Hi,
I've been looking for such solution without lucks. The prompts (either
by Playback or Background or app_queues) will have to complete before
the Dial cmd kicks in, which takes a lot of time.
Please let me know if you become aware of any solutions for this
apparently obvious problem.
Regards,
H
Hi Markus
I did the same to test one application on intelligent cards from
Pikatechnologies. I had no E1 in office so I set up one Asterisk box with
TE110P to simulate a CO.
The only thing is to change support for protocol.
In ZAPATA.CONF of one system use(the real PABX):
signalling = bri_cpe
on
Yes I know this project however my goal would be
something like this :
[FAX]
PSTN--[VOICE]--ASTERISK--(e)groupware
[SMS] |
Mail Server
So (e)groupware' clients should be able to
send/receive
voice messages fax and sms from/to e-mail click to
dial
contacts
I'm unable to register soft phone on * that is behind NAT. I have SP on public
address 195.29.109.0 which is dynamically changed. * is in private address
10.0.0.81 that is behind NAT on address xxx.xxx.xxx.xxx
When I try to register this is message that I receive on * CLI
# Testing 195.29.1
Thanks for your advises
> > it's no what i expect the easier solution you
> provide
> > the more customers you get !
I don't agree you ! the best solution you provide the
more customers you get (apache projects) !
> Indeed. However, I tend to be of the opinion that
> you should have enough
> mo
Julian -
What hardware are you using? Proc, RAM, SCSI or IDE, etc.
The reason I ask is that I have multiple hardware platforms, all on FC1 or
FC4, and none of them hit 100% for each IRQ. I am usually in the high 98%
with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU).
Two
The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.
Based on your post, seems that you have an issue with codecs more than
creating an IAX trunk.
What version of Asterisk are you
Gervais de Montbrun ha scritto:
I downloaded the chan_sccp as you suggested, but it does not seem to
support my Cisco 12 SP+. I can see that it would support the ata, but
if it doesn't support my other phone, then I need the skinny protocol
and then can't use sccp... :-(
the 12SP should wo
You can play music instead of providing a ringtone. ( I think it's the M
option for the dial command)
We used this for a reception solution so that the caller would not know that
they were not being ignored.
PaulH
- Original Message -
From: "Hugh Jackman" <[EMAIL PROTECTED]>
To: "Asteri
On Thursday 10 November 2005 10:55, Jason Walker wrote:
> The statement of zaptel being required is strange...I use IX trunking
> exclusively for my servers. Two of them have no zaptel/Digium hardware and
> the trunk calls are fine.
I don't know where I read it, apparently it is needed for timing o
Hi to all,
i have installed the latest CVS asterisk version as well as the
asterisk-oh323-0.7.3.
I have also installed the openh323-v1_17_2 and pwlib-v1_9_1 ( i also tried
the Mimas patched oh323 and pwlib but they did not behave well as far as the
gatekeeper registration was concerned).
Hi all,
I am trying to test Asterisk with TE210 and SPANDSP. So i connect
back-to-back (with E1 crossover cable) the two E1 ports of the TE210
and it seems that everything is fine. The i create a script that
calls an extension that starts the rxfax application and initiate
another extension t
Hi all,
I am trying to test Asterisk with TE210 and SPANDSP. So i connect
back-to-back (with E1 crossover cable) the two E1 ports of the TE210
and it seems that everything is fine. The i create a script that
calls an extension that starts the rxfax application and initiate
another extension t
Sorry for posting many times
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h
I have just upgraded my server to Asterisk 1.2.0.rc1 from the beta1
release. Most seems to work just fine, except for endpoints trying to
use h263 as video algorithm. Result: Audio is ok, video NOK.
Anyone else with the same problem? Tips on how to fix it?
Trond
__
Hello,
Does asterisk's team will improve IM and presence in
asterisk-1.2 !
Send Sip MESSAGE is impossible.
When the buddies status change nothing is happened.
How asterisk's team plan to solve this problem ?
Regards
Harry
___
Hi asterisk lovers,
Does anyone know a good trancoder to produce g729 files from gsm or wav.
Regards,
Olivier
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It's all in the email, just look a little lower ;)
hint:
HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned
off).
