I've been using the following macro to ring SIP and IAX devices for a
few seconds, and then add on a cell phone if there is no answer on the
SIP or IAX device. Periodic problems began a few versions ago and now
the problem happens every time with 1.2.9 and 1.2.9.1.
The problem is that when a
Hi!
Could somebody help me with pickup feature? I've set
callgroup = 1
pickupgroup = 1
for my phones in sip.conf, but if I try to pickup call with *8
asterisk output to console
Jun 6 15:04:44 WARNING[11857]: pbx.c:2401 __ast_pbx_run: Invalid extension
'*', but no rule 'i' in context
On Tuesday, June 06, 2006 12:10 AM Andrei (MPI) wrote:
I'm using SIP-to-Skype/Skype-To-SIP software gateway called Uplink
(found in Wiki): http://nch.com.au/skypetosip/ - which is free and
working great so far. Downsides are:
Only that it produced not RFC conform SIP headers which are
Hi all,
I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in
my office. the out going calls symptom like when called party pickup the
phone but the calling party still hearing the ring tone from the IP phone.
Please light me up. it been many sleepless night by googling around
I am using syslog-ng, with mysql, and php-syslog-ng, so you get a web interface to search for logs, and a huge capacity on the mysql databse, I have a syslog-ng with the above configuration, and is handleing 5 million syslog message per day.
On 6/6/06, Matthew Warren [EMAIL PROTECTED] wrote:
Does
2006/6/7, James Harper [EMAIL PROTECTED]:
I've thought of this before, but my idea was to have a small box (aboutthe size of an nt1) with an S/T interface on one side and ethernet onthe other. A SBC with built in Ethernet and a minipci slot might do, but
a dedicated device should be able to be
hi,
actully, i need asterisk as a stun client...
have any idea
On 6/6/06, unplug [EMAIL PROTECTED] wrote:
HI,
There is a parameter NAT can be set in the configuration file. Is
it the way that we can use to support NAT by setting nat=yes in the
file instead using other NAT resolving tools
--- [EMAIL PROTECTED] a écrit :
hello,
How asterisk could support res_snmp even this module
don't help to monitor all asterisk features?
monitoring asterisk with snmp would be a good
thing.
Which solution ?
Harry
--- Kristian Kielhofner [EMAIL PROTECTED] a écrit :
[EMAIL
On 6 Jun 2006, at 05:05, [EMAIL PROTECTED] wrote:
Thanks for the info. It would be an external program. I have been
looking at the originate manager command, but it looks like it
would not bridge 2 external numbers. One of the number has to be a
local extension. Suppose I want it to dial
Hi
do you know how to make a cable for powering a POE Cisco Phone from an
not cisco POE Switch ?
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Erick Baum wrote:
The worst ongoing issue has been the echo and the really crappy
speakerphone. The customer is pretty much used to it now. But we're
slowly replacing them with Polycom's as new people come on and as
others just get fed up. Unfortunately one of the phones met it's
doom by
Hi all,
I would like to setup a redundancy/load balancing of an asterisk
system as follow.
Internet DNS asterisk1 -- mysql DB
+---asterisk2 --+
In such case, all user account, dial plan and other necessary
information is stored
I am running 1.1.0.13 and there are no issues which are causing a
problem for us. The speakerphone is not much use but we can live with
that.
1.0.1.9 would stop registering after a while causing incoming calls to
go straight to voicemail.
1.0.2.13 fixed this but had a bug where sometimes
On Wed, Jun 07, 2006 at 10:10:16AM +0200, [EMAIL PROTECTED] wrote:
hello,
How asterisk could support res_snmp even this module
don't help to monitor all asterisk features?
monitoring asterisk with snmp would be a good
thing.
Which solution ?
Harry
--- Kristian Kielhofner [EMAIL
Hello List
We are a VoIP telco, running Asterisk.
We have been having problems with our IAX2 channels for some time now.
Our problems are jitter, and lost packets, resulting in bad audio quality.
The weird thing is, that this mostly occurs on our local network.
We have tested the network with
Yet another set?
I get about 50 downloads a week for mine.
Mark
On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote:
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds and additional sounds (it should
be complete
Jon Schøpzinsky ha scritto:
We have been having problems with our IAX2 channels for some time now.
Our problems are jitter, and lost packets, resulting in bad audio quality.
The weird thing is, that this mostly occurs on our local network.
We have tested the network with pinging an hour,
Hi,
I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and
several SIP phones and ATA's.
We have a terrible delay on calls between the PSTN (isdn BRI) and the
SIP phones. All internal calls are fine. My first thought was that
the transcoding could cause the delay but all of the
Hi All,
I need a suggestion.
