Asterisk wrote:
Hi guys,
I have an IVR configured in my PBX, which callers use to browse thru the
list of stores. Once they choose a store, the call gets redirected to
that store (obviously using Dial() application). Now, my question is:
Each of this calls is logged in the calllog as
Mmm i have set it in my MySQL Database in the row: Variables
buggymwi = yes
But can't see MWI
Regards rene
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Zachary
Whitley
Gesendet: Freitag, 27. April 2007 00:09
An: Asterisk
Hi,
I'm using Grandstreams as the agents phone of a queue. Attended
transfers in a normal situation (direct call to the extension) work
fine, but when the agent has a queue call and tries to transfer it to
another sip extension the called party is hung up.
Transfer is to pick another line on
I haven't tried the app_conference yet. I want to know if the conference is
consisting of 3 users with G.722, does the app_conference perform
transcoding? If it is not, then app_conference will solve the issue of
having conference consists of only G.722 user since no transcoding is
needed. Is my
hello, I have instaled a analog line, and when I call on the console apears
that:
I want to redirect the call to 101 extension.
*CLI -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still
Alex Balashov wrote:
On Thu, 26 Apr 2007, Knud Müller said something to this effect:
Tzafrir Cohen wrote:
On Thu, Apr 26, 2007 at 08:43:17AM -0500, Jason Fuermann wrote:
1024 open files will get you around 120 concurrent calls.
8 file-handles per call? Why is that?
Depends on what
Hi all,
I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server. Is it possible to control the call pass through those cards?
Any example for me to reference?
ango
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Thanks for your reply.
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Both actions don't help much for the file descriptor growing.
What I want to know is:
Do I need to reboot if I insert the following in /etc/security?
*
On Thu, 26 Apr 2007 12:39:32 -0400, Dave Miller wrote:
Dave Miller wrote on 4/26/07 11:46 AM:
We upgraded our asterisk server to 1.2.18 last night to pick up the
security update. Since then, any calls coming in on IAX2 links get
dropped if they try to enter a MeetMe conference room.
Hi, the holdtime in queue log entry is not what we want, that holdtime
only records the duration that caller stay in the queue before an
agent answers. However what we want is the duration that agent put the
customers on hold(i.e music on hold, for SIP, the device will send a
re-Invite as I
I think you need to explain control the call pass through those
cards a little please.
On 4/27/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server. Is it possible to control the call pass through those cards?
Any
It was fixed in 1.2.17.1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 26 April 2007 21:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17
On Wed,
SIP == SIP [EMAIL PROTECTED] writes:
SIP Premium Rate Services think like 900 and 976 numbers in the
SIP US, but not every country allocates a particular block of numbers
SIP or prefixes to its premium rate services, so with some, they're
SIP pretty close to impossible to block.
Perhaps
In article [EMAIL PROTECTED],
Dinesh Nair [EMAIL PROTECTED] wrote:
is there a patch for this against 1.2.18 ? it would sure help those who're
tracking the release tarballs instead of having to svn and compile it.
Have a look at:
I also have
nat=no
qualify=no
I haven't checked to see if they're necessary. I think I've read some
suggestions that the phone needs to be on the same subnet as the
asterisk server but I haven't been able to check that either.
On Fri, 2007-04-27 at 09:19 +0200, René Enskat wrote:
Mmm i have
Erik Anderson wrote:
On 4/26/07, Per Jessen [EMAIL PROTECTED] wrote:
I was just browsing my local suppliers list of headsets - not a
single
one with a single 2.5mm jack. Either USB or 1-2 3.5mm jacks.
Can anyone recommend a headset that works with e.g. SPA-921 and -941?
Try your local
Whenever I turn the weight option on, it locked the *. It happens several
times a day ( abount every two to three hours). When this happen, the
incoming call can still connect to * but will not hear any music on hold. If
I issue the 'show channels' command, it shows the connected channels
Hi,
Is it possible to host call waiting service on Asterisk for a SIP device?
