Does anybody know what these are? Started getting them last night when
I upgraded from 1.2.6 (Zaptel 1.2.6) to 1.2.10 (Zaptel 1.2.7). Then my
E1 ISDN PRIs go down...
I've had to roll back to 1.2.6 :-(
Matt.
___
--Bandwidth and Colocation provided b
Hi,
I hava a ZAP device (a premicell), and it sends polarityswitches when the call
starts and when the
call ends.
in zapata.conf with
answeronpolarityswitch=yes
then when the phone starts to ring, you dont hear it ring, only when the
person answers the phone
do you start to hear him talk
Searche the entire lise (as of 14 months ago) wiki
etc. and I cant seem to find the following.
I havd 3 Snom phones. There is a queue. 1 member is
a regular member (member => SIP/123) and the rest are used with local (Member
=> Local/[EMAIL PROTECTED]. Ext. 456 is
Exten => s,1Dial(SIP/999&SI
Is unicall actually in use in Australia?
PaulH
On Thu, 2006-07-20 at 13:17 +1000, MBIT Technologies wrote:
> Hi
>
>
>
> Has anyone set this up and what protocol variant did you use for
> unicall? There is no Australian setting that I know of so any help
> would be greatly appreciated.
>
>
Hi,
I am using realtime function to query the table.
Table sipprop
name,number,forward
Peter,1234,0
May,4321,1
RealTime(sipprop|name|Peter) and it can simple get the output of
number if the record exist.
If I issue RealTime(sipprop|name|John), asterisk will show "No
Realtime Matches Found". W
On Thu, 2006-07-20 at 09:18 +0800, unplug wrote:
> Do you mean I can set the header first and use the same command to
> reset the header afterward without adding 2 same header field? Does
> there is only one remote-pary-id header in the sip message finally in
> the example below?
> e.g.
> exten =>
Daniel Oakes wrote:
Hi All,
I have a problem with conferencing, but it's more to do with the g729
codec. I have purchased six licenses for g729 for all our phones, and
occasionally want to do conferencing, but at the moment it only allows
two people in before the licenses run out.
When two
Hi
Has anyone set this up and what protocol variant did you use
for unicall? There is no Australian setting that I know of so any help would be
greatly appreciated.
Regards
Mark Brooker
___
--Bandwidth and Colocation provid
Hi,
We have a few PSTN Lines connected to our Asterisk server (via SPA-3000 FXO
interfaces), and everything works great except for 1 major annoyance.
When PSTN callers get to a user's voicemail and they just hangup, a few (4)
PSTN disconnect tones are recorded before the connection breaks off.
S
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
> On Tue, Jul 18, 2006 at 10:13:58AM +1200, Matt Riddell (NZ) wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> From: http://www.sineapps.com/news.php?rssid=1377
>>
>> ISS Xforce has published details of two security
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Lacy Moore - Aspendora wrote:
> Can Digium legally sell this? It is my understanding that if the
> license of
> all parts cannot be legally sold, then there is no way it is going to be
> included.
Well yes, if the licence is BSD like and all code is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Zeeshan Zakaria wrote:
> I installed the packages listed on this webpage. But What is meant here by
> Server and Clients. Do I have to write these programs and then run then
> somehow, one on server and one on some client. From server I understand
> As
I always like to activate the syslog and debug on my SPA's. Sometimes this
will tell you what they are doing.
Shanon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, July 19, 2006 8:30 PM
To: Asterisk Users-List
Subject: [
The Metermaid patch is the closest you can get for now.
It seems to work ok on v1.2.9 but I haven't tried it on v1.2.10
http://bugs.digium.com/view.php?id=5779
> -Original Message-
> From: Douglas Garstang [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 18, 2006 12:58 PM
> To: Asterisk U
Need a little help trying to understand what's happening here.
spa941 -> Asterisk-A -> iax2 -> Asterisk-B -> spa942
When the spa941 (x3000) calls spa942 (x1004), the spa942 returns a "busy
here" sip message. The spa942 is not busy and does not have DND or any
other option set to cause a busy-h
Hi,
I looked into my box today via SSH and there was an asterisk process
consuming all available CPU.
