It works very fine for me.
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Monday, July 25, 2005 11:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Soft Phone
Any suggestions for
Hello
Here you go :
[wengo-outgoing]
type=peer
fromuser= username
username= username
secret=password
host=voip.wengo.fr
fromdomain=voip.wengo.fr
disallow=all
allow=alaw
allow=ulaw
dtmfmode=inband
canreinvite=yes
nat=yes
insecure=very
dtmf=inband
context=wengo-outgoing
authname= username
This is
Hi,
I am configuring Asterisk with TDM400 card with 1 FXS and 1 FXO
module. My first goal is to allow phones to be able to call out
through the asterisk PBX.
After channels and dial plans setup, Zap/1 connect to phone and Zap/4
connect to provider , when I dial the phone, the following message
I now get:-
/usr/src/asterisk/dsp.c:1395: undefined reference to
`ast_dsp_busydetect'
The Make file changes modify BUSYDETECT but if you have
BUSYDETECT+= -DBUSYDETECT_COMPARE_TONE_AND_SILENCE
the above error is shown.
Renabling BUSYDETECT+= -DBUSYDETECT_MARTIN corrects the problem.
--
Dave
wassim darwish wrote:
i ve downloaded
asterisk-oh323-0.6.6.tar.gz
I am getting this and anybody know howto fix this?
#tar zxvf asterisk-oh323-0.6.6.tar.gz
oh323]# cd asterisk-oh323-0.6.6
asterisk-oh323-0.6.6]# ls
asterisk-driver CONFIGURATION Makefile rpm
TESTS
BUGS
Hi everybody,
I have corrected this line in extensions.conf by
stripping spaces off and now it executes:
exten =
22999,1,VoiceMailMain(s${CALLERIDNUM})
when it runs, the mail box number is asked and
password too. I expected no question were made, because I inserted
CALLERIDNUMBER and s in
I was looking through the cards, and could not tell the difference between
TE410P and TDM400P.
TDM400P has the possibility of up to 4 FXS or FXO modules for use with
regular analog telephones (into FXS mods) or regular POTS lines (using
FXO)
Looking at the picture, both of them can support
Just FYI for anyone else who might run into this problem:
After unloading the zaptel and zaprtc modules the audion works
again!
-- Arnd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Try this:
exten = fax,1,Dial(${FAX})
exten = fax,2,Congestion
exten = fax,102,Congestion
Bruno.
Eric Wieling aka ManxPower wrote:
It seems that exten = fax does not work in a macro.
Asterisk detects the fax, since it complains about no fax extension,
but I have an exten = fax in the macro.
Hi,
I am getting Got
SIP response 405 Method not allowed error on CLI. I am
also getting Port restricted Cone NAT
error on my SJ phone.
Please help!
Afzaal
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello all,
I'm trying to set up a call between two users - basically
1) dial user A
2) dial user B
3) connect the two calls legs
I've read the Wiki and looks like theremay be a few things I can try, but wanted to get the list's opinion on the best/easiest way to do this?
Thanks very much for
On Thursday 21 July 2005 17:42, [EMAIL PROTECTED] wrote:
Thanks Adam. This helps some, but I'm still not sure how you mean
for me to acheive 1). I would have to perform a Dial-command no matter
what, so I guess I would have to make an interruption from the
manager API, but I don't
I tried a little more with following extension.conf
;dial out
ignorepat = 9
exten = _9.,1,Answer
exten = _9.,2,Dial(Zap/4/${EXTEN:1})
exten = 100,1,Answer
exten = 100,2,Dial(Zap/4/2197723)
dial 100 works fine, my other phone can receive the call.
but dial 92197723 doesn't work. In theory, they
hello,
i got this error when i run make after downloading
asteirsk from cvs.
gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT
-D_GNU_SOURCE -O6 -Wcast-align
-DSOLARIS
Thanks for the response but still no luck.
I added those two lines just after the [channel] and updated my dial plan
but the result is the same (there is no CallerID):
Asterisk Ready. -- Starting
simple switch on 'Zap/1-1'Jul 26 00:43:34 NOTICE[8867]: chan_zap.c:5367
ss_thread: Got event 2
Hi,
I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c.
