On Thu, 19 Jun 2008 11:36:27 +0200, Vincent
[EMAIL PROTECTED] wrote:
Will do, although it could be a problem in the Zaptel code, which is
not written by the mfg. Thanks.
I also notice that I can't restart the driver:
# /usr/local/etc/rc.d/zaptel restart
zaptelkldunload: can't unload file:
Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21
The update seems to work ok, when asterisk is started all is fine.
However after some time it is not possible to call anymore, my Snom
display simply shows Not available and incoming calls from the PRI fail,
like the PRI is not connected.
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
The power cable is plugged into the card, the port is configured to use
fxo-signalling. Also,
On Fri, Jun 20, 2008 at 10:15:48AM +0200, Paul Schewietzek wrote:
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
Is Asterisk actually running?
On Thu, Jun 19, 2008 at 3:50 PM, Paul Belanger [EMAIL PROTECTED] wrote:
List,
Could anybody speak to the status of development in 1.6 branch? I
know support for SIP over TCP is pretty new / experimental but it
seems active development of it has slowed or stopped in recent months.
Is that a
Most SIP clients have a logging ability.. you can use those.. but
turning on debug on the server is the best mechanism, as its whats going
on there that counts.
sip set debug options
And if you want to get really into the lower levels, then tcpdump will
let you capture the packets for offline
maggie1:~# cat /proc/zaptel/*
Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) HDB3/CCS/CRC4
IRQ misses: 31
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In
Doug wrote:
There is a bug in these units that won't let
you put punctuation in the extension name.
A Grandstream product with a bug... what an unusual concept. cough
Seriously, with all the grief I've had with GXP-2000s, BT-200s and
GXV-3000s, I wouldn't touch Grandstream gear with a
Not to my knowledge. The PAP2 is designed as a fairly basic ATA and
each line registers as a separate SIP extension. I have no doubt that
you could use the call forward feature within the ATA to divert calls to
the second extension if the first one was busy, but AFAIK you'd need to
register
Hello, I noticed a bit problematic behavior in ChanSpy function. This is the
scenario:
1. agent is making a conversation,
2. I call an extension with ChanSpy and start listening (so far so good),
3. agent completes the call (I am still on the ChanSpy extension),
4. new call is distributed to an
Tilghman Lesher [EMAIL PROTECTED] writes:
One very big benefit of using a database with cron jobs is that your web
application does not need to run as the same user (or otherwise weaken
security permissions) as the Asterisk daemon. If running as the same user,
you'd have to either set both
Thanks guys, we solved the problem. There was a non-standart cable
between the devices that didn't work correctly...
o.O
-paul
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix,
They still have issues. If you use TCP and reboot the server, the phone
will never reconnect as it tries to use a closed TCP session. I opened
a ticket with them and after a week their answer is . use udp.
Rob Hillis wrote:
Doug wrote:
There is a bug in these units that won't let
you
Hi Remco,
On Fri, 2008-06-20 at 09:33 +0200, Remco Barendse wrote:
Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21
I did the same yesterday.
The update seems to work ok, when asterisk is started all is fine.
Yup.
However after some time it is not possible to call anymore, my Snom
Hey can you guys give me a mailing list/group where I can have little on
commercial discussion?
Regards
jimmy
--- On Thu, 6/19/08, Steve Totaro [EMAIL PROTECTED] wrote:
From: Steve Totaro [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Trouble with PRI config
To: Asterisk Users Mailing List -
Remco Barendse pisze:
However after some time it is not possible to call anymore, my Snom
display simply shows Not available and incoming calls from the PRI fail,
like the PRI is not connected.
Reverting back to 1.4.20.1 solves the problem.
I tried re-installing and re-compiling 1.4.21
Hello Gordon,
On Thu, 19 Jun 2008, Jan Prunk wrote:
* You might want to try:
**
** exten = _**.,1,Pickup(${EXTEN:2})
** exten = _**.,n,Hangup()
*
*
** Ok I have tried adding these 2 lines, and the error which I get when calling
** 01 5863165, which then rings extension 65, and I try to accept
You can ask any hardware/soft choice/cost here, but for support and
commearcial offers, asterisk-biz or asterisk-biz forum on asterisk.org
2008/6/20 Jimmy Jill [EMAIL PROTECTED]:
Hey can you guys give me a mailing list/group where I can have little on
commercial discussion?
Regards
jimmy
On Fri, 20 Jun 2008, Jan Prunk wrote:
One other thing - do you have
exten = 65,1,Dial(SIP/65)
Yes I do have this:
exten = 65,1,Dial(SIP/65,30,rtk)
exten = 65,n,Hangup()
exten = 70,1,Dial(SIP/70,30,rtk)
exten = 70,n,Hangup()
As pickup works on the extension not the channel... (ie.
Thanks Tony,
First of all, thanks for answer.
