12 mar 2007 kl. 23.33 skrev Nikhil Jogia:
Bruce Reeves wrote:
Does SIPAddHeader(Alert-Info:) not do it?
No, but from another thread, setting the _SIPADDHEADER variable works.
You misunderstand. The prefered way is to use SIPAddHeader(Alert-
Info: slakfj aslkfjaklsdf)
But in the
Lee Jenkins wrote:
Try putting quotes around the value. I played with it a while back only
a little, but I can't remember if quotes did it or I ended up having
stripping the quotes off myself when I retrieved the value ...
My first mail was copy/paste, so I'm positive I didn't make any error
Hi
i have a big change or bproblems to update a asterisk 1.2.12 server to
asterisk 1.4.1 ?
Thanks bye
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13 mar 2007 kl. 09.53 skrev Noc Phibee:
Hi
i have a big change or bproblems to update a asterisk 1.2.12 server
to asterisk 1.4.1 ?
There won't be any problems if you take some time to read the
available documentation
to see what changes you need to do in your configuration.
Make sure
Ken D'Ambrosio wrote:
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Are you talking about the warble that
On 9 Mar 2007, at 17:51, Octavio Ruiz (Ta^3) wrote:
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the
calls from
the PRIs to * boxes using IAX I'm planning on having four or five
* boxes.
The * boxes run AGI
On Mon, 2007-03-12 at 22:12 -0700, Trevor Peirce wrote:
Patrick wrote:
Hi all,
Does anyone have any experience with creating a Single sign on (SSO)
concept where if someone logs in on their PC the phone next to that PC
is also automatically assigned to that user?
Yup, I've done
Hello
I have following problem.
After upgrade from 1.4.0 to 1.4.1 my musiconhold stops immediately after
start.
Bellow some logs from 1.4.0 and 1.4.1 (same configs and situations)
First, the one from 1.4.0 (everything works)
[Mar 12 13:44:00] -- Executing [EMAIL PROTECTED]:1]
KD == Klaus Darilion [EMAIL PROTECTED] writes:
KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I
KD have a displayname with special characters?
KD E.g. if I want to have the Umlaut ä in the display name:
KD callerid=Jeff Gräser 11
Is your sip.conf UTF-8-encoded?
/Benny
Mark Davies wrote:
I’ve just been told from an ex workmate that my VSP (who I used to work
for) has put an anti flooding limit of 80 SIP messages per IP per minute
in place.
I run the phone system for a facility that has a lot of extensions, but
would rarely have more than 4 or 5
Hi,
Is there a way to unlink CallerID and the CDR values? I'd like my CDR to
have, in the src column, the extension of the person calling, for my
records (let's say 201). But if that person is calling outside the
company, I want the callerid to show 555-555-1234).
At first sight, the two
Wai Wu wrote:
I am use Fedora 3, and run into a 1.4 compile issue.
I recommend you to start using Cent OS 4.4 - it's basically RHEL.
--
Tomislav Parcina
[EMAIL PROTECTED]
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Dear Lewis,
Can you please post you gtalk.conf and jabber.conf for me? I also make
it under Fedora Core 6. But I got no audio at all.
I use X-Lite as SIP client (under NAT).
2007/3/7, Ronald Lewis [EMAIL PROTECTED]:
I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got
That does sound low, especially if you have multiple devices behind a NAT.
I have customers with 8 analog lines going into their analog phone system
and just have 4 ATAs with 2 lines each. Of course, all of this traffic
would seem to come from the same IP!
On 3/8/07, Mark Davies [EMAIL
Hi all,
i need help to implement a voicemail scenario. What i
am trying to do is the following.
user X dials a direct access for user Y voicemail and
is asked to enter a number (e.g 12345678) and then
leaves a message. Then asterisk sends a notification
with attachement. The problem is that i
hello,
We encountered signaling problem with a french national carrier.
They ask us, which signaling is configured on our single E1.
I need to know if it's ETSI, VN4 or VN6.
I know what ccs, and hdb3 mean but I do not succeed to make the link
between the signaling type.