What i am trying to achieve is that while a SIP user is busy on a call and
a new call for that user comes in, asterisk should play the call waiting
tone to that user.
I have a vague idea that if i can get hold of the
I've had mixed results with changing ulimit and not restarting asterisk.
Best bet is to stop and start asterisk so that it calls a new shell
Rilawich Ango wrote:
Thanks for your reply.
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
On 4/26/07, Michael Kamleitner [EMAIL PROTECTED] wrote:
On 4/26/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Michael Kamleitner [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 17:47:34 +0200
however, I've continued to experiment again and again, and strangely it
seemed to work _some_ times,
HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in
chan_sip (sip.conf) i've found nothing, only a misterious externip = stun...
But where i have to put the ip of stun server?
No infos around Google
On Fri, 27 Apr 2007, Marco Ciacci wrote:
HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in chan_sip
(sip.conf) i've found nothing, only a misterious externip = stun...
But where i have to put
Brad Sumrall wrote:
I would not rule your firewall out as the problem!
Port 5060 is only the authentication port, the rtp stream is normally 10,000
thru 20,000.
Some of your phone may have STUN modules on them.
Open 10,000 thru 20,000 and 5060 on the firewall.
Stick some holes in it for testing
Hi,
I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them. I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.
What's the best way/device to turn a wired Polycom 501 (or any Polycom
I was afraid of an unavailable NFS mount hanging the app and I also
wanted to keep all of the communication over IAX for simplicity sake. I
also hacked together my own MWI over IAX. I did write ups of how I
did both.
http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voic
Yes, i use Asterisk behind a retricted-cone NAT in many cases
(portforwarding on router, nat=yes and externip=dyndns o static ip),
but not ever is a possible solution, sometimes my client isn't
router's owner (telco's router) and i can't do port forwarding.
I known that 1.4 has stun support,
Hi,
I need to use a new table in my Asterisk database, to add new
data. I want to use the data of this new table in my Asterisk
app_voicemail.c source code. I want to know if someone has an idea how to do
it.
Thank you.
___
On Fri, 27 Apr 2007, Mike wrote:
Hi,
I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them. I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.
What's the best way/device to turn a
Hello all:
I upgraded to 1.4.3 last night and use MySQL for CDR.
I have noticed that 1.4.3 seems to log a lot of crap to CDR that
1.4.2 did not. I use a few macros in my dialplan to handle outgoing
calls (lcr type stuff) and in addition to the proper CDR for the call
itself I also have records
Hi,
Is it possible to have early audio while waiting for answer in a Dial()?
Say that I want to do this:
1,Progress() // Establish early audio possibillites
2,Dial(SIP/user,20,z(repeated-musicfile))
Where z would be like a function for playing early audio.
Or z would just start MOH
Hi Mike,
How close together are these phones? If you have a few clusters of them, you
can use the Linksys WRT54G devices to act as wireless bridges (with some
open source firmware - I use DD-WRT). Each device will give you 4 ports to
plug into. It's not a particularly cost effective solution to
On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:
[snip]
As suggested earlier I replaced this with:
/etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1
[snip]
dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.17.1
On Fri, 2007-04-27 at 11:49 -0400, Alex Robar wrote:
Hi Mike,
How close together are these phones? If you have a few clusters of
them, you can use the Linksys WRT54G devices to act as wireless
bridges (with some open source firmware - I use DD-WRT). Each device
will give you 4 ports to plug
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:
SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway
I get the following error:
Unable to find a codec
Asterisk 1.2.17
When try to play moh, I can only use old format in musiconhold.conf file to
play moh like this:
[moh_files]
default = /var/lib/asterisk/mohmp3,r
If I use the new format like this:
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
I hear no music at all.
Can anybody
Cross posted from -users to -dev
I was looking at adding this functionality in last night.
I saw that in app_queue when a call is bridged it determines hold time.
