened to me one time when I forgot to compile the pri
library before the asterisk! Could you have done that?
Asterisk
noticed
that there are no PRI commands available on the Asterisk CLI. We cannot use
PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls.
Why would this commands be missing?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel
On Fri, 2007-10-26 at 16:35 -0400, Michelle Dupuis wrote:
I have a new asterisk system with a T1 card. It appears that running
ztcfg -vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk
startup script?
This
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP and then use GSM adapters to get the calls into the
GSM network. My problem is that his customers only want to be
identified by
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote:
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar las llamadas
On Mon, 2007-10-22 at 15:13 -0400, [EMAIL PROTECTED] wrote:
On 10/22/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP
We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1,
Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual
Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two
TE120P cards and everything was working fine. Since they needed to add
a third E1 line we
On Wed, 2007-10-17 at 21:03 +0300, Atis Lezdins wrote:
On Wednesday 17 October 2007 19:09:23 Carlos Chavez wrote:
Why would inserting a multiport card affect Asterisk and the
server? How can I debug this situation? I do not have enough slots to
insert three single cards of the same
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
Is there an example on how to use two E1 ports connected to each other
to simulate connections? Since I do not have an E1 at the office I need
for one port to act normally and the other to act as if it were the
telephone company so I can send calls from one E1 to the other. Someone
has
On Thu, 2007-10-11 at 15:07 +0200, Vincent wrote:
Hello
Has someone used the OpenVox A400P01 (ie. a supposedly
Digium-compatible A400P board with a single FXO module
www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully?
I've put it in an older PC with a Gigabyte GA-7ZX
I have a customer that recently started having a problem with their
Call Center SIP extensions. The problem is that after some time the
caller will hear a triple tone (beep, beep, beep), a 5 second pause,
another triple tone and then the call will be dropped. This usually
happens between
On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote:
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
On Fri, 2007-09-21 at 14:03 -0500, Ricardo Melendez wrote:
Help I need to install asterisk 1.4.X using unicall, somebody can tell
me which are the correct versions of spandsp, libunicall, libmfcr2,
libsupertone, to install with asterisk 1.4, I have installed a
prepatched version, but I need to
when you click a
button. That will fire an event that connects to the manager interface and
uses originate to dial the external call and then dial the internal extension
if all conditions are met. The numbers will be in a database.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
Does anyone know if the Dell Power Edge 1900 has an issue with
multiport E1 cards? We've had this server running for a while now with
2 E1 cards. At first we tried to install an Openvox D210P card with two
E1 ports but after a couple of kernel panics we thought that maybe the
card was
On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
Hi folks:
I know it's come up a few times before, but I need some more detail.
I'm looking for a SIP DECT (cordless) phone for North American
installations. I've heard only of the Siemens Gigaset S450/C450 phones.
Apparently these
I have several installations of Asterisk (several versions) where we
have our own web interface that uses Mysql and Realtime. When we do
modifications to Mysql we use a Manager connection in order to reload
the configuration (we use Realtime static for extensions) sometimes
Asterisk will
On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote:
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with
asterisk using B2B E1. following are the details of my H/W,
zaptel configs and software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk. Is there a
device that
I am having a strange problem with an Asterisk server that has a small
5 seat call center. While everything seems to be working properly I if
do a core show channels the server goes into a loop:
pbxinsol*CLI core show channels
Channel Location State
I was wondering if anyone has an easy way to emulate dialing in a round
robin fashion like when you use Zap/r1 for Zap trunks. At the moment
what I do is simply make a macro that will dial the sip trunks in order
so if the first one fails it goes to the second and so on. The problem
with
On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Shouldn't you
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote:
I am trying to figure out how long a caller waited in queue for
someone to answer versus how long they stayed on the phone after the
answer. I am assuming that the duration is the total talk time and
that the billsecs are the total time in
Here is Mexico the phone company uses a DSL router from 2Wire which in
my opinion is quite bad. I am having problems getting PAP2T adapters
connected to Asterisk using these routers. They connect fine but after
about 5 minutes I get a message on the Asterisk console that the ATA is
On Thu, 2007-08-16 at 16:23 +, John Meksavan wrote:
Asterisk Users,
I have 3 FXO modules with the TDM400P Digium Card. I can dial into the
Asterisk rings my Sip phone, but dialing out with my SPA941 phone through
the zap channel is a problem. I keep getting this message on the
On Mon, 2007-08-13 at 15:25 +0530, [EMAIL PROTECTED]
wrote:
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
I am having a bit of a problem implementing the pickup command in my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and I
can get the call. The problem I have is that when a call enters my
I usually have good results when using a regular fax machine connected
to a PAP2T on a small network. I have a customer that has this setup in
several offices. Lately I have noticed that recent versions of Asterisk
have worse results with this fax setup that onlder versions. I have 3
On Wed, 2007-08-08 at 17:08 +, John Meksavan wrote:
Wes,
What kind of service outages did you experienced?
This would use for my office and I cannot afford for any dropped calls or
poor audio quality, when talking to customers.
My experience with Voicepulse has been good
On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote:
Hi Carlos,
I suggest you download spandsp-0.0.3pre22.
(http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
instead of 1,2,..,9,0,A,B,C,D,E. So,
It seems the problem with Unicall and Nextel is also present in
Asterisk 1.2 and not only in 1.4. I decided to downgrade from 1.4.9 to
1.2.23 so the customer could have CID and calls from Nextel but today he
told me that they cannot receive any calls from Nextel, they get a busy
tone
Here is a log with level 255 when a Nextel phone tries to call in:
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ]
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1
On Fri, 2007-07-27 at 11:09 -0500, Victor Toofic wrote:
Hi,
zaptel.conf:
span=1,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
loadzone=mx
defaultzone=mx
unicall.conf
[channels]
context=incoming
usecallerid=yes
hidecallerid=no
On Mon, 2007-07-23 at 11:47 -0500, Moises Silva wrote:
Alvaro,
Naming Asterisk versions is of little help since Asterisk is not
the one failing here. It would be more helpful know the libmfcr2 and
spandsp versions that were used in the working/non working tests, is
that possible? do you
between both version.
I patched mfcr2.c but I still cannot receive calls from Nextel phones
unless I put ANI to 0 on unicall.conf
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
I have a customer that has recently upgraded their network and now
their Aastra 9133i phones are loosing their connection to the Asterisk
server. They were using an external Asterisk server and now we have
installed a new internal server with Asterisk 1.4.8 on a SIP/IAX
implementation
401 - 500 of 750 matches
Mail list logo