I'm using the same extensions.conf but it seems now include instruction
doesn't want to work, here follows an extract:
[inbound_menu]
include = ins_exts
exten = _X.,1,Answer
exten = _X.,2,Wait(1)
exten =
Some further observation. Using the following config, I can dial 8 , reach the
outbound line and get a dial tone without any problem. Then I can dial
any number from that point on.
exten = 8,1,Answer
exten = 8,2,Dial(Zap/4/)
exten = 8,3,Hangup
It looks more to me a timing issue, is Asterisk
When you write code for Asterisk, you are in the middle of a piece of
code and you add debug, log and console messages that you need yourself
to figure out whether the patch works or not. As a user, some of these
messages may be hard to understand, especially since a many of them look
like Ouch,
The only other item I can think of is to play with rxgain in zapata.conf.
Try rxgain=3.0, then rxgain=6.0, and if that seems to impact
receiving callerid, then adjust rxgain to the lowest value where
callerid still works.
Thanks for the response but still no luck.
I
On 7/25/05, Jay Milk [EMAIL PROTECTED] wrote:
I'd say your hardware is out of codecs. Sipura SPA-2000's, for example,
only allow one G729-call at a time because of licensing issues. Allow
GSM as a secondary codec and you should be fine. Yep, it's more
bandwidth, but...
Thanks Jay, that
Mauro Zanin wrote:
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off
and now it executes:
*exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})*
when it runs, the mail box number is asked and password too. I
expected no question were made, because I inserted
asterisk-users
how to config E400P card?
how to config zaptel.conf and zapata.conf?
thanks a lot
dev2003
[EMAIL PROTECTED]
2005-07-26
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Not at present. Thier webite was never great but they seem to be in the
midst of a redesign.
On Tue, 26 Jul 2005 01:40:05 +0200, Marc Storck wrote:
Thanks Michael,
do they have an online ordering system, they don't seem to have a real
website
Regards,
Marc
Michael Graves wrote:
On Sun,
I'm very interested in the redirect feature of Asterisk. So far I haven't
gotten it to work. My scenario is that there is a two party call going on
where I want to send one of those parties somewhere else. In the wiki is only
an example how to send both parties to a meetme room. Is the
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago, there interface looks good, but
there are still not live, I believe they won't be any time soon.
I found sixtel, but order take eternities, they probably won't get my
orders right any soon.
So
On Mon, 25 Jul 2005, Eric Wieling aka ManxPower wrote:
Leon Sun wrote:
Try to use like following
[tnt]
type=friend
context=fromtotnt
dtmfmode=rfc2833 host=XXX.xxx.xxx.xxx
I am using this way.
You do realize that host= only applies for calls from Asterisk to the TNT,
right? You need
The no sound problem is very probably related to an incompatibility
between the zaptel driver and the Teles AVM B1 ISDN card i am using.
(kernel-capi 2.0)
Anyone knows a work-around for this or is there no way to make both
work?
cheers,
Arnd
___
Hi,
I test the video on asterisk with eyebeam. When I use a public IP for
the softphone, the video work. However, when I test eyebeam under nat
the video doesnt work. I use a routeur linksys WRT54G. I try also to
configure my laptop under DMZ for redirect all the traffic IP and the
video
Hi Steve,
I do not think is on the phone, I think it is Asterisk... and it is the
right behaviour. Or, as Daniel sais, press the # key.
I do not think there are other solutions. But I may be wrong, of
course. :-)
Giorgio
Steve Totaro wrote:
Sounds like its on the phone. What type of
if any one can tell how to compile asterisk-oh323 and
what it is dependencies.
Regards;
wassim
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
___
Try to use like following
[tnt]
type=friend
context=fromtotnt
dtmfmode=rfc2833 host=XXX.xxx.xxx.xxx
I am using this way.
You do realize that host= only applies for calls from Asterisk to the TNT,
right? You need permit/deny to match for inbound connections. Your
Asterisk
Hello all,
I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow
something is not quite right with my vMail setup. I would have sworn this was
all working, but maybe I was just dreaming.