Using the following:
holdtime = abs((now - qe-start) / 60);
and for queue.log the following:
(long) (callstart - qe-start)
My
Failed
On 4/26/07, gc [EMAIL PROTECTED] wrote:
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On Fri, Apr 27, 2007 at 12:31:16PM -0400, gc wrote:
Asterisk 1.2.17
When try to play moh, I can only use old format in musiconhold.conf file to
play moh like this:
[moh_files]
default = /var/lib/asterisk/mohmp3,r
If I use the new format like this:
[default]
mode=quietmp3
Hello,
Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP-H323 calls?
I mean exactly what canreinvite=yes option do in SIP-SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about
There is a live conference call on Asterisk regarding Adhearsion
occurring now. Check out www.X2Z.eu http://www.x2z.eu/ for details.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
Rilawich Ango wrote:
Thanks for your reply.
You're welcome. = )
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
I don't think it can cause any problems, but I've never had to adjust
anything in the /proc filesystem, and I'm
On Fri, 27 Apr 2007, Elman Efendiyev said something to this effect:
Hello,
Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP-H323 calls?
I mean exactly what canreinvite=yes option do in SIP-SIP calls.
I don't need a transcoding, only a signaling conversion, and
I was afraid of an unavailable NFS mount hanging the app and I also
wanted to keep all of the communication over IAX for simplicity sake. I
also hacked together my own MWI over IAX. I did write ups of how I
did both.
http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voic
Hi everyone
I have a Digium card TE110P and after plug it, turn on the computer and
configure, the LEDs don't light up in spite of what Digium FAQ says about the
LEDs:
When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP?
... The TE110P LED's will light up RED when the span is
Happens to me again, SIP-Zap or SIP-SIPProvider with a quite simple
dialplan, it generates an 's' record in the context of both sides just
like if it was doing a per-channel CDR instead of a per-call...
Scott Lykens wrote:
Hello all:
I upgraded to 1.4.3 last night and use MySQL for CDR.
I
--Original Message Text---
From: Mike
Date: Fri, 27 Apr 2007 10:24:05 -0400
Hi,
I'm stuck doing an install with Polycoms at a small office with no RJ-45. They
went wireless 100%, poor them. I insist on using Polycom unless it's
impossible because that's what I am standardized on for many
Hello,
We've released another update to our astGUIclient suite: 2.0.3
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality and the
Hi, is there any way to configure a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.
Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus
How do you overcome the 302 moved temporarily messages? I tried using
a wireless bridge with one of my Cisco phones Polycom 301's and when
I tried to receive a call it would give me the 302 error.
On 4/27/07, Michael Graves [EMAIL PROTECTED] wrote:
--Original Message Text---
From: Mike
Date:
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 27, 2007 12:55 PM
Subject: Re: [asterisk-users] Problem of configuring musiconhold.conf file
On Fri, Apr 27, 2007 at 12:31:16PM -0400, gc wrote:
Asterisk 1.2.17
When
equis software wrote:
Hi, is there any way to configure a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.
Example:
No more than 3 simultaneus calls to extension 33
No
Hi,
I would like to set up a call center with free seating agents. However I
would like to indicate the agent status somehow on the terminal, to tell
the agent if she has been logged out due to non-answer.
Does anyone has a good idea how this can be achived?
best regards
--
Alexander Topolanek
On Fri, 27 Apr 2007, equis software wrote:
Hi, is there any way to configure a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.
Example:
No more than 3 simultaneus
Michael and all those who replied,
This Linksys WBP54G does seems to be what I need, but it also seems very
much made for Linksys phones. Isn't there some sort of equivalent thing
that comes with it's own power supply (at the cost of needing another outlet
for the phone)?
Alternatively,
Manu,
Perhaps it is possible to do this by running the call through an RTP
proxy (there are various) that supports inserts muxing outside audio
feeds into the media stream? Then you would just need to implement
some sort of middleware layer to map Asterisk SIP channels to corresponding
I´m trying this:
exten = 99,1,Set(GROUP(99) = GROUP99)
exten = 99,2,GotoIf($[${GROUP_COUNT(99)}1]?103)
exten = 99,3,Goto(context2,s,1)
exten = 99,103,Hangup
but doesn't work...I call to extension 99 from two different phones and
Asterisk sends both to 'context2'.
