http://lists.digium.com/pipermail/asterisk-dev/2004-September/006163.html
- Original Message -
From:
Boris
Bakchiev
To: asterisk-users@lists.digium.com
Sent: Monday, April 18, 2005 1:31
PM
Subject: [Asterisk-Users] Digium G.729
vs. IPP G.729
Hi,
Rod,
Here is my macro for this:
[macro-sipexten]
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
exten = s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten = s,2,Dial(${ARG2},${NATIMEOUT})
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,102,Goto(s,350)
exten =
I assume you'll be using IAX2 to connect all the servers? In each case, all
you need is to match the pattern for the extension then send the call to
another * server for final processing. If you only want to maintain this in
one place, you could use ARA (Asterisk Realtime Architecture) and
This could be any one of about 1.32 million things.
Did the PC work OK before you put RH9/Asterisk on it? What sort of BRI card
is it? Have you tested the card under another application/OS/platform? What
version of Asterisk are you running? Is the BRI card sharing interrupts with
anything else?
Thanks Boris. I think I can follow that logic!
- Original Message -
From: Boris Bakchiev [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 18, 2005 4:17 PM
Subject: RE: [Asterisk-Users] Dynamic Dialplan -
Jesse Guardiani [EMAIL PROTECTED] writes:
I don't know about X-Lite, but sjphone seems only to support OSS. One
of my requirements is ALSA support. Thus linphone and gnomemeeting.
But, interestingly, gnomemeeting seems to be the only client capable
of full duplex audio using
Bruno Hertz wrote:
Jesse Guardiani [EMAIL PROTECTED] writes:
I don't know about X-Lite, but sjphone seems only to support OSS. One
of my requirements is ALSA support. Thus linphone and gnomemeeting.
But, interestingly, gnomemeeting seems to be the only client capable
of full duplex audio using
On Mon, 2005-04-18 at 16:18 +1000, Rod Bacon wrote:
I assume you'll be using IAX2 to connect all the servers? In each case, all
you need is to match the pattern for the extension then send the call to
another * server for final processing. If you only want to maintain this in
one place, you
Anyone tried to build app_conference lately?
I'm trying to setup a large conference where i speaker can talk to many
listeners, for example 1 speaker and about 100 listeners (who can not speak
back to the speaker, 1 way audio only)
However, if i try to build app_conference against 1.0.6 or
+++ Min Hwan Chang [16/04/05 12:48 -0700]:
Vikram,
Would they really be able to tell if I have VOIP and POTS terminating
on the PBX? Theoretically, its not like I'll be using this 100% of
the time for sending VOIP calls to the POTS line. Probably maybe once
or twice a month? It's main
Brian Capouch [EMAIL PROTECTED] writes:
Bruno Hertz wrote:
Jesse Guardiani [EMAIL PROTECTED] writes:
I don't know about X-Lite, but sjphone seems only to support OSS. One
of my requirements is ALSA support. Thus linphone and gnomemeeting.
But, interestingly, gnomemeeting seems to be the only
Yes, sipuras work well in Russia.
Actually, they're so configurable that I think they'll work everywhere.
You'll need to re-configure to make them detect/generate Russian tone
standard.
snacktime wrote:
Will sip/iax devices designed for European use also work in Russia?
I'm specifically looking
tgj wrote:
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work. The
problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
tgj wrote:
Thorben,
I hope you find some time to make all more smoothly. It is a great
product, but there are still some unclear things.
3. One IAX2 is simple to taken
The three lines in Exensions / Extensions tab look like:
IAX2 623 IAXy at home 623 Unspecified
Good day all
I have a analog gsm router and a 4 port bri card:-)
How do I get the gsm router to work with asterisk
I tried adding a voicetronix card but the 2 cards doen not seem to work
together,it gives a unresolved symbols error when starting up
Any Ideas Please
Can you add 2 zaptel
Hello everyone.
How was your weekend?
Anyway...