Anyway here is what is happening, say I am on extension 200 and I want to
Try this;
nat=yes
qualify=yes
Erdem HAKI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillaume
Sent: Tuesday, July 26, 2005 2:15 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Eyebeam Video+Nat
Hi,
I test the video on asterisk
http://www.digium.com/index.php?menu=whatpcislot
On 7/26/05, Wilson Pickett [EMAIL PROTECTED] wrote:
I was looking through the cards, and could not tell the difference between
TE410P and TDM400P.
TDM400P has the possibility of up to 4 FXS or FXO modules for use with
regular analog
Is it possible, that you mean:
if (!ast_db_get(SCCP, d-id, result, sizeof(result))) {
instead of
if (ast_db_get(SCCP, d-id, result, sizeof(result))) {
in line 222 sccp_utils.c
Quoting Sergio Chersovani [EMAIL PROTECTED]:
http://chan-sccp.berlios.de/
Howard Leadmon wrote:
Hello all,
I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow
something is not quite right with my vMail setup. I would have sworn this was
all working, but maybe I was just dreaming.
Anyway here is what is happening, say I am on
On Tuesday 26 July 2005 00:41, Nate Kapi wrote:
I have a really good high speed connection on my dedicated server, but
I still get jitter even if there is only 1 call going on at a time.
Will tweaking my iax.conf's jitter buffer settings really make any
significant difference? Does anyone have
On Monday 25 July 2005 23:26, [EMAIL PROTECTED] wrote:
Highly recommended to everyone to stay away from this issue
I do not have a name for the company right off hand, but they got sued
really bad when they tried 911 via VOIP and the 911 drop kept occurring in
different areas and someone died!
On Monday 25 July 2005 19:06, Adam Robins wrote:
-- continue fxo operation after the magical 25 days
Could someone please translate this highly technical explanation into
something more meaningful? I already spend far too many hours dealing with
the nuances of Digium hardware.
Could someone
Hi Jose.
What is the packages version that are you using? What MFCR2 variant
are you using, I mean, wich country?
Maybe Steve could help us on it. I told him about this problem.
Keep in touch.
Denis.
On 25 de jul de 2005, at 15:36, Jose Chiantera wrote:
Hi,
I got the same error, when
On Tuesday 26 July 2005 07:17, Giorgio Incantalupo wrote:
Hi Steve,
I do not think is on the phone, I think it is Asterisk... and it is the
right behaviour. Or, as Daniel sais, press the # key.
I do not think there are other solutions. But I may be wrong, of
course. :-)
If hitting # is
There's one built into Asterisk. It can run on a number of different SQL
databases.
RTFW
Mark
CM Rahman Jr. wrote:
does anybody know any CDR server in public domain or low cost?
Thanks
___
Asterisk-Users mailing list
Oops sorry about that, as I said very new to this stuff, and guessing most of
this is stuff put in place by AAH. Here is what I show in my extensions.conf
for the dial macro, let me know if I missed anything else:
; Rings one or more extensions. Handles things like call forwarding and DND
;
Andrew Kohlsmith wrote:
On Monday 25 July 2005 23:26, [EMAIL PROTECTED] wrote:
Highly recommended to everyone to stay away from this issue
I do not have a name for the company right off hand, but they got sued
really bad when they tried 911 via VOIP and the 911 drop kept occurring in
different
Nope, the ast_db_get return -1 on failure.
The code is ok.
Sergio
asterisk_on_oelf ha scritto:
Is it possible, that you mean:
if (!ast_db_get(SCCP, d-id, result, sizeof(result))) {
instead of
if (ast_db_get(SCCP, d-id, result, sizeof(result))) {
in line 222 sccp_utils.c
Hello
My name is
Will.
I have a problem
with SIP on ASTERISK
How many ways it has
to register and to work in sip.conf?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Eric Wieling aka ManxPower wrote:
You do realize that host= only applies for calls from Asterisk to the
TNT, right? You need permit/deny to match for inbound connections. Your
Asterisk server is open to anyone that claims to be the user tnt.
That's not true; incoming INVITEs that don't
I found with some more testing that you have to setup a 5 digit number (or
something longer than your phone extensions) to make the voicemail work.
Now the trick is making the MWI work.
Rick
-Original Message-
From: Lull, Rick
Sent: Friday, July 15, 2005 3:55 PM
To: 'Asterisk Users
I have been learning AGI, and have got to grips with most of it, one thing
just confuses me.