I am running 1.2.10 from tar ball on Mac OSX 10.46 (G3/400).
There was no seeming issue with the performance of asterisk ie
everything was working (that I could hear). I noticed there was
Do you mean I can set the header first and use the same command to
reset the header afterward without adding 2 same header field? Does
there is only one remote-pary-id header in the sip message finally in
the example below?
e.g.
exten => 1234,1,Set(SIP_HEADER(Remote-Party-ID)[EMAIL PROTECTED])
..
I think you're being a little too subtle. I recommend a clue x 4.
Nick
On Wed, Jul 19, 2006 at 08:28:15PM -0400, C F wrote:
> IDIOTS
> STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
> STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
> STOP STOP S
You using the t or T options in the dial app?
On 7/19/06, Delca <[EMAIL PROTECTED]> wrote:
Hi, Now that i fixed the problem with roundrobin, now i can't get
Blind Transfer to work. I already tried to modify blindxfer option in
features.conf with almost any number and still doesn't work. When i
d
IDIOTS
STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP STOP
STOP STOP S
I have seen a very small number of posts on this type of
setup;
1. mysql replicated failover cluster (Linux HA) for the
realtime databases and ODBC voicemail storage
2. multiple asterisk servers (~4) connected to the SAME
realtime tables and VM store.
3. Any defined SIP client connect
On Wed, 2006-07-19 at 10:29 -0500, Moises Silva wrote:
> Carlos. Unblocking the remote side is NOT your responsibility, unless
> you own the 2 end points :). I suppose you are getting connected to
> some telco (avantel, telmex, etc), if so, is telco's responsibility to
> unlock their side.
>
et("IAX2/isphone-2", "FROMCONTEXT=exten-vm") in new stack
-- Executing Macro("IAX2/isphone-2", "record-enable|200|IN") in new
stack
-- Executing GotoIf("IAX2/isphone-2", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s
et("IAX2/isphone-2", "FROMCONTEXT=exten-vm") in new stack
-- Executing Macro("IAX2/isphone-2", "record-enable|200|IN") in new
stack
-- Executing GotoIf("IAX2/isphone-2", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s
> thanks a lot for answering! This solves my problem perfectly.
>
> ciao,
> morel
Glad to be of help! Back when I first started on Asterisk, the person
who told me about the wait() command didn't say anything about answer()
first, so I figured I'd make sure you got it :)
You can also revert to an older version of chan_sccp - try using 20060204 -
the segfaults stopped when we reverted to that version.
-Scott
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Doug Lytle
Sent: Wednesday, July 19, 2006 7:27 AM
To: Asterisk Users Ma
Hello,
Well I was having
transfer issues in 1.2.9.1 so I downgraded to 1.2.7.1. For testing I
installed 1.2.10 on a test server and setup two Polycom SIP phones. Tried
the transfer on this configuration and had the same issues. Here is a log
from the console:
This is how the flow
goes:
heh. Just got permission to drop 1.2.10 on there and it seems to be
working the way i want it now.
--Wayne
On Jul 19, 2006, at 3:40 PM, Wayne P. HIll wrote:
I'm working on a server being implemented for a client right now
which, due to a long string of issues I won't go into, has decided
Do you have a contact information for Mike Workman with Virtell
networks as he ripped me of for $5,790 (Took the money and never sent
product)
714-279-0204 ext 102 mark
Thanks!MarkVoice
InternationalAnaheim, Ca 92868714-279-0204 Ext 102WEDSITES
www.voiceinternational.com www.digiumboa
Steven,
I’ve been searching that you say, but certainly
I don’t know where to search or those lines isn’t there.