I want to run only IAX on two windows based PCs and asterisk
Can you suggest which asterisk, libpri and zaptel versions should i use?
do i need some othermodules also?
Also where will i find the guide to compile astreisk
Actually i have installed,comnpiled and used
http://www.asterisk.org/download
http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS
amna saleem wrote:
Hi All,
I need a suggestion.
I want to run only IAX on two windows based PCs and asterisk
Can you suggest which asterisk , libpri and zaptel versions should i use?
do i need
Hi
I am facing some problems in making calls to
Nokia E60 ,from other sip extensions, I am able to
hear clearly when I use the X-lite clients , but on
Nokia E60 , I cannot hear anything ,ie whenever a call
is made , the user who uses X-lite hears everything
what the Nokia user says , but
I have a client who has about six of these phones. Luckily (for me, not
for them) they were purchased before I came into the picture.
Daniel Salama wrote:
I have heard complaints from my client about the speakerphone and they
are now
You don't notice any problems when using the speaker-phone,
Hello
Be aware that the Nokia E60, E61 and E70 does not support NAT.
Just to be shure that you know that.
A clever choice from Nokia, so that users has to have some local equipment from
the telco.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote:
Well, these are encouraging words :)
You're basically telling me that I should tell my client to buy other
phones. I agree that you cannot compare these phones with Cisco or
Polycom. After all, like you said, what do you
Hi;I've a question:
I use asterisk -R so I can see what's appening in my asterisk and the session of the calls:
I use the vrrp protocol, I use 2 asterisk box;when the master falls down, the slave goes up, and I use X-lite,Phoner,3CXphone;some of this softphones are immediately registered to the
This isn't really a fix for the people missing calls, but one solution I found was to limit the amount of time a call rings for to a cell phone. If it doesn't answer in X seconds, then dial again. This isn't a perfect solution, but helps some.
On 6/6/06, Colin Anderson [EMAIL PROTECTED] wrote:
We
On Wed, June 7, 2006 14:09, Louis-David Mitterrand said:
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote:
Well, these are encouraging words :)
You're basically telling me that I should tell my client to buy other
phones. I agree that you cannot compare these phones with Cisco or
While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000.BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice.Thanks,DanielOn Jun 7, 2006, at 8:09 AM,
A schema for the RJ45 cable pinouts to power a Cisco phones from a non-Cisco
switch can be found on the WIKI here
http://www.voip-info.org/wiki/index.php?page=Cisco+POE
Another option is to use the PowerSense BL-8858-01 PoE Converter, which
converts IEEE 802.3AF to Cisco CDP, and will run you
They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another
Has anyone had any experience with this router??
I am looking to use it because I want to use a DECT phone in conjunction
with VoIP and this seems to check all the boxes for Wi-Fi, ADSL and VoIP
all at a good price.. I have never used Speedtouch hardware before so
any feedback would be
On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote:
While I would agree with you, the price difference between a GXP-2000
and a Polycom 430 or a Thomson ST-2030. These latter units are, at
least, twice as expensive as the GXP-2000.
BTW, I never heard of the Thomson ST-2030,
Hello to all
I had Asterisk dialing the PSTN through a defined trunk.
But when I enabled the SIP URI calls Asterisk stopped contacting
the PSTN trunk
The SIP URI dial code (who created the problem) is this:
exten = _.,1,NoOp(Incoming Call from from-internal-custom extension
Hi Jon
Thanks for the mail
I am just checking NokaiE60 and E61 as PBX client
only , right now , the NAT issue does not arise for my
problem
--- Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello
Be aware that the Nokia E60, E61 and E70 does not
support NAT.
Just to be shure that you know
Ok... I am reluctant to ask this question as I believe that it may be
like asking what someones favorite linux distribution is... but I need
to make an informed decision.
We are getting ready to upgrade from a TE210P to a quad T1 card with
echo cancellation. I am trying to decide between the
Any chance of the resellers details ?
fadge
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Louis-David
Mitterrand
Sent: 07 June 2006 13:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: GXP-2000 (steer clear)
On Wed, Jun 07, 2006 at
I failed to transmit dtmf via voipbuster to the destination.
Does anybody have success, if how to set it up?
bye
Ronald Wiplinger
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On Wed, Jun 07, 2006 at 01:55:04PM +0100, asterisk wrote:
Any chance of the resellers details ?