I have written a PHP class that does input and outs stuff to the AGI. It all
seems to work. However on a stream file command. It's a bit different.
If I have a php file that just plays a file and hangs
Hi,
I am upgrading a
Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to
7.5
However in my first
attempt to go from V.5.1 to 6.0 this is hat happens:
- The phone
reboots
- The phone then
reads the fileOS79XX.TXT from the TFP server. In the file I added the
version
Hi to everybody,
i tried to find an asnwer before posting this, but
most astcc billing issues i searched refer to the case when no billing occurs at
all.
In my case i get only initial charges and any
subsequent minute does not count for billing.
In my iax.conf i entered the "notransfer =
Look this link:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html
Regards.
Jsalas
-Mensaje original-
De: wassim darwish [mailto:[EMAIL PROTECTED]
Enviado el: Tuesday, July 26, 2005 7:30 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] how to
Appears to be a diversified company (Google of it shows that they do web
design, VOIP, etc.). But this link caused me concern.
http://www.ripoffreport.com/reports/ripoff145784.htm
Thoughts? Experiences?
___
Asterisk-Users mailing list
I assure you I have read the asterisk handbook many times. The
immediate=yes is for picking up a phone on an fxs and having it immediately
dial an extension. I am looking for someone to dial an extension and have
it immediately pick up the phone on an fxs port.
-Chad
- Original Message
Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
You do realize that host= only applies for calls from Asterisk to the
TNT, right? You need permit/deny to match for inbound connections.
Your Asterisk server is open to anyone that claims to be the user tnt.
That's not true;
Once upon a time Tuesday 26 July 2005 8:00 am, Julio Arruda wrote:
Implement 1 single hard wire for this service and cover your tush!
This is what I tell my customers as well, not because I can't do it but
because they typically have the line there anyway for fax and/or security
which
I believe you need the OS79xx.txt file to
be the P003 file and the SIPDefault.cnf file to have the POS3 file name inside.
There are some docs on the wiki about it;
upgrading the Cisco phones can be tough.
Rick
From: Walid Azab
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, July
Andrew Kohlsmith wrote:
Could someone also cite the bugs.digium.com bug number for this, I'd like to
see the patch.
There is no bug number, it did not come through that channel (we created
the 'patch' internally after obtaining specific details of the problem
from a customer). The change
That's strange.
With your code my asterisk crashs everytime a new device tries to register. I
thought, that's because ast_db_get returns nothing or NULL.
(-1 will be returned if dbinit() fails.)
Jens
Nope, the ast_db_get return -1 on failure.
The code is ok.
Sergio
asterisk_on_oelf ha
Thanks for the reply, Adam.
If this is the case, it would seem to me (because the degradation
happens only after a period of time, and quite suddenly) that the issue
lies with digium's implementation of g729.
As an interesting note, I had the same problems using ulaw - ulaw over
the local
Hello all,
I've written the small patch to do what is said in the subject.
kpfleming sais that the same logic can be accomplished in the dialplan but I
could not find how (as I don't know how to verify if an agent is busy or not)
anyone of you have an idea?
Thank you
There was a thread about sixtel not too long ago. Quality is ok, when
it's working. DID requests can take weeks or months, some are never
answered. Termination usually works, and with decent quality. Looks,
feels and smells like a one-pony show -- and the pony needs a vet.
Try this:
Title: Message
I
believe you have to upgrade to 5.3 in order to go from unsigned to signed
executables. Once you're at 5.3, you can go directly to 7.5. I did
this recently with a couple of 7960s I had in the lab and it worked
perfectly.
Regards,
-
Brad
-Original
chouck wrote:
I assure you I have read the asterisk handbook many times. The
immediate=yes is for picking up a phone on an fxs and having it
immediately dial an extension. I am looking for someone to dial an
extension and have it immediately pick up the phone on an fxs port.
What does this
Ronald_Wiplinger wrote:
I would like to put / get some data from an MySQL database.
I want to use this MySQL database also via a web page.
bye
Ronald
app_addon_mysql or use RealTime.
-Matthew
___
Asterisk-Users mailing list
asterisk_on_oelf ha scritto:
With your code my asterisk crashs everytime a new device tries to
register.
the db code is ok.