I found these:
Configuring VoIP DigitMap
dialing pattern
- empty -
Configure FXS Setting Parameters
Ringing Timeout = 180 seco
Hi all,
I'm trying to connect to a H323 gatekeeper (an old Swyx) with the oh323
driver (had stability problems with the h323 driver).
However I cannot get asterisk to register to the gatekeeper with a
normal user account. Trying the same account in
Ekiga/gnomemeeting/ohphone the registratio
I have a customer on one of my Asterisk boxes that wants a small number
of DIDs in Hong Kong. Referring to voip-info.org, we found the provider
HKBN and their 2b service at www.2b.com.hk.
Following the information at
http://www.voip-info.org/wiki/index.php?page=asterisk+settings+HKBN+2b
we success
I have added a header using the SipAddHeader command. At some point
later, I would like to clear this header, as I no longer need it.
For instance, I add the Call-Info header for auto-answering a SNOM
phone. When I transfer the call to another snom phone, the auto-answer
header travels along w
Will this card serve as a zaptel timer?
It's kind of hard to imagine that it wouldn't, I just want to make
sure since this is the first Digium card with no telco interfaces.
Thanks,
MATT---
On 7/19/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
- Steven <[EMAIL PROTECTED]> wrote:
> Can a
- Steven <[EMAIL PROTECTED]> wrote:
> Can a SIP call use the cards transcoder or must the call go through
> the card for it to be used? (like the echo can. does)
This is not a line interface card, so calls don't go 'through' it. It is a
resource that provides transcoding channels, that is all
Tzafrir Cohen wrote:
On Wed, Jul 19, 2006 at 09:30:30PM +0300, Cosmin Prund wrote:
I don't use Debian but I'm going to give Debian a try. I'm downloading
the Debian distribution right now. Unfortunatelly it will be about a
week till I'll download the whole 8Gb (2xDVD iso).
Im not a PRI guru, just configured once one of our asterisk servers in
loop using PRI for testing purposes (we were getting problems with
MFCR2). But for my experience i can tell you that OK in zttool (no
alarms) does not means that everything should work. Only the physical
link is ok, but the cal
IAX2 does not need such a thing, since always send the DTMF out of band.
On 7/19/06, Peter Beckman <[EMAIL PROTECTED]> wrote:
I know you can use dmtfmode in sip.conf, but does it do anything in an
iax.conf context?
ie.
iax.conf:
...
[super]
auth=md5
type=friend
username=super
secret=man
host=i
I'm working on a server being implemented for a client right now
which, due to a long string of issues I won't go into, has decided
that they wish to use cisco 7960s over sccp with asterisk. Now it's
up to us to write in the many features that this setup doesn't
support by default. The cu
Hi all,
We have been having stability issues with one of our servers, a Dell
2850. The machine has been locking up a few times a week for the past
several months, except for a two month period when we were running on
a Sangoma card (we have since moved back to Digium hardware). The
machine was
I'm doing something similar, using a php script to access a postgres database. I'ts quite easy using the phpagi class.Take a look at: http://phpagi.sourceforge.net/
good luck,Mauricio MantillaOn 7/19/06, Stefan Reuter <[EMAIL PROTECTED]> wrote:
Camilo Echeverry wrote:> 1- receive the call (obvious)
On 7/19/06, Matthew Warren <[EMAIL PROTECTED]> wrote:
We build custom scripts for Asterisk. We can build this for you, for
reletivly inexpensive. But you will need to contact me thru email at
mwarren "at" procomconsulting "dot" com .. This is a commercial app you
need but requesting on a non
I would agree with you on just about everything.
Except the op had his fax connected via channel bank directly to *
and a pri on the other port - ie no packets involved here.
However - all faxing does involve the transfer of frames from one fax
to the other and that is was ecm handles. But
I know you can use dmtfmode in sip.conf, but does it do anything in an
iax.conf context?
ie.
iax.conf:
...
[super]
auth=md5
type=friend
username=super
secret=man
host=iax.super.com
context=inbound
dtmfmode=inband
...