On 4/27/07, Eric ManxPower
I have a few commands I wish to run after a hangup.
It looks like only the first 2 commands are run after hangup.
I am using 1.4.3
How can I get the entire loop to run 10 times. ( I know my example just
has noop's but its an example).
exten = h,1,Set(i=1)
exten = h,n,While($[${i} 10])
exten
On Fri, 27 Apr 2007, Eric ManxPower Wieling wrote:
equis software wrote:
Hi, is there any way to configure a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.
Example:
From: Josu Lazkano Lete [EMAIL PROTECTED]
Date: Fri, 27 Apr 2007 10:09:56 +0200
hello, I have instaled a analog line, and when I call on the console apears
that:
I want to redirect the call to 101 extension.
*CLI -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1
Any way to use chan_bluetooth as FXS?
Yuan Liu
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A pop up on their pc display using Adhearsion to drive the resulting
logged in/out popup?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
http://click.mexuar.com/webuser/click/7/userurl/Cognation
Using the latest SVN of zaptel and asterisk, I can no longer receive
incoming analog calls. The caller just hears it ringing but Asterisk
doesn't pick up.
I am seeing these error messages:
[Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID
returned with error on
mount -o intr,nolock ought to do the trick. we're using those
options now, but thankfully haven't had reason to find out if they work
or not yet.
CP
Doug Garstang wrote:
No, you can get Asterisk and NFS to work fine together. It was in my
past job, so I can't remember the exact
On Fri, 2007-04-27 at 11:32 -0400, Scott Lykens wrote:
Hello all:
I upgraded to 1.4.3 last night and use MySQL for CDR.
I have noticed that 1.4.3 seems to log a lot of crap to CDR that
1.4.2 did not. I use a few macros in my dialplan to handle outgoing
calls (lcr type stuff) and in
The Asterisk.org development team has released Asterisk version 1.4.4.
A good number of significant bugs have been fixed in the past few days,
so a new release was made to get these fixes to the community as soon as
possible. Some of the fixes include:
- Fix a crash in chan_zap
- Fix some
oliver,
what gateway provider are you referring to?doesn't your sip phone
connect directly to * as your diagram indicated?
DSL providers should not be doing any codec anything!
daveC
Oliver Brandt wrote:
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use
I use Asterisk in my house. Each phone is a different extension. I
really like the ability to have multiple simultaneous calls in the
house. However, I do miss being able to be able to pick up a phone in a
different room. Currently, I have to either transfer the call or
transfer the call
Two words for you... parking lot.
Try to transfer your call to extension 700 and see what hapens...
On 4/27/07, Jim Duda [EMAIL PROTECTED] wrote:
I use Asterisk in my house. Each phone is a different extension. I
really like the ability to have multiple simultaneous calls in the
house.
On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:
[snip]
As suggested earlier I replaced this with:
/etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1
[snip]
dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.17.1
You can purchase the Linksys part PA100-NA and plug it into a WBP54G and
then ignore the power connector hanging off the WBP54G.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: April 27, 2007 3:00 PM
To: 'Asterisk Users Mailing List -
You can look for headsets made for Motorola cell phones. Also, Plantronics
has some compatible models - I can dig up part numbers if you're interested.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Per Jessen
Sent: April 27, 2007 8:32 AM
To:
Hi all,
I've compiled zaptel drivers and reconfigure asterisk afterwards from
source --with-zaptel.
Modules are loaded accordingly:
asterisk-1.4.2 # lsmod |grep z
Module Size Used by
ztdummy 5472 0
zaptel194504 5 ztdummy
crc_ccitt
Am Freitag, den 27.04.2007, 16:25 -0400 schrieb Dean Collins:
A pop up on their pc display using Adhearsion to drive the resulting
logged in/out popup?
Sorry, I forgot to tell that the Agents don't have PC's. Is it possible
to trigger the MWI from an AGI-script that is fired when an Agent is
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