I searched through RFC
On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote:
On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote:
More importantly, how many calls per day and how long per call.
Then you can figure out the other bits.
He wants to make 50 simultaneous calls. What difference does the length
and frequency
I've got IAX2 setup between two servers with this config:
I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is
192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls
from asteriskm to asterisk1 which will run an AGI IVR for the call.
Config is below,
His vindictive dialer isn't playing while it is listening to rings or
busy signals.
Forgive my ignorance, but what on earth's a vindictive dialler? Is it one
with a strong sense of revenge? :-)
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit
Greetings list,
Quite a few of our users seem to be experiencing poor voice quality when
they're using internet connections over which we have little or no control
(i.e. they're using their own router with no QoS, etc.). Some of these
connections are giving a qualify time within asterisk of
For startes I will keep it on the list and we can discuss some major
concepts, and I will possibly make some contact off list later for the
nitty-gritty :)
In-reply to Steve:
I did have a look at the bicomsystems product and it does appear to do
everything I am looking for. However, I have looked
On Mon, 2007-03-12 at 23:51 +0400, NetSys Admin wrote:
Hi everyone,
I'm completely new to Asterisk and before I buy any card, I would like to
ask for some information.
1. We'll be using analog PSTN phone lines. Is there anything that I should
ask the telecom company before I buy the
Ivo Zivkov wrote:
Sorry, I can only give you a general outline, because the code is
proprietary.
Call anywhere *from* anywhere... for just 12 cents a minute! (Some
restrictions apply; see 47 page contract for details)
-Stephen-
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Who is tired of dealing with DST changes?
I have asterisk running on FC4, FC4 has been patched and shows the
correct MDT timezone and time.
Email notifications of voicemail show the message time an hour early
(standard time, not daylight). This si the time in the message body, not
the
can you tell me about your physical layer cable..
i know that in frensh (I m talking about France Telecom) that they use
1,1,0,ccs,hdb3,crc4
and euroisdn pri_cpe
2007/3/13, Cedric MILLET [EMAIL PROTECTED]:
hello,
We encountered signaling problem with a french national carrier.
They
On 3/13/07, Chris Bagnall [EMAIL PROTECTED] wrote:
His vindictive dialer isn't playing while it is listening to rings or
busy signals.
Forgive my ignorance, but what on earth's a vindictive dialer? Is it one
with a strong sense of revenge? :-)
A normal predictive dialer determines from
Hi all,
In your experience, what is the maximum number of *concurrent* zap channels
that you've ever tried with one box of Asterisk open edition?
In my case, the max that I've tried was 63 simultaneous connections in a
Quad T1/E1 card installed on a Intel Pentium D 3.4Ghz 2GB ram system.
Your
My solution.
With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config.
Compile and without rtc module, load ztdummy. It work good with usbcore
and uhci_hcd modules.
I have installed libusb-dev for my debian etch system. Now, my kernel is
2.6.20.2, but it work good with
Damon Estep wrote:
Who is tired of dealing with DST changes?
I have asterisk running on FC4, FC4 has been patched and shows the
correct MDT timezone and time.
Email notifications of voicemail show the message time an hour early
(standard time, not daylight). This si the time in the
The problem is rtc module. My servers don't have a standard pc chip
for it.
I like a ztdummy working with genrtc. Exist this option ?
Now My ztdummy work with usb clock
Germán Aracil Boned escribió:
And If I execute:
./zttest -v
I can see:
Opened pseudo zap interface, measuring
On Tue, Mar 13, 2007 at 12:18:50PM -0500, Héctor Maldonado wrote:
Hi all,
In your experience, what is the maximum number of *concurrent* zap channels
that you've ever tried with one box of Asterisk open edition?
With Zaptel, the limit is pretty clear: the number of channels your
hardware
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3)
exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
The communication problem boiled down to iptables rules, but I'm still
getting the No private structure for packet? error message. It doesn't
seem to cause any problems and only occurs when an IAX2 peer has been
unavailable for at least three minutes, but I would like to know why it
happens if
I simply called the vendor I bought it from. Myriad.