For the ST-2030 I use this reseller:
http://www.hl2d.com
Sales contact: Jehan-Philippe Le Roy
Responsable des Ventes Partenaires
[EMAIL PROTECTED]
Tel: +33 1 39 51 60 32
Fax: +33 1 39 51 86 91
49
On Wed, Jun 07, 2006 at 08:53:27AM -0400, Sean Cook wrote:
Ok... I am reluctant to ask this question as I believe that it may be
like asking what someones favorite linux distribution is... but I need
to make an informed decision.
We are getting ready to upgrade from a TE210P to a quad T1
Sean Cook wrote:
Ok... I am reluctant to ask this question as I believe that it may be
like asking what someones favorite linux distribution is... but I need
to make an informed decision.
We are getting ready to upgrade from a TE210P to a quad T1 card with
echo cancellation. I am trying to
One of the primary differences between the two cards is the Sangoma
h/w echo canceler handles more cases of echo then do the Digium cards.
Whether you need that additional coverage is 100% dependent on your
specific implementation (eg, your T1/PRI provider), and not on what
the list thinks
many thanks for your reply
i've tried to make a cable with that configuration but it seems that
it doesn't work...
i'm using a 7905G Cisco ip phone and an ALL0484 Switch POE
thanks
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I get the following * Notice ocassionally and I was curious what it means
and if it can safetly be ignored or corrected.
Jun 7 05:40:47 NOTICE[32153]: res_musiconhold.c:511 monmp3thread: Request
to schedule in the past?!?!
Doug
* Doug Crompton *
*
Please can any one help me how to make directories at
[EMAIL PROTECTED]
To use it from the ivr *411
Thanks
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail
Sean Cook wrote:
One of the primary differences between the two cards is the Sangoma
h/w echo canceler handles more cases of echo then do the Digium cards.
Whether you need that additional coverage is 100% dependent on your
specific implementation (eg, your T1/PRI provider), and not on what
the
I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now. However, no sound is heard and I get this message from the CLI
>From the IVR someone will usually be dialing #, not *411 (although I suppose both work). In AAH, the directories are setup automatically when you setup your extensions. Type in the name of the person and the user's extension in the extension setup page, and that person is automatically added to
2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]:
HelloBe aware that the Nokia E60, E61 and E70 does not support NAT.Just to be shure that you know that.A clever choice from Nokia, so that users has to have some local equipment from the telco.Jon
What do you mean by users has to have some local
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and join a conference but when
i do that i
received an error saying nomber do not exist. but if i call a
Hello Olivier
Ive been testing the E61 phone for some
days now, and we need to have an inhouse asterisk server, connected to our main
asterisk server, to get it to work.
That means, that you cant just walk down
to your local airport, and use the IP part of the phone on their network.
Did you check your mpg123 version ?, asterisk needs
a specific version in order to work...
- Original Message -
From:
Richard Reina
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 07, 2006 6:02
AM
Subject: [Asterisk-Users] Music On Hold
not
Flames about GXP-2000 poor quality are frequent on this mailing list.
I recently setup 80 of these (and I'm waiting for other 30...) to move whole
company from a legacy Alcatel PBX to an Asterisk-only solution.
At first, I tried some chinese phones (AtCom) and they were a disaster.
Then I tried
For all the noise about this noone has mentioned one important thing. We should be gratefull that we have access to G.729a in Asterisk, whatever the mechanics of the licensing. It's obvious that its a pain in the [EMAIL PROTECTED] for Digium who absolutely not making ANY on it money for their
If you can't afford to purchase both cards, then a safe bet is to
simply
purchase the Sangoma card since it can address more echo issues then
the
Digium card.
Also, don't forget that the high-end A104d has more than on-board EC.
It has on-board DSP handling and a 5 year warranty. Check it
Hello Mimmus,
* Mimmus [EMAIL PROTECTED] [07-06-06 16:52]:
At first, I tried some chinese phones (AtCom) and they were a disaster.
you talking ybout this phone?
http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2
Has anyone some experience with this phone?
Best regards,
Matthias
On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote:
The (current) problem is that the registration program does not ask which
ethernet card you wish to bind to, nor does it look at the Asterisk config
and use the MAC address of the ethernet card whose IP address is referenced
in bindaddr (as
Hello,
I have done a lot of testing on both the Digium TE406P and the Sangoma
a104d and was involved in debugging both of them with Digium and
Sangoma in their early releases.
Since we are on a Digium-owned list right now and I don't want to be
branded an enemy of Asterisk again for suggesting
* Mimmus [EMAIL PROTECTED] [07-06-06 16:52]:
At first, I tried some chinese phones (AtCom) and they were
a disaster.
you talking ybout this phone?
http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2
Yes
DV
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Hi All,
I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the
phone will just ring and ring, even if I answer the phone on the other
end. Whats strange is that the * phone will continue to ring even after
I've answered and (sometimes) hung up the dialed phone. If I make an
On Wed, 2006-06-07 at 07:55 -0700, [EMAIL PROTECTED] wrote:
For all the noise about this noone has mentioned one important thing.