OS: CentOS 4.2
Dual Embedded NIC enabled
USB disabled
serial disabled
printer disabled
2x73GB SCSI in HW Raid 1
Jason Walker wrote:
Julian -
What hardware are
> A nasty screech. That's what callers here sometimes when they dial into
> my FXO port from the PSTN. But usually, it works OK.
>
> Is this common?
That was fairly common on the original TDM cards (rev E/F) with older
drivers. The problem would usually show up after the card has been
in oper
Please post a bug on bugs.digium.com with a full sip debug trace with
verbosity of at least 4 and a debug level of at least 4 so we can
track down and fix any possible bug before 1.2 is released.
Thanks.
On 11/10/05, Trond Andersen <[EMAIL PROTECTED]> wrote:
> I have just upgraded my server to
Quoting Matt Riddell <[EMAIL PROTECTED]>:
They are not DTMF tones they are 1100Hz, 400Hz and 440Hz tones, used in call
shop systems. They monitor call progress and trigger billing.
Regards
Obelix
> Obelix wrote:
> > Quoting Matt Riddell <[EMAIL PROTECTED]>:
> >
> > Is there a way of converting
> Yes, but the problem is, I think from a T1 theoretical perspective,
> that because the T1s are from different providers, their timings may
> be different. I would assume that I need to be able specify a timing
> source per provider. Correct?
No, all real telco's will sync against a higher
Harry,
The monitoring of buddies on Polycom phones is possible with the
release candidate for v1.2. We've asked for a sip debug/trace from you
to try and troubleshoot your problem, and you haven't provided that to
date.
On 11/10/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Does ast
Rich Adamson wrote:
Yes, but the problem is, I think from a T1 theoretical perspective,
that because the T1s are from different providers, their timings may
be different. I would assume that I need to be able specify a timing
source per provider. Correct?
No, all real telco's will sy
Hi,
Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ?
http://www.its-tel.com/main/home/doc.asp?mCatID=1977&mCatPID=1972&tpMID=0
They appear to be very favourably priced...
Rgds
Pete
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Olivier;
Merci pour ta réponse, le problème était au niveau de mon zapata.conf, il fallait que je rajoute la fonction overlap=yes, parcontre je ne passe pas par France Telecom je remplace plutot France telecom, en gros:
Pabx ericsson connecté avec sa carte E1 directement sur asterisk avec la
> I've been trying to set up my Polycom phones to get the boot server info
> (tftp-server-address) from DHCP on a Cisco router. I've previously just
> specified it manually on the phone, and that works well enough, but I need
> to change now (because of the number and geographic locations of the
I have solved one part of the problem. I'm able to register. I'm able to call
SIP phones and I can hear them. The only problem is that they can't hear me.
So, this is the situation.
Softphone_1 (on public IP) => Internet => Router => * (private IP) =>
Softphone_2 (private IP)
SP_1 can call and
I did it !?
//
Connected to Asterisk 1.2.0-rc1 currently running on
serveur1 (pid = 1125)
Verbosity is at least 4
serveur1*CLI> sip show subscriptions
Peer UserCall ID Extension
Last state Type
192.168.0.21
I would say your best bet is to change your system into a distributed
dialing system. We did this with Vicidial and have installations on
multiple servers with over 100 agents all working off of the same
lists and campaigns. A distributed system will also allow for more
redundancy and less total do
I have a question. Is there any way to have a caller entering a Queue to
go to voicemail if there is only one Agent and that extension has the
phone set to DND? We have one extension that is the primary service
technician and have it set to always be a member / logged in, so he
cannot just logo
Hi,
I tried hunting for a little more info. I think all that happens with
this is they use the Q.421 spec for handling the ABCD bits, and then
simply send the DNIS through as DTMF after the seize if acknowledged.
That means they loose some of the functionality of real R2 signalling -
e.g. no
Hello
I am still new to Asterisk, but looking at some
products to offer small and medium sized buisnesses.
Is it possibel to have the sip ”ends” talk
directly to eachother? Have authorisation and call setup on the asterisk, but
leave the actual conversation p2p?
BR
Amund Nygaard
> What kind of DTMF method signaling is "AVT" ?