Call Andy: (212) 366-6996 x111
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Saturday, February 24, 2007 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Try
exten = s,1,GotoIf($[${MACRO_EXTEN} = 220408XXX]?2:3)
exten = s,2,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Tue, Mar 13, 2007 at 10:44:08AM -0600, Damon Estep wrote:
Who is tired of dealing with DST changes?
I have asterisk running on FC4, FC4 has been patched and shows the
correct MDT timezone and time.
Email notifications of voicemail show the message time an hour early
(standard
On Tue, Mar 13, 2007 at 06:25:37PM +0100, Germán Aracil Boned wrote:
My solution.
With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config.
Compile and without rtc module, load ztdummy. It work good with usbcore
and uhci_hcd modules.
I have installed libusb-dev for my
I'm having issues with a Cisco 7970. It seems to ignore minor
changes in its config file. Is there something like the versionstamp or
some other setting I need to increment in order to get the 7970 to
update each time? It does seem to download the file from the TFTP
server, but it never
Try
exten = s,1,GotoIf($[${ARG1:0:5}=220408]?2:3)
This looks at the first 5 digits of ARG1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 12:39 PM
To: Asterisk Users Mailing List - Non-Commercial
I don't believe this will work. He wants it to goto if EXTEN =
220408235 or 220408743 or any other digits for the last 3 of the
extension block 220408xxx. When Asterisk processes both his and your
line it's going to look to see if the EXTEN is exactly 220408XXX, which
of course it will never be.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, March 13, 2007 12:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DST and VM timestamp
On Tue, Mar 13, 2007 at 10:44:08AM -0600,
Using an octal(8 T1 ports) card I have kept an average of 150
concurrent Zap channels open on a single server over 8 T1s. It's all a
matter of what the hardware will support.
Pure Zap channel conversations isn't always the limiter, what else are
you doing on this server?
MATT---
On 3/13/07,
From: Ricardo Carvalho [EMAIL PROTECTED]
Subject: [asterisk-users] How to match wild card inside a GoToIf?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't
Just how many SIP packets do you think it takes to set up a call?
Remember AUDIO IS NOT SIP! SIP is for call control, setup, and teardown.
Do a sip debug in the CLI and see just how many packets it takes to
setup a call.
Matt wrote:
That does sound low, especially if you have multiple
Is there any way to implement t38 in asterisk 1.2.15
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal Code: 1102-2080
Tel: (961) 1- 868 686
Fax :(961) 1-808 810
GSM: (961) 3-979 343
*
No
I went back to a simplified config. Although it sits at registering
now forever.. Can't dialout once it does give up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Connolly,
Tim
Sent: Tuesday, March 13, 2007 2:01 PM
To: Asterisk Users Mailing List -
[default]
exten = _220408XXX,1,Hangup
exten = _2,1,Macro(outcall,${EXTEN})
Ken Williams wrote:
I don't believe this will work. He wants it to goto if EXTEN =
220408235 or 220408743 or any other digits for the last 3 of the
extension block 220408xxx. When Asterisk processes both his
Hi Users, Administrators and Pavel Jezek,
You prefer chan_h323 from asterisk tree and it's of course that use channels
by tree is very good.
But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad.
And I work with chan_ooh323, that it's too from Digium and work good!
And I am
2007/3/13, Matt Florell [EMAIL PROTECTED]:
Using an octal(8 T1 ports) card I have kept an average of 150
concurrent Zap channels open on a single server over 8 T1s. It's all a
matter of what the hardware will support.
Pure Zap channel conversations isn't always the limiter, what else are
you
On 3/13/07, Héctor Maldonado [EMAIL PROTECTED] wrote:
2007/3/13, Matt Florell [EMAIL PROTECTED]:
Using an octal(8 T1 ports) card I have kept an average of 150
concurrent Zap channels open on a single server over 8 T1s. It's all a
matter of what the hardware will support.
Pure Zap channel
Hi,
i finally managed to get it work using GlobalVar.