We should be gratefull that we have access to G.729a in Asterisk,
whatever the mechanics of the licensing. It's obvious that its a pain
in the [EMAIL PROTECTED] for
Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf for version 1.2 say the mpg123 is no longer needed. | Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123
Asterisk Hater.. :) Sorry matt couldn't resist..
_.._
Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom Engineer
IT Administrator
Planet Telecom, Inc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Wed, 2006-06-07 at 11:17 -0400, Ben Klang wrote:
On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote:
The (current) problem is that the registration program does not ask which
ethernet card you wish to bind to, nor does it look at the Asterisk config
and use the MAC address of the
Hello,
I install the latest release of Asterisk, QuadBri driver.
I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for
launch qozap...
zaptel.conf:
---
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=fr
Hello Mimmus,
* Mimmus [EMAIL PROTECTED] [07-06-06 17:20]:
Yes
good to known.
I played with the idea to buy one of these.
You would suggest GrandStream then?
Best regards,
Matthias
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Anydody need some access to polycom ftp server ?
Harry
__
Do You Yahoo!?
En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible
contre les messages non sollicités
http://mail.yahoo.fr Yahoo! Mail
Hi,
I have troubles setting the userfield in mysql ( using asterisk 1.2.8 /
addons 1.2.3 )
I use this in my dialplan:
exten = s,n,SetCDRUserField(SOMEVALUE)
I tried also:
exten = s,n,Set(CDR(userfield)=SOMEVALUE)
But everytime i look at the cdr database the userfield is still empty
Does
hello
How can I configure a bewan phonebox with
asterisk
thanks
issam
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convert the moh sounfile to pcm or sln
save the file to
/var/lib/asterisk/moh/default
set the musiconhold.conf
[default]mode=filesdirectory=/var/lib/asterisk/moh/default
turby@ www.canistec.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
ReinaSent: Wednesday,
On 6/6/06, M.Hockings [EMAIL PROTECTED] wrote:
William Piper wrote: For Problem #2: I'm not sure what you are asking. Perhaps post your dialplan for this
problem we will take a look. bp On 6/4/06, *M.Hockings* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: Problem 2) Incoming sip calls from
Appearntly after letting Asterisk run overnight, the problem is back,
from the inconsistancy of the problem I'm going to assume it's a DTMF
problem, I will try working on that and see if it helps.
The calls are coming in thru a Mediatrix 1204, I guess I will have to
play around with the DTMF
The POE switch needs to support always on. Most switches check the device
for 802.3af support before turning on power. The phones only support the CDP
power activation, not 802.3af. I've used always on POE injectors from
wireless access points successfully with Cisco phones.
Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648]
Leonimar,search check_zaptel at http://www.nagiosexchange.org, there is
a simple plugin that checks alarm status on zaptel interfaces and can be
used with nrpe.
Sample usage:
# /usr/lib/nagios/plugins/check_zaptel -s1 -s2 -s3
ZAPTEL OK: TE4/0/1 , TE4/0/2 , TE4/0/3
#
Hello,
I ran into something similar and found the following in the wiki...
Note : If using cdr_mysql addon make sure to set userfield=1 to in
cdr_mysql.conf. If using cdr_csv, edit cdr_csv.c and (re)compile to enable the user field. This command has no effect if the user field is not enabled.
Anyone try out the Snom 300 phone yet? Seems like a decent price.
On 6/7/06, Matthias Fechner [EMAIL PROTECTED] wrote:
Hello Mimmus,
* Mimmus [EMAIL PROTECTED] [07-06-06 17:20]:
Yes
good to known.
I played with the idea to buy one of these.
You would suggest GrandStream then?
Best
Gruys,
How can I record a specific channel if Monitor doesn't receive it as a
parameter?
Can I do a combination with the ZapBarge app?
I want to record calls in some channels.
Thanks in advance.
Fernando Lujan
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Hi,
I have Asterisk 12.7.1 installed through [EMAIL PROTECTED] CD. and explicitly I
have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database
which in on Windows 2003 Server on remote location.
I tested connectivity through isql and tsql, both utilities are working
fine.
I
On Jun 7, 2006, at 6:42 AM, Doug Crompton wrote:
I get the following * Notice ocassionally and I was curious what it
means
and if it can safetly be ignored or corrected.
Jun 7 05:40:47 NOTICE[32153]: res_musiconhold.c:511 monmp3thread:
Request
to schedule in the past?!?!