Beckman
-
Jerry Jones wrote:
without ecm -
line errors will cause slight imperfections (dots) on transmitted
image
with ecm -
retry, retry, retry, fail
In the ECM retransmissions only the frames that were not received
properly are resent. Once the data to send is assembled by the sender
He wasn't asking for someone to do it for him...he was asking if it was
capable of doing it...
- Original Message -
From: "Matthew Warren" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, July 19, 2006 2:50 PM
Subject: [asterisk-users] re:Simple But important question (for me)
We build cus
On Wed, Jul 19, 2006 at 09:30:30PM +0300, Cosmin Prund wrote:
> I don't use Debian but I'm going to give Debian a try. I'm downloading
> the Debian distribution right now. Unfortunatelly it will be about a
> week till I'll download the whole 8Gb (2xDVD iso).
Unless you have a bad internet connec
We build custom scripts for Asterisk. We can build this for you, for
reletivly inexpensive. But you will need to contact me thru email at
mwarren "at" procomconsulting "dot" com .. This is a commercial app you
need but requesting on a non commercial group.
Matthew Warren
_
Salve *!
I got it¹ ;)
Robert Michel schrieb am Mittwoch, den 19. Juli 2006 um 17:27h:
> What is the trick to let the caller hear 10 seconds free-ringing sound
> and then the busy signalisation of his telco without costs for him?
>
> exten => sip1/Unknown,1,Wait(10)
> exten => sip1/Unknown,2,Hang
Maxim Vexler wrote:
Check the paragraph on ECM at http://www.voip-info.org/wiki-Asterisk+fax
Everything that you read on a wiki must be considered potentially bogus
or otherwise misinformed.
Let me rewrite that paragraph for you...
ECM - er
I'm using a HFC-S ISDN card and a TDM400P zaptel card with 3xFXO and
1xFXS modules.
How did you finally manage to compile bristuff on Centos 4? I'm
downloading Debian right now but the 2 DVD images will take about a week
to download so I'm willing to try anything else in the time. I've got
both
I don't use Debian but I'm going to give Debian a try. I'm downloading
the Debian distribution right now. Unfortunatelly it will be about a
week till I'll download the whole 8Gb (2xDVD iso).
Tzafrir Cohen wrote:
On Wed, Jul 19, 2006 at 03:17:15PM +0200, Filip Drągowski wrote:
Firs
Chris, This is exactly what I found. The script does pipe the sound OK, but it's obviously in a format that Asterisk doesn't like. Arecord can turn that audio into just about anything, so if there's someone on the list here that could tell us format Asterisk is expecting, we can probably figure thi
Hi all,
I have configured a connection my sip voip provider. I can make outbound
call without trouble. But I cannot recieve voip calls. The sip
negociation seams to start well but at some point during the rtcp
dialog, things seems to block.
As you can see on the above log sample, I recieve s
Hi,
I've followed the instructions on the Wiki for pulling music-on-hold
from my sound card's line input. It doesn't work, however. MoH
starts and immediately stops. Apparently, I'm not the only person
having this problem. I'm thinking that maybe arecord(1) is not
sending the right kind of au
- Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> The first include should just set some variables.
> I tried to number this extension block either with _. or with s
> and since it matches, the function (setting some variables) have been
> done.
> After that, I want to go to the next include,
On Jul 19, 2006, at 2:15 AM, Chris Mason (Lists) wrote:
Martin Joseph wrote:
I found a terminator called sellvoip.net, whose website is crap
(currently), but whose route from my server is very clean and short.