'Got SIP response 302 Moved Temporarily back from 192.168.10.24'
Lately I've been getting this error... well i am at a loss as to why I am
getting this when on Friday I was able to make a pass-through call with no
problems.
+--+
I believe you need to modify a little bit member.c file
in CVS version they use cid, but in stable version callerid.
Just replace properly cid with callerid.
It should help with that problem.
For example:
chan-cid.cid_num change to chan-callerid
On Mon, 2005-04-18 at 10:04, E rikje wrote:
Hi,
Can you add 2 zaptel device,different ones?
Like the Junghannes and a diguim analog card?
Please help and advice
yes you can. use fxo port cards for this.
Matteo.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Got some debug info... please see attachement.
Quoting [EMAIL PROTECTED]:
Hello everyone.
How was your weekend?
Anyway...
'Got SIP response 302 Moved Temporarily back from 192.168.10.24'
Lately I've been getting this error... well i am at a loss as to why I am
getting this when on Friday
Sorry- Solved my own problem. I was playing around with the GS BudgeTone 100
and had set up call forwarding on...
-- SIP read from 192.168.10.24:5060:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b
From: asterisk sip:[EMAIL PROTECTED];tag=as4a953271
To:
Hi List
I am working with a pilot project for a Norwegian regional government to
evaluate Asterisk for a large number of sites and users. The goal of the
project is to have a unified VoIP-system to replace the disorganized
collection of legacy PBX in use today.
By distributed organization I mean
Hello all.
I have a problem with Cisco 7970. At startup this device asks for CTL
(Certificate Trust List) file,
and startup process stops. I am even can't boot this device. Does anyone
know how to avoid this problem?
It is said that in older versions of SCCP dummy file with the name
CTLSEPmac
Hi list!
When doing a new install of AMP I get this error:
Configuring install for your environment../usr/src/AMP/apply_conf.sh: line
67: /usr/sbin/amportal: Permission denied
OK
Is this something I should be worried about?
By the way, I have created some install scripts to download spandsp,
http://www.asteriskguru.com/xlite.html
/Z
Vaniah Voip wrote:
Vamsi Pottangi wrote:
It would be easier if you could get send us your sip.conf entry and
confiuration made in x-lite
Also, please let us know where exactly the problem is. Is it
while registering the x-lite or during the call and the
Dear Richard,
On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote:
The latest firmware for optipoint420 advance SIP seems to be version
4.0.22A, released for HiPath8000.
thanks for this information. I've contacted my customer adviser at
Siemens, he'll try to organize me this version.
What siemens
As only one individual, I thought their statements were very straight-
forward and clear. Having worked as a senior manager in a technical
organization, a large number of tehcnical people simply do not
comprehend some words (or read other words into whatever they happen
to be reading), or, jump to
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more home user type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the
Hi Franz,
ok, can you please inform me (the list) if the Optipoint 420 with the firmware 4.0.22A work with Asterisk. If so i will try to contact our contact at Siemens and organize some Optipoint 420.
chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree
Hi folks,
I'm still having troubles with broadvoice. I can either make calls or
receive calls but not both. It all depends upon how I setup the SIP stanza.
Here's my incoming settings (these allow me to receive calls)
register = 9738281625:PASSWORD:[EMAIL PROTECTED]/
[broadvoice]
On Mon, Apr 18, 2005 at 10:02:48AM +0800, Eddie wrote:
So the Panasonic extension dialed by Zap/3/206 command will ring and
Zap/4/221 will not ring at all, even before ext 206 is picked up?
Yes, exactly. Zap/4/221 won't ring at all.
If you have two extensions numbered 211 212, why are
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in
On Sun, Apr 17, 2005 at 01:50:56PM -0700, snacktime said:
On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:
I have been trying a did company for a few days. I find the service
decent, but sound quality only moderate.
Rather than spending 35 or so for monthly with did, I
Eivind
Most obvious solution is snmp. Using snmp you can collect statistics and
provision your remote systems. However, SNMP is an enabler and not the full
solution. You still need to write SMUX agents and develop application MIBS
that allow you to get/store application specific data. To my
Sorry! Got it! All set.