The sipura admin manual refers to AVT as a.k.a rfc2833.
> My Sippura seems to support only InBand, AVT, INFO, InBand+Info, Auto
> INFO does not work with Asterisks voicemail system so it is useless for
> me.
Auto works just fine for me.
> InBand -
how much is that per pc.?
On 11/10/05, Pete Barnwell <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ?
>
> http://www.its-tel.com/main/home/doc.asp?mCatID=1977&mCatPID=1972&tpMID=0
>
> They appear to be very favourably priced...
>
> Rgds
Hi,
Users of my MFC/R2 software may be interested to know that new versions
are available. These fix a bug where a timer was not always correctly
cancelled. The result could be the locking up of a channel. You can
download the updates from http://www.soft-switch.org
Regards,
Steve
_
> I followed your steps to the letter but after resetting to factory defaults
> unfortunately it still doesn't record the configuration changes I do.
>
> 2005/11/9, Adam Moffett <[EMAIL PROTECTED]>:
> > If you unplug the ethernet cable on a Sipura SPA and then reset the
> > power it'll boot up in
Anyone out there got a SIP phone (mine's a Cisco 7940) to work through a
VPN with a Netscreen 5gt? It has always worked for me with any ScreenOS
version 4.x. I had the need to upgrade it to ScreenOS 5.x and it breaks
the phone. Here's the goofy part, it works enough to still register
with th
yes
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Amund Nygaard
Sent: Thursday, November 10, 2005
8:19 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Call p2p
Hello
I am still new to Asterisk, but looking at some products
to offer smal
I have a question which may be about the SIP protocol, or may be about
SIP features supported in Asterisk, I don't know.
Let's say I have three Asterisk boxes, A, B and C, which pass calls to
each other using SIP.
A call comes into box A from somewhere, and A determines that the call
should be ro
James Armstrong wrote:
> I have a question. Is there any way to have a caller entering a Queue
> to go to voicemail if there is only one Agent and that extension has
> the phone set to DND? We have one extension that is the primary
> service technician and have it set to always be a member / logge
Hi all,
i'm still experiencing a one way call only between a
ipPhone and an analog one through a oh323 channel
between my asterisk and a Nortel GK.
Doing some sniffing and some debug with ethereal and
tcpump i can say (i hope, as newby to say the right
thing) that i can't see any rtp traffic
betw
I've think I've been working on the same thing. Many SIP phones have a
built in conferencing feature...but they may not all work the same and
may have all different instructions. So doing it in asterisk is
preferable to me so I can give users one set of instructions for it.
It's not a simple
This is good debugging info you've listed below, but this isn't a sip
debug/trace.
To do that, first verify in your logger.conf file you have the following line:
full => notice,warning,error,debug,verbose
Then, if you needed to add anything to logger.conf, please first
restart Asterisk so th
Olle has said he has a working patch for this scenario, but it will
be a couple of weeks yet before it's ready to be merged into the HEAD
tree so it will be a post 1.2 thing.
On 11/10/05, Tony Mountifield <[EMAIL PROTECTED]> wrote:
> I have a question which may be about the SIP protocol, or may b
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have
listed below.
set policcy id 1001 from "Trust" to "Trust" "Local" "Remote" "SIP"
permit log count
set policy id 1001 application "IGNORE"
set policy id 1002 from "Trust" to "Trust" "Remote" "Local" "SIP"
permit log count
se
Assuming an XP or 2003 box, I use the free xlite client. Create a user
for each instance that you want to run. Right click on the shortcut and
select run as...
enter the username and password of the account, setup the settings for
the phone, and repeat the process for each additional instance.
I am in search for a terminal emulation application like
securecrt, putty, or penguin that can use SIP. It can be either linux or windows
application.
Thanks,
Chip
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Looks interesting. Analog single port only, though, so you would be subject
to the vagrancies of a TDMXXX analog card. A VoiceBlue gateway is SIP so you
can do IP-only until it hit the GSM network, and they aren't that expensive,
$2500 US.
My VoiceBlue is stuck in customs! Chomping at the bit to
Hi I've got an interesting problem.