My calls all sound perfect now. I keep teliax and nufone
configured as backu
without ecm -
line errors will cause slight imperfections (dots) on transmitted image
with ecm -
retry, retry, retry, fail
On Jul 19, 2006, at 12:23 PM, Maxim Vexler wrote:
On 7/19/06, Lee Howard <[EMAIL PROTECTED]> wrote:
Jerry Jones wrote:
> Also if possible turn off ECM o
Hi, Now that i fixed the problem with roundrobin, now i can't get
Blind Transfer to work. I already tried to modify blindxfer option in
features.conf with almost any number and still doesn't work. When i
dial an extension. I pick up the phone, and then i press # to transfer
the call and nothing ha
didn't test it , but i think it will be /sounds/se/digits , setting
Language to se will point to /sounds/se and then asterisk will keep
the same logic as per default is sounds directory.
This is a guess, please test it.
On 7/19/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Okay, thanks! I al
Hello Moises. I enabled debug mode in asterisk. When I dial, I get:
Jul 19 14:24:38 DEBUG[4463]: build_route: Contact hop:
Jul 19 14:24:38 VERBOSE[4463]: -- Executing
Dial("SIP/192.168.0.6-08137090", "Zap/1/4502") in new stack
Jul 19 14:24:38 NOTICE[4463]: Unable to create channel of type
I have a problem here, when an ACD agent is stuck in PAUSED mode.
As you can see from the outout of 'show queues' below, the agent 80014133 has a
status of paused.
Why is there a 'not in use' after the paused?
hestia*CLI> show queues
oe_techsupp has 0 calls (max unlimited) in 'rrmemory' strategy
On 7/19/06, Lee Howard <[EMAIL PROTECTED]> wrote:
Jerry Jones wrote:
> Also if possible turn off ECM on the FAX machines
This is unsound advice. Why do you think this could possily help?
Lee.
___
--Bandwidth and Colocation provided by Easynews.com
Use festival text to speech for saying the
address
- Original Message -
From:
Camilo
Echeverry
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 19, 2006 12:49
PM
Subject: [asterisk-users] Simple But
important question (for me)
Hi.I'm 100% newbie (
On Wed, Jul 19, 2006 at 07:21:20PM +0300, Maxim Vexler wrote:
> On 7/19/06, Maxim Vexler <[EMAIL PROTECTED]> wrote:
> >On 7/18/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>
> >wrote:
> >> I think when a PSTN line says 'Ring' it's simply for aesthetics... The
> >> line is 'answered' the in
> "Tzafrir" == Tzafrir Cohen <[EMAIL PROTECTED]> writes:
Tzafrir>
Tzafrir> On Wed, Jul 19, 2006 at 10:31:05AM +, Tony Mountifield wrote:
>> In article <[EMAIL PROTECTED]>, Russ Price <[EMAIL PROTECTED]> wrote:
Tzafrir>
>> > The problem is that this will have to be done with each new kerne
On 7/19/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
Hi
I figure I'm technically someone who has first-hand experience with the
Astribanks :-)
On Tue, Jul 18, 2006 at 10:22:07PM -0400, C F wrote:
> Well tzafrir will know :)
> Tzafrir here are a few questions:
> 1. Does the FXO module support: A
The short answer is yes
On 7/19/06, Camilo Echeverry <[EMAIL PROTECTED]> wrote:
Hi.
I'm 100% newbie (in asterisk)
I need to know if i can use astersik for something like this:
1- receive the call (obvious)
2- get the Caller ID
3- Send the CID to another application and get some info from a Dat
Camilo Echeverry wrote:
> 1- receive the call (obvious)
> 2- get the Caller ID
> 3- Send the CID to another application and get some info from a Database
> example: Your address is "some address"
> 4- Get that info and convert it into voice (by mixing various audio files)
> 5- return it to the Call
Hi GroupDespite of the limits on Asterisk and PacketCable, I've found this web site where I found patches to * to work with packetcable NCS.http://asterisk.urtho.net/tiki-index.php
I know this is a halted project but it give me some hope to make some research and to make work the eMTA from Motorola
Jerry Jones wrote:
Also if possible turn off ECM on the FAX machines
This is unsound advice. Why do you think this could possily help?