I see that on
I'm trying to change the ring tone on my 7960 from the dialplan. I've
tried the example on the wiki but it doesn't seem to work. Something like:
exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer
exten = 3010,2,Dial(SIP/3010,15)
I'm not sure what the Bellcore-dr1 ringer is
Shame on me, that was my trouble, seems like I didn't read enough...
Thanks a lot !
Lewis Agosta a crit:
Hello,
I ran into something similar and found the following in the
wiki...
Note : If using cdr_mysql
addon make sure to set userfield=1 to in
cdr_mysql.conf. If using
i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this?
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On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote:
Hello,
I install the latest release of Asterisk, QuadBri driver.
I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for
launch qozap...
Bad place. rc.local is just about the last place in the init sequence to
On Jun 7, 2006, at 7:35 AM, Jon Schøpzinsky wrote:
x-tad-smallerHello Olivier/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerIve been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to
Michael Collins wrote:
If you can't afford to purchase both cards, then a safe bet is to
simply
purchase the Sangoma card since it can address more echo issues then
the
Digium card.
Also, don't forget that the high-end A104d has more than on-board EC.
It has on-board DSP handling and a 5
Tristan wrote:
Hi,
I have troubles setting the userfield in mysql ( using asterisk 1.2.8 /
addons 1.2.3 )
I use this in my dialplan:
exten = s,n,SetCDRUserField(SOMEVALUE)
I tried also:
exten = s,n,Set(CDR(userfield)=SOMEVALUE)
But everytime i look at the cdr database the userfield
if on freebsd..stop asteriskkillall mpg123cd /usr/ports/audio/madplay/make make installedit musiconhold.conf[default]mode=customdirectory=/usr/local/share/asterisk/mohmp3
application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12then restart asterisk.. Mikehttp://www.theclubvoip.com
On
Hi,
I'm looking for a great tech support person to take over the admin of
our asterisk system. If you are a networking person as well, with some
experience in firewalls and desktop support even better. The system is a
multi-group system with IVR, Follow-me dialing, voicemail, and
Have a customer running a 3rd party PBX
implementation based on Asterisk, not utilizing SIP inbound and outbound calls
I believe are coming through a Digium TDM2402B. They are utilizing
Polycom phones. They are experiencing frequent static on the line, and
overall insufficient volume on
Not that this is particularly an Asterisk problem, but make sure
unixodbc is listed when you do a phpinfo(); , also, you might want to
make sure you have extension=unixodbc.so in your php.ini since you're
compiling it as a shared module.
Hope that helps a little bit.
Wasif wrote:
Hi,
I
What specifically were the voice quality complaints about the spa-841
phones? The only thing I have noticed is calls can be louder than
expected. What else have you seen?
Daniel Salama wrote:
They don't all go down at the same time, or at least, my client hasn't
noticed. I just added the
Glad I could help.
Cheers.
On 6/7/06, Tristan [EMAIL PROTECTED] wrote:
Shame on me, that was my trouble, seems like I didn't read enough...Thanks a lot !Lewis Agosta a écrit:
Hello,
I ran into something similar and found the following in the wiki...
Note : If using cdr_mysql addon make
Pietro U wrote:
i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users
without any registration in the asterisk. how to block this?
Point your default value in sip.conf to an empty context.
Florian
___
--Bandwidth and Colocation
In your sip.conf or iax.conf you need to
change the default context to something that will not interact with your main
dialplan.
_.._
Brian
Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom
Engineer
IT
Administrator
Planet
Jeremiah Millay wrote:
I'm trying to change the ring tone on my 7960 from the dialplan. I've
tried the example on the wiki but it doesn't seem to work. Something like:
exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer
exten = 3010,2,Dial(SIP/3010,15)
Try something like this:
http://www.nytimes.com/2006/06/07/technology/07cnd-voice.html?hpex=1149739200en=0f01d0becf766f0bei=5094partner=homepage
Free to read, but you have to sign up.
Anyone know the details of this caper?
B.
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On Wed, 2006-06-07 at 14:06 -0300, Pietro U wrote:
i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users
without any registration in the asterisk. how to block this?
Remove the guest user from sip.conf and iax.conf
--
Carlos Chavez Prats
Director de Tecnología
On Jun 7, 2006, at 10:22 AM, Cory Andrews wrote:
x-tad-smallerHave a customer running a 3/x-tad-smallerx-tad-smallerrd/x-tad-smallerx-tad-smaller party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are
The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically
thanks! it [EMAIL PROTECTED]On 6/7/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Pietro U wrote: i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this?Point your default value in
sip.conf to an empty
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