A few days ago (maybe even a week or two) all my sip phones lost
registrations with my asterisk box. All that is but one.
The asterisk box is out on the internet, I have two phones at my
location and 1 at another separate location.
The only phone that rem
Tried that. The queue has a static agent of SIP/107. When calling the
queue it shows 107 as being BUSY (DND enabled). The caller just stays in
the queue. What I really need is to have the caller stay in queue when
the extension is busy (because that is that queues are all about), but
have the c
I am looking for a simple dial plan for if my zap channel is
busy/unavailable send to Voicemail. I couldn’t find anything simple online.
-chip
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Hello James,
you could approach this problem in many a way. I'd suggest to make your
support guy log on to the queue using AgentCallBack and enforce
"joinempty=no" in the queue itself. When your agent goes to lunch, he logs
off and people cannot join the support queue anymore, so you move t
Goal:
I would like to use the cheap cellular phone from my "family share"
plan to add an * trunk. With this, nights and weekends are free as is
cell(*) to cell.
As an * noob, I have been scouring the threads for information on using
a cell phone as a trunk (not a handset). Aside from using an an
Just to clarify this in my head :-)..
So...
They are using E1/R2 (the R2 Digital)in fact, for all the line signaling
(nothing unusual)
The register signaling, that I was under impression would be MF in each
timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF
in this trunk
Hi folks,
I have an asterisk server behind a NAT'd gateway that is using iptables.
Internally, I have no problems connecting to asterisk. I would like to be able
to use a sip softphone from outside the gateway, and become an extension on my
asterisk PBX.
I have a laptop running X-Lite. When
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
Is pretty
simple. Replace the 102 priority with a call to voicemail and you’re set. hth
-Original
Message-
From: Pleasants Email Lists
[mailto:[EMAIL PROTECTED]
Sent: Thursday, November 10, 2005
8:22 AM
To
This seems to be what Cisco have implemented as r2-digital-dtmf-dnis.
Cisco have quite a few other combinations of strange R2 related options.
I can't imagine they are all really used. It seems this one is, though,
in Venezuela
Regards,
Steve
Julio Arruda wrote:
Just to clarify this in my
Hi.
Im
having sound quality problems using the new BT 101 and 102 models (the ones with
solid colour bottoms like the gxp model). Im using firmware
1.0.6.7.
Does
anyone as the same problem with these new
models?
Sound
quality has no cuts or noise. But the sound is much more low
Thx Steve!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Steve Underwood
|Sent: Thursday, November 10, 2005 7:24 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users] New revision of my MFC/R2 software availabl
Hi,
Just so I am clear for version 1.2 has chan_modem.so been depreciated?
That means I should also remove this module from loading in the
modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to
replace this functionality (I do not really understand what
chan_modem.so was use
On 11/10/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Just so I am clear for version 1.2 has chan_modem.so been depreciated?
> That means I should also remove this module from loading in the
> modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to
> replace this functionali
I have a TE110P that I
will be connecting to a T1 PRI. This seems pretty standard, but I am only using
7 channels for voice. It’s a shared voice/data T1; 7 channels voice, 16
channels data and 1 D-chan, it comes into a telco router and is split into a
voice PRI and an Ethernet connection. T
On voip-info.org there is a claim that asterisk and a BCM can interconnect
via H.323. There is little on the page beyond setting the H.323 connection
on the BCM to "other". Hardware restrictions at the moment make the H.323
solution preferable to ISDN or SIP. I am using oh323.
Every time that
Colin Anderson wrote:
Forrest: Any secondary effects you can see from running SP on an SMP kernel,
any bitching from dmesg at boot? Cool hack.
Nope... no other side effects I can tell. Of course, it boots like a
SMP kernel (looking at the processor table and all).
-forrest
___
Hello BJ & all ,
On Thu, 10 Nov 2005, BJ Weschke wrote:
On 11/10/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
Hi,
Just so I am clear for version 1.2 has chan_modem.so been depreciated?