Lee.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or u
Hi.I'm 100% newbie (in asterisk)I need to know if i can use astersik for something like this:1- receive the call (obvious)2- get the Caller ID3- Send the CID to another application and get some info from a Database example: Your address is "some address"
4- Get that info and convert it into voice (
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam
Sent: 19 July 2006 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] call forwarding to mobile phone
You will need an asterisk server + X1
Are you sure the other end is configured properly?
What does "zttool" says?
Have you turned on all the asterisk debug messages to look further?
Regards
On 7/19/06, Lincoln Zuljewic Silva <[EMAIL PROTECTED]> wrote:
Hello all. I have a Digitum TE110P board configured and working (I think
that i
Just a couple checks...
You are using G711u for the FXS - right?
Also if possible turn off ECM on the FAX machines
Otherwise I have never used Sangoma cars but this configuration works
very well with Digium cards, at least with asterisk, I do not use aah
On Jul 19, 2006, at 11:11 AM, Bruce
Okay, thanks! I already have set language to 'se' in indications.conf.
Next question. If asterisk where to play a digit - does it look in
/sounds/se/digits or /sounds/digits/se ?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta
Sk
On 7/19/06, Maxim Vexler <[EMAIL PROTECTED]> wrote:
On 7/18/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> I think when a PSTN line says 'Ring' it's simply for aesthetics... The
> line is 'answered' the instant * connects to it for two-way audio...
> (well not that instant but so
Greetings all,
I'm running Asterisk 1.2.9.1 installed from /usr/ports/net/asterisk on
FreeBSD 6.1-RELEASE.
I'm experiencing a guaranteed asterisk core dump with any Sipura device
set to forward all calls to an extension that is mapped to a queue:
-- Executing Macro("SIP/10040-4c43", "ca
Hello all. I have a Digitum TE110P board configured and working (I
think that it's working). When I configure in extensions.conf to a
extension route to that board I get "Unable to create channel of type
'Zap'" on log.
Here are some configuration:
lspci -vv
:01:07.0 Network controller: Ti
On Wed, 2006-07-19 at 18:10 +1000, RR wrote:
> Think this has been covered several times on the list. Sounds like the
> spinlock.h issue. You need to go into the kernel directory, for you it
> seems like the
> /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/linux/spinlock.h
> file and replace an
Location of the sound files
Asterisk normally looks for a sound file with an extension used for
the codec used. If a language is set for the channel with the
SetLanguage() application, Asterisk first looks for
countrycode/filename where countrycode is the language code (example:.
'fr' for french).
I had similar problems with a Sangoma card in this configuration. I recently recieved from Sangoma an updated driver that fixed issues with resyncing the clock on the card. You might try getting a hold of Sangoma, David Yat Sin if possible and ask him about it, it may very well be the same problew.
On Wed, 2006-07-19 at 16:27 +0800, unplug wrote:
> I have a header field in the sip message: Remote-Party-ID:
> [EMAIL PROTECTED]
> I want to replace it to :Remote-Party-ID: [EMAIL PROTECTED]
In Asterisk 1.2 and the trunk, the SipAddHeader and SipGetHeader
applications are deprecated. There is a
Hi,
I am wondering how I can change the language of queue hold position.
This is probably pretty simple (yes I know I have to record my own
soundfiles). What I don't get is where to set the "numbers"?
In queues.conf there are settings for:
queue-youarenext = queue-youarenext
queue-th
Dean,
Thank you for your help. I have it up and running. As soon as I get some
free time lets chat about what we need going forward. I have some
dollars to move this forward. If I can accommodate additional
requirements, all the better.
Michael
-Original Message-
From: [EMAIL PROTECTED]
I have AAH2.8 on a dual Xeon system with a
Sangoma A104 and an Adtran Channel Bank.
The system has a single PRI connected to port 1 and port 2 has the T1
cable connected to the Channel Bank. Both are configured properly and
work for the inbound/outbound calls and soft-fax reception.
I have
If he modifies the local dialplan on the SIP device, the 3-way issue should go
away because he will no longer need to dial a #.