On Sun, 17 Apr 2005 15:54:37 -0400, SCollins [EMAIL PROTECTED]
wrote:
Just curious what syntax did you use to load the VMware tools on Fedora
Core 3?
Thanks,
Sean
On Sat, 16 Apr 2005 16:50:56 +0200, [EMAIL PROTECTED] wrote:
I installed asterisk 1.0.7 successfully on
Hello,
I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation.
We have a call center with 4 agents, which should receive calls from their queue. But we also have a "call center management" team which should be able to
Are you behind a firewall? If so, did you NAT an IP
to your * machine with a port forward for yourIAX
port?
Have you done IAX2 debug? Help iax2 should get you
the correct syntax.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
MasonSent: Thursday, April 14, 2005
Hi guys,
Any other ideas on this one ?
Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___
Computers are
Hi, does anyone here tried using wcte11xp
(e1) for R2 signaling. I need help because I cant make libsupertone,
linunicall and libmfcr2 work. Im getting an error every time I issue the command
make. Btw, the R2 variant is Philippine R2.
Please help.
Thanks.
Angelo
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in
No, it's in a datacenter. The IAX stuff is working, just
not registering. I did debug it, all it says is
"UNAUTHENTICATED"
Chris Mason
www.anguillaguide.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
SilerSent: Monday, April 18, 2005 9:20 AMTo:
Not to ignore the fact that this is the cheapest and installtion free
VOIP device that you can use for a real conversation, without bothering
about the protocols.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Mahler
Sent: Saturday, April 16,
Hey Everyone,
I've been running a version of the CVS without issue until late last
week when suddenly Asterisk would randomly hit 99% CPU and stop
registering my DIDs.
If I stop Asterisk with a 'stop now' and restart Asterisk all is
well... for a bit.
So far I have deducted the following.
I have a company wants a pbx that has follow-me type rules, i.e., the user
has a series of contact numbers comprising of home numbers, overseas
numbers, cell phone numbers, and they are dialed in sequence. This is easy
enough but the option they want that I am having trouble with is the ability
Hi,
Il giorno lun, 18-04-2005 alle 21:28 +0800, Nathaniel Angelo A. Torres
(247talk) ha scritto:
Hi, does anyone here tried using wcte11xp (e1) for R2 signaling. I
need help because I cant make libsupertone, linunicall and libmfcr2
work. Im getting an error every time I issue the command
Title: Untitled Document
Hello list,
Could you tell me if you ever succeeded in configuring Cisco 7940 and
chan_skinny. How ? (I cannot configure my phone, almost any submenu is
unavailable)
Thx.
--
Thomas RULMONT
Responsable Commercial
Alterys SA
T. +32
87 325939
T. +32 486 863216
E.
On Apr 13, 2005, at 5:01 PM, Andrew Kohlsmith wrote:
On April 13, 2005 03:42 pm, Trent Tuggle wrote:
The symptom is a loud, brief buzz, almost exactly every 6 seconds, on
the dot. It is only audible to remote parties, when I use an analog
phone connected to my Digium TDM card. All other audio
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio
Sent: Monday, April 18, 2005 9:16
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] queue -
transfer calls
Hello,
I am setting up an ACD using *, but found a an issue
Hi martin. Maybe setting callprogress=no and busydetect=no, or
increment the busycount parameter. all in zapata.conf
you can read more about these parameters in the wiki at voip-info.org
best regards
- moy
On 4/16/05, Martin Renschler [EMAIL PROTECTED] wrote:
Hi,
I have a Panasonic
Here in Sweden when I make a call through the regular POTS, I get an polarity
reversal when the callee has lift his phone and answered.
Now I've got an Adit 600 with 40 FXS channels and want to emulate an regular
POTS. But the Adit doesn't seem to support polarity reversal.