That means I should also remove this module from loading in the
modules.conf if I am using Asterisk 1.2
Is there some kind of limit to the number of TDM04B cards you can use in
your Asterisk system (Red Hat 9, kernel 2.4, Asterisk
CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8
analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines
but the third card (rev
Lists Pleasants wrote:
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have
listed below.
set policcy id 1001 from "Trust" to "Trust" "Local" "Remote" "SIP"
permit log count
set policy id 1001 application "IGNORE"
set policy id 1002 from "Trust" to "Trust" "Remote" "Local"
1.2-beta2 is more efficient against echo issues with ECHO_CAN_MG2 :-)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jon Reynolds
Envoyé : jeudi 10 novembre 2005 08:58
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Use
Shaun Singh wrote:
Is there some kind of limit to the number of TDM04B cards you can use in
your Asterisk system (Red Hat 9, kernel 2.4, Asterisk
CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8
analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines
bu
I am looking for sip phones which do not have keypads but only a
ringer/light for use in factories, outdoors, etc.
--
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
- -
- Jason Pyeron PD Inc.
Lists Pleasants wrote:
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have
listed below.
set policcy id 1001 from "Trust" to "Trust" "Local" "Remote" "SIP"
permit log count
set policy id 1001 application "IGNORE"
set policy id 1002 from "Trust" to "Trust" "Remote" "Local"
On Thu, Nov 10, 2005 at 12:57:45PM +0800, Dinesh Nair wrote:
>
>
> On 11/10/05 08:52 Pablo Allietti said the following:
> >yes but both of them have problem with voice. some skype too anybody can
> >have this problems in freebsd? i hear cutted conversations`:
>
> perhaps there's contention for y
Guys.
I just discovered a bug in rc1, whenever We try to do an addqueuemember,
asterisk core dumps.
Here is the dialplan:
exten => 766,1,AddQueueMember(Ventas)
exten => 766,2,AddQueueMember(Soporte-Tecnico)
exten => 766,3,AddQueueMember(Soporte-Contrato)
exten => 766,4,UserEvent(Agentlogin|Agent:
Can this be done?
I have a customer service que that if full go to v-mail.
I would like to know how I can put two e-mail address for it to go to.
Is that possible?
Thanks!
-J
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Anton -
Thanks for the report. I've just posted a bug for you on the bug tracker at
http://bugs.digium.com/view.php?id=5705
Please refer to that URL for further information/resolution.
On 11/10/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Guys.
> I just discovered a bug in rc1, whenever We
De: André Rodrigues ( Cheyenne)
[mailto:[EMAIL PROTECTED] Enviada: quinta-feira, 10 de
Novembro de 2005 16:18Para:
'asterisk-users@lists.digium.com'Assunto: Sound quality of the new BT
101 and 102 models
Hi.
Im
having sound quality problems using the new BT 101 and 102 models (th
I have a request to have an extension to ring silently or different
When a call comes into a queue. This extension is a manager that is
monitoring the queue that the customer server is taking calls in.
Is this Possible?
-J
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Hi,
Just pulled out the BT-102 because I need to use it again, entered in
the TFTP server to get the latest firmware so its now in 1.0.6.7 and i
now was to factory default the phone and set it up from scratch..
I tried the instructions (copied below this message) from the latest
available ve
I apologize if this question has been asked before. Did something change
with the behaviour of the 'sip show inuse' command between 1.0.9 and
1.2-rc1? I used to be able to see a list of extensions and the number of
in/out calls. Now it just reports:
asterisk*CLI> sip show inuse
* User name
If you are using Sendmail you can alias a single email address to multiple
email addresses:
http://www.uwsg.iu.edu/usail/mail/aliasing/
If you are using Exchange you can create a distribution list with a single
email address that expands to multiple recipients:
http://imanami.com/support/viewer.
At 11:27 11/10/2005, Jason Brashear, wrote:
>Can this be done?
>
>I have a customer service que that if full go to v-mail.
>I would like to know how I can put two e-mail address for it to go to.
>
>Is that possible?
You can type in the emails and see if it works. I think
I tried, but didn't have
The example I gave was going over a VPN with tunnel terminating in the
trusted zone. Put the polices how our traffic traverse through the
netscreen. I would config a policy for trust to untrust traffic and for
untrust to trust or untrust to global if you have MIPing going on.