Pablo, Look for something like
(0T|011x.T|101x.T|x.#|9x.T|*x.T|#xx|393*1x.T|8|5xxx|x11) in the SIP devices
config.
You would want to change the 9x.T (which means
From where did you downloaded the snapshot? could you post a link to
the sources?
I think this is a problem of missmatch version of old libunicall an
newer libmfcr.
Those undefined macros should be part of the libunicall headers, so
when compiling the new libmfcr2, it does not find the newer lib
Hi - I am using Asterisk Home 2.7 on a dedicated Linux server to make out going calls. For the most part everything works fine except that most of the voice calls on playback are jittery/choppy at the beginning of the call. After a couple of seconds of choppyness the rest of the message plays b
Some corrections:
> 3. What are the configurations available for the asteribank 32?
4 FXS (32 FXS ports)
3 FXS + 1 FXO (24 FXS ports, 8 FXO ports)
2 FXS + 2 FXO (16 FXS ports, 16 FXO ports)
1 FXS + 3 FXO (8 FXS ports, 24 FXO ports)
> 4. Pricing?
Contact [EMAIL PROTECTED]
--
Tzafrir Cohen
Hi people,
When a I upgrade my asterisk 1.2.4 to asterisk 1.2.9.1 or to asterisk
1.2.10, app_queue, after some time up, doesn't work (I think)
When I call to the queue, the channels up:
Zap/1-1 [EMAIL PROTECTED]:4 Up Queue(suporte3600)
but nothing happens, the a
Hi
I figure I'm technically someone who has first-hand experience with the
Astribanks :-)
On Tue, Jul 18, 2006 at 10:22:07PM -0400, C F wrote:
> Well tzafrir will know :)
> Tzafrir here are a few questions:
> 1. Does the FXO module support: A. Hangup detection?
What type of hangup-detection do
Carlos. Unblocking the remote side is NOT your responsibility, unless
you own the 2 end points :). I suppose you are getting connected to
some telco (avantel, telmex, etc), if so, is telco's responsibility to
unlock their side.
To discard any problem with Asterisk, try using "testcall" utility
in
Salve *!
What is the trick to let the caller hear 10 seconds free-ringing sound
and then the busy signalisation of his telco without costs for him?
exten => sip1/Unknown,1,Wait(10)
exten => sip1/Unknown,2,Hangup
Will not create a free-ringing, but:
exten => sip1/Unknown,1,Dial(SIP/hardwarephone
As we are talking about pbx boxes (for large office/enterprises/ maybe ITSP?)
i think we are using server grade boxes (like hp ml3xx or bigger)
I have some servers with fan on cpu heatsink, but most of them are using
only heatsink on cpu, and redundant fans.
I think, we need some real life compar
i don't think there is ANY difference with 1 or 2 SATA HDD.
however here is my single proc Xeon2.8 (512k)
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 2 2
No. In SIP these features are configured on the SIP device. If you
cannot disable three-way calling, or modify the dialplan on your SIP
device, then there is nothing you can do to fix the problem.
Pablo Mora wrote:
I really don't understand what you say.
I've been searching in my SIP de
There is an updating.txt file in the web app under WEB-INF/README
In practice, updating from 1.2.0 means changing the webapp and keeping the
same database. It's really a 5 minutes operation.
If you istead used yum to install, just type
yum update queuemetrics
and you should have it updated in
I agree with Eric, that it must be the local
dialplan on the SIP device.
-- -- Steven
http://www.glimasoutheast.org
"Pablo Mora" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
Really
dont.
Dialplan is very simple, please
take a look
[incomin
Interesting I wasn't able to find answer to any of the questions in 1.
Only some confusing answer to 2, no answer whatsoever for 3, the
website just says that it could be mixed. No pricing on the site
either.
On 7/19/06, Michael Graves <[EMAIL PROTECTED]> wrote:
On Tue, 18 Jul 2006 22:22:07 -040
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