Is there other
Jesse Guardiani wrote:
Thank you for you time to help setting up fax.
I still have some questions.
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten =
During the zaptel configuration at the end of it there is this error:
post-install tor2 /sbin/ztcfg
post-install wcusb /sbin/ztcfg
post-install wcfxo /sbin/ztcfg
post-install ztdynamic /sbin/ztcfg
post-install ztd-eth /sbin/ztcfg
post-install wct1xxp /sbin/ztcfg
post-install wct4xxp /sbin/ztcfg
On April 18, 2005 10:17 am, Trent Tuggle wrote:
Opened pseudo zap interface, measuring accuracy...
--- Results after 109 passes ---
Best: 100.00 -- Worst: 99.987793
What exactly does zttest test?
That's not terribly bad; Were you able to tell if the buzz occurrs when the
timing drops
Hello,
I have spend a long time trying to figure out exactly what is the problem
with one of my Asterisk servers, it is the only one at any of our locations
that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of
the rest of our Asterisk servers run identical hardware except
Rich Adamson [EMAIL PROTECTED] writes:
As only one individual, I thought their statements were very straight-
forward and clear. Having worked as a senior manager in a technical
organization, a large number of tehcnical people simply do not
comprehend some words (or read other words into
Hi Matteo,
Please find attached excerpts of the error below:
supertone.c:337: invalid type argument of `-'
supertone.c:337: syntax error before xmlChar
supertone.c: At top level:
supertone.c:344: redefinition of `cur'
supertone.c:263: `cur' previously defined here
supertone.c:344: invalid type
I've heard this problem could be caused by the hold music. I forgot the
name of the process mpeg or wavmpeg, something along those lines...
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
Are you running any AGI scripts?
On 4/18/05, Moody [EMAIL PROTECTED] wrote:
Hey Everyone,
I've been running a version of the CVS without issue until late last
week when suddenly Asterisk would randomly hit 99% CPU and stop
registering my DIDs.
If I stop Asterisk with a 'stop now' and
try auto-apt for getting dependencies satisfied on the fly while compiling.
Manuel Casal wrote:
During the zaptel configuration at the end of it there is this error:
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Check the telco equipment you are plugging into (PBXes) with the
crossovers.. Unless they are all on the same power grid and protected
I would blame them. my two cents...
On 4/18/05, mattf [EMAIL PROTECTED] wrote:
Hello,
I have spend a long time trying to figure out exactly what is the
Thanks Ariel.
Your 2nd suggestions seems a good bypass for this problem... it might be helpful here, thanks!
About the 1st one (using paid X-Ten software), I am using paid X-Pro, which does have a transfer button... but ifIuse this button instead of pound, the calls simply hangs up..
But I think
As only one individual, I thought their statements were very straight-
forward and clear. Having worked as a senior manager in a technical
organization, a large number of tehcnical people simply do not
comprehend some words (or read other words into whatever they happen
to be reading),
For this particular server all telco equipment is in a climate controlled
room kept at 66 degrees F and they are all on APC SmartUPS rackmount power
battery backups, Also all of these connections had previously been connected
to other Digium cards in the last year with no issues.
MATT---
thanks for the help... I knew I missed some info...
Music on hold.. I am not using any form of it.
As for AGIs... I do have AreskiCC installed but it is used for only
some calls. I discounted it as being the culprit as the problem seems
to occur even when no one is connected and for sure when
Hi Ariel,
Thinking a little bit more about your idea of parking calls for 'simulating' a consultive transfer, I realized the following problem:
If an agent is making an outgoing call (or even receiving a call that is not coming from the queue), he is not considered busy to the queue manager.]
That
I have dealt with livevoip on several issues as the new account I just set
up had a number of problems, unlike the first one I purchased. They were
responsive, offered fixes within an hour, fixed their problems within the
day, and I have had no problems or rude responses with them. I would tend to
I have a Snom-190 that I've successfully used on a * box with the LED's
lighting up when a line goes active.