-chip
-Origina
Receiving faxes with Asterisk.
Is there a good resource for learning how to set this up?
-J
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Is anyone else having all IAX peers die right after receiving this in
the log? I have CVSHEAD from about 2 weeks ago. Packet capture shows
Asterisk stops transmitting all IAX packets after this messages appears.
- Dustin -
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Thx BJ, Ill monitor the bug there in case more info is needed.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|BJ Weschke
|Sent: Thursday, November 10, 2005 11:30 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asteris
Does anyone know how to ignore/send straight to voicemail all calls with invalid or no CID?
Thanks
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I still can’t get it to work. My
traffic will be coming from the PSTN (Zap/1) into one context and will Dial a
SIP extension in another context. I have tried making the changes to both
without luck. In the example exten => s,3,Dial(${theChannel}/12345678)
confuses me. Why am I dialing the
Hi,
I get the following when I reload:
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
Nov 10 10:57:34 WARNING[4475]: chan_zap.c:10816 setup_zap: Ignoring
signalling
Nov 10 10:57:34 ERROR[4475]: chan_zap.c:10249 setup_zap: Unable to
reconfig
Hi,
I am getting the following from the Asterisk console:
Nov 11 10:33:50 NOTICE[3578]: pbx.c:1747 pbx_extension_helper: Cannot
find extension context 'default'
Nov 11 10:34:10 NOTICE[3578]: pbx.c:1747 pbx_extension_helper: Cannot
find extension context 'default'
I installed the Asterisk sam
Is anyone using these high-density TDM2400P cards? I'm cautious about using
anything that's brand new.
Regards,
Shaun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Becker
Sent: November 10, 2005 11:59 AM
To: Asterisk Users Mailing List - Non-Commerci
On Mon, Nov 07, 2005 at 01:23:26PM -0500, Branko Samardzic wrote:
> I am trying to tweak my Asterisk servers to talk to each other using Speex
> codec.
> I downloaded and installed speex and speex devel libraries, recompiled
> asterisk (including make clean), did set up speex codec as only one allo
Hi,
I am using 1.2rc1 and my modules.conf looks like this:
[EMAIL PROTECTED] ~]# vi /etc/asterisk/modules.conf
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preloa
I'm using IAX2 with VP-320I hardphones for remote users. Everything seems to
be working fine except for call transfer. Is this an issue with the IAX2
itself or the phone? If I flash the same phone with SIP, the problem
disappears.
Regards,
Shaun Singh, Manager
Travelwave
1655 Dufferin Street, Sui
Hi,
I get the following during a reload with Asterisk 1.2rc1
Nov 10 11:07:31 WARNING[4568]: pbx.c:3757 ast_merge_contexts_and_delete:
Requested contexts didn't get merged
-- Reloading module 'codec_gsm.so' (GSM/PCM16 (signed linear) Codec
Translator)
== Parsing '/etc/asterisk/codecs.conf'
Hi,
Someone have running a MTNT,SIP and Asterisk please let me know really I
don't know which way to take.
Greetings,
JC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Cesar Pinto
Sent: Wednesday, November 09, 2005 3:56 PM
To: asterisk-users@lis
You are probably not going to find a ip phone that does that. I
recommend taking a look at http://www.vikingelectronics.com, they have
a number of emergency/hot phone type devices. Then you would simply
plug it into a Sipura SPA FXS configured to dial a number when it
senses off hook.
We are lookin
On 11/10/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Thx BJ, Ill monitor the bug there in case more info is needed.
>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |BJ Weschke
> |Sent: Thursday, November 10, 2005 11:30 AM
> |To: Asterisk Users
On Thu, 2005-11-10 at 17:36 +, WipeOut wrote:
For example, the Mac address is 000b8200e395,
User should encode it as 000222820095 .
Step three: Access the phone screen menu, then select the -- reset --"
with the up or down arrows keys.
Step four: Dial in the encode of the Mac a
Anyone with SER knowledge could you point me in a direction to setup SER to
rewrite the
SIP URI?
Currently I have the following
[EMAIL PROTECTED]
I am setting it so it does the change but its still showing up with the prefix.
I need it to look like this:
[EMAIL PROTECTED]
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