I still have a question. I have several context in my
voicemail.conf like
[default]
[customer_1]
[customer_2]
[customer_3]
How can i set a different emailsubject for each
context?
thx
--- richard Coco [EMAIL PROTECTED] wrote:
Hi all,
I was doing VICIDIAL capacity testing on this server with a quad
processor(dual core) server, so the load was high and the call volume
was also very high, not a good comparison to what you are doing.
I have handled more than 100 concurrent channels before with two
quad-T1 cards in a singl P4
Hello everyone,
I have previously asked this question on the asterisk-video list, but I
got directed here.
I have a setup consisting of asterisknow beta4 (not sure if that is
crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the
local network. My computer has a USB-Camera
for 1.4 you have only two choices chan_h323 and chan_ooh323,
chan_oh323 from inaccessible networks, is death project, more than year
unmaintained,
I'm using chan_h323 both from 1.2 and 1.4 without problems (opposite
site to chan_h323 is ci$co gateway or callmanager)
also, chan_ooh323 isn't
Just how many SIP packets do you think it takes to set up a call?
Probably around 8 - 10 per call, excluding any ReINVITES, DTMF, etc.
INVITE, Authentication Required, ACK
INVITE w/AUTH INFO, TRYING, RINGING, OK
BYE, OK
--Luki
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Tzafrir Cohen escribió:
On Tue, Mar 13, 2007 at 06:25:37PM +0100, Germ�n Aracil Boned wrote:
My solution.
With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config.
Compile and without rtc module, load ztdummy. It work good with usbcore
and uhci_hcd modules.
I have
I'm helping with an article in New Scientist on the use of ISN
(http://www.freenum.org/) and the reporter with whom I'm working is
trying to get some quotes from users of normal ENUM services
(e164.arpa, please) from a telco perspective as a comparative basis.
If you run an SS7
Ok so let's go with 10. Now say you have a busy call-center behind a NAT
with 8 lines. That's 80 SIP messages. And if you have short calls, you
could easily exceed that, especially if you are placing calls on hold,
forwarding, etc.
On 3/13/07, Luki [EMAIL PROTECTED] wrote:
Just how many
--- Klaverstyn, David C [EMAIL PROTECTED] wrote:
Can you provide some specific details as I would like to implement
something like this.
I wrote this application a while ago for FreePBX, maybe it helps:
http://samyantoun.50webs.com/asterisk/callback/callback.gif
Check out Nerd Vittles at nerdvittles.com. There's an article on this
kind of scenario
On 3/14/07, Samy Antoun [EMAIL PROTECTED] wrote:
--- Klaverstyn, David C [EMAIL PROTECTED] wrote:
Can you provide some specific details as I would like to implement
something like this.
I wrote this
I dont think you can but you can use a variable. Have a look at
voicemail.conf. You can edit the message the asterisk sends out. If you want
the CID to be in the subject you can use the variable ${CALLERID(number)} .
- Original Message -
From: richard Coco [EMAIL PROTECTED]
To:
benedikt,
* 1.4 does no video codec translation... it is just a pass through so
using the same unit on both ends is a plus. you might try adding this
codec too.
allow=h264
I assume that audio is ok, just no video, right!?
there may be a nat problem, try
nat=no
I have some video experience
Does anybody know what DTMF coding does S101i adapter using?
I've been testing one for over a week and here are my observations:
- DTMF signaling is working perfectly with Asterisk, much better than
Sipura 3K
Though, I think the Asterisk iaxy firmware is buggy, the unit is using
auto-update
Ok, I'm going to have to lay out how we have this set up so you can
understand. :)
We're using a VOIP provider (ViaTalk) and have our main trunk
provisitioned with them. We also have 2 of what they call virtual
numbers ... we have one set up to do a SIP forward to [EMAIL PROTECTED]
and [EMAIL
Vernier Umali wrote:
Check out Nerd Vittles at nerdvittles.com. There's an article on this
kind of scenario
proprietary :P
-Stephen-
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I'm thinking about replacing my Birch T1 integrator with an Asterisk box.
The Integrator has 12 voice 768k data, so the Asterisk box would
become a router PBX.
Has anyone done anything similar, what experiences have you seen /or
read about.
TIA
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