I have moved it to another box, though, and I'm having trouble with it.
It almost seems as though there is a limit to how long a sip channel name can
be for the subscribe/notify to work
Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and another
to complete the remote call)??
Thanks
Hello wonderful asterisk users list.
I have some energy traders that are currently using 2 wire
hoot-n-hollers (squawk box, always open direct line) to different
trading floors throughout the country. Each box has one hoot-n-holler
line. I would like to make these boxes IP based by connecting
On Mon, Apr 18, 2005 at 11:34:03AM -0400, Michael George wrote:
I have a Snom-190 that I've successfully used on a * box with the LED's
lighting up when a line goes active.
I have moved it to another box, though, and I'm having trouble with it.
It almost seems as though there is a limit to
Hi all,
Maybe someone encountered similar issue.
I have an * with the incoming DID over SIP. * is behind a firewall. I
have no issues with other SIP devices connected from the outside
network, however on that DID when I receive a call I can hear only
incoming audio, no outgoing. If I setup a
On Apr 17, 2005, at 12:23 AM, Lance Grover wrote:
I have rebooted the phone and restarted asterisk after each change.
Did you do it in that order? If so, that is probably a source of
trouble (you should restart or reload asterisk before the phone boots,
not after).
--
Joshua P. Dady
Title: Asterix Manager Proxy in Java/EJB?
Anyone doing/done a manager proxy to Asterisk in Java?
Looking to avoid the Python/PERL/etc. managers (not that theres anything wrong with them or the languages) but were running a Java environment already and Id like to not re-invent the wheel if
What is it you're trying to accomplish?
Squawk Box--fxo---*--IAX2/SIP clients?
or
replace the 2 wire solution between the different locations with IAX2/SIP?
The only thing I'd caution you about hear is as you're going back and
forth between 2 wire and IAX2/SIP
I recently installed [EMAIL PROTECTED] and got one of the
TDM400p cards configured to connect to my POTS line. I can make outgoing calls
with no problem however I seem to have a short delay followed by 5 beeps before
the line starts ringing out. Does anyone know what would cause this ?
Hi all,
i'm getting about 1-3 NOTICE[1915]: rtp.c:453 ast_rtp_read: RTP:
Received packet with bad UDP checksum message per call on CVS HEAD
from 31 Mar. which seems some changes regarding rtpchecksums is made
at that time.
setting rtpchecksums to no or yes in rtp.conf doesn't make any sense.
now
Thomas:
It sounds like you may need to unlock your phone.
If I recall, you can hit **# to unlock it; then go to the settings menu.
On newer firmwares, you'll have fun trying to get past an actual password.
Check http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
as well as this list's
Title: Asterix Manager Proxy in Java/EJB?
Ok, I just answered my own question, for
the edification of the group:
http://www.voip-info.org/wiki-Asterisk-java
Colin Stefani
Tideworks Technology
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Stefani
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Huddleston,
Robert
Sent: Monday, April 18, 2005 10:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Calling Card
Anyone experimented with Calling Card support in
Hello wonderful asterisk users list.
I have some energy traders that are currently using 2 wire
hoot-n-hollers (squawk box, always open direct line) to different
trading floors throughout the country. Each box has one hoot-n-holler
line. I would like to make these boxes IP based by
Huddleston, Robert wrote:
Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and another
to complete the remote call)??
It depends wether you
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was hoping to hear if it helped you to change that?
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en
With auto-apt the problem is not solved...
Thanks
Andres Paglayan escribió:
try auto-apt for getting dependencies satisfied on the fly while
compiling.
Manuel Casal wrote:
During the zaptel configuration at the end of it there is this error:
___
Yes, if I understand what you are asking.
1. The Card User calls to your asterisk PBX.
2. Asterisk answers the call on line 1.
3. Asterisk places an outgoing call on line 2 bridging the lines.
(That is how it works in the SIP world.)