The possible solution to solve the problem with auto hangup is to use 'h'
extension, which can execute some commands after hanging up, here we call
MeetMeAdmin(confno,K) from either caller or callee, what will hang up call
when caller drops the call
I will answer my own question :-)
The following line in the app_chanspy.c was causing my problem:
waitms = count ? 100 : 5000;
and of course sequentially then the line:
res = ast_waitfordigit(chan, waitms);
in the same beginning of the same for loop but in the next iteration.
I have changed
Hello Gordon,
Same error if I change the extension ...to ring directly 5863165 (see the
maximum group number is 63)
-- Accepting overlap voice call from '015852977' to '5863165' on channel
0/1, span 1
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1]
Hi
not the issue here, but yo asked and thus I'll answer:
On Fri, Jun 20, 2008 at 11:51:27AM +0200, Paul Schewietzek wrote:
Could it be
possible that a channel with FXO signalling ignores the group= option in
zapata.conf?
A. no problem with that.
B. This is only related to dialing out, not
On Friday 20 June 2008 02:33:04 Remco Barendse wrote:
Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21
The update seems to work ok, when asterisk is started all is fine.
However after some time it is not possible to call anymore, my Snom
display simply shows Not available and incoming
Hi,
[2008-06-20 15:38:19] WARNING[9772]: channel.c:4347 ast_get_group: Ignoring
invalid group 5863165 (maximum group is 63)
we had a similar error, found somewhere (voip-info.org?) this solution:
exten = _**2XX,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _**2XX,n,PickUp(${EXTEN:2})
where
Like I said, I wouldn't touch Grandstream gear with a barge pole any
more. I can point you at at a few companies that hold exactly the same
opinion too.
Don't expect any useful action on the TCP stuff anytime soon - at least
not on a version of firmware that doesn't introduce any other nasty
All,
I did finally get my server up and running. I thought I'd share some
tools I used.
1) use this to see if asterisk is even listening on port 5060 with UDP
nmap -sU -p5058-5062 localhost
Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at
2008-06-20 10:49 CDT
Hi all,
i've recently acquired a callcentric account.
i've perfectly setup my sip.conf and extensions.conf to make outgoing calls.
but the problem is with incoming calls! when i call in, asterisk doesnt even
see the incoming call!
how is tht possible!
please see the following my config:
Great, thanks guys!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 19, 2008 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Website callback
On Thu, Jun 19, 2008 at 9:57 AM,
Hi All,
I've been playing with Openfire Asterisk-IM plugin installed on the
same server with Asterisk 1.4 with MySQL as the Openfire database.
Using Spark IM as the client on user machines.
It seems to work fairly well, not too bad to install. This first
thing I notice is all the packages that
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson
[EMAIL PROTECTED] wrote:
So now the PBX is over 1.2 Gig for the installation. Typical PBX
installs are under 600 Meg. This makes me wonder about server
stability, reliability and performance as uptime creeps on and user
count increases over 50
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension
Hi everyone!
I have been reading for a couple of days online in order to setup a remote
phone, but no luck so far. The remote phone registers and I can call
extensions, but the call only lasts 19-21 seconds before it drops (when I
call from remote phone). When someone calls the remote phone we
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are
Yes. Both Asterisk and Cisco is behind it.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To:
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To:
Greetings..
i'm having a disconnection problems with Calls comming to my Call Center..
i'm using the free version of G729 .. and i'm starting to suspect it would be
the reason.. i just need to know if it's possible or there will be other
problems??
i asked the technician in the location to
Sounds like a network connectivity issue. Start at your physical layer
and work up.
Tariq .. wrote:
Greetings..
i'm having a disconnection problems with Calls comming to my Call
Center..
i'm using the free version of G729 .. and i'm starting to suspect it
would be the reason.. i just need
See below:
Erik Anderson wrote:
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson
[EMAIL PROTECTED] wrote:
So now the PBX is over 1.2 Gig for the installation. Typical PBX
installs are under 600 Meg. This makes me wonder about server
stability, reliability and performance as uptime creeps on
Dear List,
I have a customer who owns a little Motel, and he wants to upgrade to a
Asterisk PBX. There is one analog phone per room (aprox 80), and the cable
is CAT 3.
Any recommendations on what card to use?
TDM24XXP vs Channel Bank?
Regards,
Ing. Arturo Ochoa N
Electrosystems
I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.
I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
80 rooms? I guess you and I have slightly differing opinions as to what
a small motel is. :)
If you have 80 analogue channels, then you'd need 4 TDM2400P cards.
Unless your server is powered by a small nuclear reactor, you'll be
better off with either 3 E1 or 4 T1 channels banks or 2 32
Well to be honest, our experience with asterisk never works with under NAT.
if you got DMZ then it will otherwise don't hold your breath for it.
If you want to use it as a production server
Your option is 1. Get a Real IP
2. there is no 2 really just get an ReaL Public IP
Sam
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