For the crash problem please join the sccp mailing list and follow the
how to debug guidelines at http://chan-sccp.berlios.de
Sergio
Title: Message
Has anyone else seen
this problem? MWI works, but when you press the messages button the display
shows -1 urgent, 1 new, and 0 old.
Anyone know how to
fix this?
Chad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am wondering if anyone can help me figure out how to do something.
I'm running a simple perl AGI which at the end of the call, creates a
MySQL row with some of the tasks done during the AGI session.
However, when the call is unexpectedly dropped (possibly the caller
hangs up before
On Tuesday 26 July 2005 09:43, chouck wrote:
I assure you I have read the asterisk handbook many times. The
immediate=yes is for picking up a phone on an fxs and having it immediately
dial an extension. I am looking for someone to dial an extension and have
it immediately pick up the phone
Title: Message
Thanks
to all of you guys. I managed to fix it. It turned out to be that the ZIP file
has to be extracted inside the TFTP root not outside then copied to the TFTP
root. It is working now.
Thanks
Walid
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Watkins,
Eric Wieling aka ManxPower wrote:
Thank you for the correction. Is this new to CVS-HEAD, or does it apply
to 1.0.x as well? 1.0.x sip.conf.sample only lists the host option
for a peer, not for a user.
It only applies to 'peer' entries, but the important point is that
'peer' entries _are_
On Tue, 26 Jul 2005 10:24:20 -0400
Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Tuesday 26 July 2005 09:43, chouck wrote:
I assure you I have read the asterisk handbook many
times. The
immediate=yes is for picking up a phone on an fxs and
having it immediately
dial an extension. I am
Done both reload and Linux box re-load, after a
power down.
Still have request for Box
Password!
Regards and thanks
Mauro
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi denis
I am using Country ve,10,4Venezuela 10 ani 4 dnis
please let me know if I can do some test, or anything to help
Thanks
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
That is what I was worried about. Let me try and explain the situation a
bit better... Basically I have an existing P.A. system that works by simply
picking up the line that it is connected to. So basically I just pick up a
phone and I am instantly talking over the P.A. So what I wanted to
Good day all
Is there a fix for these errors yet for the junghanns cards
Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes
6 z1 1 z2 107
Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes
6 z1 10 z2 116
Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio
On Tue, Jul 26, 2005 at 08:51:35AM -0500, Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Could someone also cite the bugs.digium.com bug number for this, I'd like
to see the patch.
There is no bug number, it did not come through that channel (we created
the 'patch' internally after
Hi,
I'am new to * and googled/read a lot, but did not find (yet)
a lot of info to do the above.
Some months ago, I did find a 'story' from somebody having
put * between his PRI and current PBX as IVR, but I can not
find it back :-(
Any help/pointers are appriciated.
Txs
alex
--
NEW: aXs GUARD
Hello,
Anyone tried
reverting to SKINNY from SIP. I have a problem I cannot fix and need to get back
to SCCP to be able to use the phone.
ThanksWalid
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Title: Message
I
wentfrom 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have the
warning message (Protocol Application Invalid)
Please
any help.
Walid
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Watkins,
BradleySent: Tuesday, July 26, 2005 4:12 PMTo:
Tzafrir Cohen wrote:
There seems to be a ' Tag: v1-0' in the message but it is in the body.
Any header to filter by?
No, that is the only thing that the CVS commit messages contain that
indicate the branch/tag the commit went into.
___
Walid Azab wrote:
Thanks to all of you guys. I managed to fix it. It turned out to be that the
ZIP file has to be extracted inside the TFTP root not outside then copied to
the TFTP root. It is working now.
Walid, you should be able to unzip it anywhere and copy it into
the directory. It
On Tuesday 26 July 2005 10:06, Jay Milk wrote:
There was a thread about sixtel not too long ago. Quality is ok, when
it's working. DID requests can take weeks or months, some are never
answered. Termination usually works, and with decent quality. Looks,
feels and smells like a one-pony
Joseph wrote:
On Fri, 2005-07-22 at 16:53 +, Johann wrote:
Tim,
There isn't such a feature in 1.0.x. I don't know of any work being
down toward such a feature in the CVS either. You could try placing a
bounty on the feature to get it added or add it yourself if you are a
On Tue, 2005-07-26 at 18:11 +0300, Tzafrir Cohen wrote:
I suppose you refer to:
http://lists.digium.com/pipermail/asterisk-cvs/2005-July/007125.html
How do I track only the changes to the stable branch? For a user of
Stable most of the messages on the CVS list are rather irrelevant.
Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
Thank you for the correction. Is this new to CVS-HEAD, or does it
apply to 1.0.x as well? 1.0.x sip.conf.sample only lists the host
option for a peer, not for a user.
It only applies to 'peer' entries, but the important point
On Tue, 2005-07-26 at 12:03 +0900, Vic wrote:
Hi, everybody,
can someone please clarify Digium cards for me?
I was looking through the cards, and could not tell the difference
between TE410P and TDM400P.
The TDM400P is for FXS and FXO ports, up to 4 in any combination. The
On Tuesday 26 July 2005 11:31, chouck wrote:
That is what I was worried about. Let me try and explain the situation a
bit better... Basically I have an existing P.A. system that works by simply
picking up the line that it is connected to. So basically I just pick up a
phone and I am
Hi all,
Today I've stumbled upon a very strange behaviour with an analog fxs/fxo
gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html)
connected to a CVS HEAD(from today) Asterisk server. This manifested
itself after enabling the CallerID on the pstn lines connected to the
FXO
Would appreciate your answer. is it related to span in zaptel.conf?
Michael
-- Forwarded message --
From: Michael Jia [EMAIL PROTECTED]
Date: Jul 26, 2005 1:26 AM
Subject: Re: why zap call transfer fails?
To: Asterisk-Users@lists.digium.com
Some further observation. Using the
To deal with this problem, I've used DeadAGI in the h, t, and i events
(as appropriate) to detect and call the script again.
As such:
exten = h,1,DeadAgi,perlscript.pl|${UNIQUEID}|hangup
I pass the word hangup as the second word to flag to the AGI script
that this is a hangup event, not a
On 18:21, Mon 25 Jul 05, Johann Steinwendtner wrote:
Hello !
I would like to get working a Fritz PCI card using chan_misdn
operating in ptp mode.
As far as I know the fritz cards do not support ptp mode.
We tried all the possible config file options with chan_capi
and in the end we trashed
I am having one problem with the Polycom 600 phones. All phones on the
local network are fine and indicate presence to other phones perfectly.
One phone that is outside the network can see presence indications of
the other phones correctly, but that phone always shows as off the hook
to
As previously mentioned on the lists by Olle Johannson, we are actively
trying to get Asterisk in shape for a 1.2 release within the next 60
days. To accomplish this, we need a few things to happen:
1) A feature freeze - This will occur at the end of this month, with no
new feature
Hello!
I have an [EMAIL PROTECTED] instalation with 7 users working OK, and I'ld like
to implement either a
-- Web dial feature, where the user would fill one form field with a phone
number and a connection would be created between his extention and the
entered number.
OR
-- Dial using an URI
But which packages are you using?
libunicall
spandsp
asterisk
zaptel
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r
http://www.isolve.com.br
On 26 de jul de 2005, at 12:27, Jose Chiantera wrote:
Thanks for reading this. Ive been pulling hair
for days trying to resolve this, and any help someone can give me would be very
much appreciated.
I have an Asterisk box that is basically a P4-3GHz, a
Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE
hard-drive with
You
need to match the extensions you have in voicemail.conf to the callerid you're
passing to voicemailmain(). For5 digit extensions it would be
exten = 22999,1,VoiceMailMain(s${CALLERIDNUM:-5)
Louie
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Appears to be a diversified company (Google of it shows that they do web
design, VOIP, etc.). But this link caused me concern.
http://www.ripoffreport.com/reports/ripoff145784.htm
Thoughts? Experiences?
My recommendation would be to stay away. They are unresponsive and they are
There are several postings about this on the web. I don't have the
details handy
anymore but a google search (or search of Cisco's site) should turn up the
answer. I remember seeing this with v7.0 code because of a problem with
the image released from Cisco. If you don't find the answer email
1 - 100 of 187 matches
Mail list logo