So you would need an FXO/FSO pair of lines to let them make a
Hi, I did not find any useful information to configure a Wildcard
TDM400P with a FXO card. I've tried everithing, I tried configure it
using the cvs and the information from digium page, I tried to
configure it using
debian packages, I tried to configure with kernels 2.4.30 and 2.6.11, I even
Where is this line in zapata.conf under the [channels] context?
channel=1
Also is that line in zaptel.conf correct? Here is mine Note the
lack of and underscore on fxsks...
fxsks=1
loadzone = us
defaultzone=us
Try these settings and the run ztcfg -vvv
Restart * and see what you get
I've been running a version of the CVS without issue until
late last week when suddenly Asterisk would randomly hit
99% CPU and stop registering my DIDs.
Similar things happened to me with the CVS version from around that
time. Randomly every 2-3 days asterisk would use 99% CPU and just sit
disallow=all
allow=g726
allow=g729
Change to this and try again:
disallow=all
allow=ulaw
Broadvoice officially only supports ulaw. g726 works some times on
some numbers, but don't rely on that. You can also drop the callerid=
since Broadvoice will not use it anyway.
--Luki
Good suggestion. It now seems to roam between access points nicely, even
while a call is in progress.
What access pooints are you using?
-rb
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
As a followup for any who has the same problem, and searches the
archives (don't you love finding the problem you have in the archive,
but no-one followed it up?), check the following references:
http://lists.digium.com/pipermail/asterisk-dev/2005-April/011291.html
and the status of the updated
Manuel,
This is from my Wiki page on running Asterisk on Debian/GNU Linux.
Build and Install Zaptel
Zaptel provides support for Digium hardware. The following steps can be
followed to build and install Zaptel.
1. Create symbolic links to the new kernel's source files by issuing the
following
Inline...
Hi, I did not find any useful information to configure a Wildcard
TDM400P with a FXO card. I've tried everithing, I tried configure it
using the cvs and the information from digium page, I tried to
configure it using
debian packages, I tried to configure with kernels 2.4.30 and
Hi everyone,
I'm struggling to get four E1 primary rate ISDN lines working in a *
server with a TE405p.
So far almost so good...
My configuration files are below but my problem seems to be that only 30
B-channels are being seen by asterisk - when I start * with -vvvgc I get
the following as
Title: Voicemail not working...
Hello All,
My voicemail seems to have stopped working and I cannot figure out why.
After call times out, the user receives a message the no one is available to take the call.
The CLI shows this...
-- Got SIP response 603 Decline back from 192.168.1.248
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
As far as the issue with DC voltage on the POTS line only
being 43.8 DC, my guess was that is just an issue with
voltage drop on the line because of distance between me
and the CO.
No possible way. If
No changes were made to chan_sip when the iax2 jitter buffer was added.
However, ive seen and hear several complaints about coredumps,
deadlocks in cvs-head chan_sip recently.
/Z
Luki wrote:
I've been running a version of the CVS without issue until
late last week when suddenly Asterisk would
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was hoping to hear if it helped you to change that?
Let's try it together:
1. Open IPswitch
2.
Thanks,
Derek
My /etc/zaptel.conf :
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124
dchan=16
My /etc/asterisk/zapata.conf:
[trunkgroups]
trunkgroup = 1,16
spanmap = 1,1,0
spanmap = 2,1,1
Hello wonderful asterisk users list.
I have some energy traders that are currently using 2 wire
hoot-n-hollers (squawk box, always open direct line) to different
trading floors throughout the country. Each box has one
hoot-n-holler
line. I would like to make these boxes IP based by connecting
Hmmm,
Hoot and Holler, hoot-n-holler, ARD: Automatic Ring Down, Hot
line and Private Line Automated Ringdown (PLAR)
You should think about VoIP via Asterisk.
Here is a quick search result on it
http://lists.digium.com/pipermail/asterisk-users/2003-March/008936.html
But not much
Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and another
to complete the remote call)??
If you use IAX2 termination for incoming and
1 - 100 of 203 matches
Mail list logo
