-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Can you specify a bit more what you whant to have help with!
Best regards
jan
Sharon wrote:
hello,
can someone help me with ser redirect to asterisk.
any help appreciated.
Thanks,
AA
___
Vic a écrit :
Hi, Zoa,
yes, these calls are from SIP to SIP. We will have more than 3000
(more like 5000)concurrent calls come into system and we will need to
handle them.
What exactly do you do with these calls?
We will also need an IVR function as well.
I am not up to speed on
Several of my SIP users are in the habit of diverting all their calls to
an assistant when they're out of the office. When these calls ring on
the assistant's phone, she wants to be able to tell which number they've
been forwarded from so that she can say Joe Blow's phone or whatever
when she
Sure enough we lost ALL sip-sip audio on 1-25
Pulled my hair out for hours before looking here or at the website
to find this problem reported...
Very greatful to find this I have upgraded to 1.2.3 but
still have no sip-sip audio!
what?!
Now I'm back to contnued hair pulling what culd I
Jan Saell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Can you specify a bit more what you whant to have help with!
I guess it is the usual question nobody wants to answer, right?
(Internet) == port 5060 = SER redirect EVERYTHING to port
5062 = Asterisk
bye
On 1/28/06, Kevin Bockman [EMAIL PROTECTED] wrote:
Joe wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
When an agent receives a call, they will be marked busy anyways as long
as you are using agent
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
I plan on setting up agent extensions (if possible via macro) something like
this for example:
exten =
On 1/29/06, BJ Weschke [EMAIL PROTECTED] wrote:
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
I plan on setting up agent extensions (if possible via macro)
To check if it's the same problem, set your system clock back 2 weeks.
If it gets better, then the upgrade didn't take. If it doesn't get
better, it's something else.
--Rob
-Original Message-
Very greatful to find this I have upgraded to 1.2.3 but
still have no sip-sip audio!
Hello,
i want to use asterisk as a ZAP-FXO / SIP gateway.
It works fine when I use a SIP provider and register my Asterisk as client
there - incoming calls are routed to an extension in a specified context.
What I want to do now is to not use the SIP provider and make asterisk accept
calls
On 11:25, Sun 29 Jan 06, Peter Molnar wrote:
Hello,
i want to use asterisk as a ZAP-FXO / SIP gateway.
It works fine when I use a SIP provider and register my Asterisk as client
there - incoming calls are routed to an extension in a specified context.
What I want to do now is to not
I try to set the username to something useful, like Peter, but it
remains the value of 621
1. I set username in the record of name = 621 to Peter Pan
2. I search for this record and found it is set: name=621 and
username=Peter
3. I go to Asterisk CLI sip show peers and find the
I got some troubles with my wifi phone.
I used to have set it to:
Proxy server:
Proxy IP: sip.elmit.com
Port: 5060
Expire time 1200
Outbound proxy
Proxy IP fwdnat.pulver.com
Port:: 5082
User account
Phone: 610
Username: 610
User Pwd:
since a while this
Hi, i'm trying to install a compatible modem to act as a X100P, and i
would appreciate some help here, this is what is hapening when i
modprobe zaptel:
# modprobe zaptel
FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format
firstly, i did:
Hi all,
look at these lines.
I created a queue named info when a caller (extension
86) place a call he is put on queue he sould hear MOH
.
What's the meaning of :
Jan 29 14:35:30 WARNING[2591]: file.c:509
ast_openstream_full: File 100 does not exist in any
format
Jan 29 14:35:30 WARNING[2591]:
Hi, i'm trying to install a compatible modem to act as a X100P, and i
would appreciate some help here, this is what is hapening when i
modprobe zaptel:
# modprobe zaptel
FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586
-up-1GB/misc/zaptel.ko): Invalid module format
firstly, i did:
On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote:
[snip]
If you do, honestly, need to handle 5k calls, you’d probably have to
have a bank of Cisco 5850s doing the termination
Or have a look at the Lucent APX8100 box for some added carrier class
humpf. Supports more than 8000 DS0's (channels)
Hi everybody,
Every time callers reach my FXO
port,asterisk producesone ring tone just beforeit executes
Answer(). How to remove this?
I have commented "#define RINGBEGIN" on
zconfig.h, but it does not help.
Thanks in advance for your
help.
Cheers,
Anto
It is waiting for the CalledID information. Set
usecallerid=no and that should do it for you.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto
RachmadSent: Sunday, January 29, 2006 9:40 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] How
Thanks Alexander,
Ijusttried that, but
itdoesn't help. There is still one ring tone produced before asterisk
executes Answer(). And thereis nocaller ID being forwarded to the
destination channel, which actually I need. That is why I have usecallerid set
to yes.
Cheers,
Anto
-
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Sunday, January 29, 2006 7:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout
question
Anto,
Callerid delays answer until after the
first ring, I would suggest you are either not subscribing to your telco for
caller id or similar.
The advice you got was correct.
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad
Sent:
vpbx*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
621/621192.168.250.76 D N 5060 OK (65 ms)
626/626192.168.250.109 D N 5060 OK (180 ms)
616/Ronald Softphone (Unspecified)D
Thanks a lot Dean,
I think thereis a way to remove that
ring tone and also still have the caller ID from the incoming call. I have been
trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find
that. Could you pleaseletme know which part of the codes handling
that?
Hi Anto, I dont know as I use
[EMAIL PROTECTED] these days as so much easier.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad
Sent: Sunday, 29 January 2006
10:40 AM
To: Asterisk Users Mailing List -
Non-Commercial
You are wrong, there is no way you can remove the ring, since the ring
is something that the callers equipment is generating to the caller,
and NOT asterisk. The most you will able to accomplish will be to have
just one ring before asterisk picks up. By setting usecallerid to no
all you are doing
On Sun, Jan 29, 2006 at 01:46:32PM +, Maxi Belino wrote:
Hi, i'm trying to install a compatible modem to act as a X100P, and i would
appreciate some help here, this is what is hapening when i modprobe zaptel:
# modprobe zaptel
FATAL: Error inserting zaptel
Hello CF,
I thought that asterisk generated that first ring tone. I didn't think further,
especially about what the caller's switching centre is doing when it gets an
instruction to reach my number. You are obviously right. That switch will
notify the caller (alerting) as soon as it gets a
hi
i just setup a test with asterisk 1.2 to see how many concurrent
calls it could handle, and I came across something quite strange;
with ~1000 calls between two asterisk servers, generated with
[looptest]
exten = _X.,1,GotoIf($[ ${EXTEN} 1000 ]?pickup:dial)
exten = _X.,n(pickup),Answer
BJ Weschke wrote:
On 1/28/06, Kevin Bockman [EMAIL PROTECTED] wrote:
Joe wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
When an agent receives a call, they will be marked busy anyways as long
as you
hi
i'm setting up a rig to handle quite a few SIP clients, so i need a
way to simulate, say, 20k SIP ATAs. Does anyone know how? This should
of course be as close as possible to 'reality', meaning n% calls per
client and the usual REGISTER/OPTION traffic.
thanks
Best regards
Roy Sigurd
To handle 5000
calls coming in over a PRI, youd need 210 or so T1s or 170
E1s.All of those would
generate 320Mega BYTES of data per second (eg, 32Gigabit/sec)[Wai Wu] He not talking about
PRI here, but rather SIP to SIP
There is no way
possible that youre going to pump that
Set up another * and use the manager api to make lots of calls to the other
one. You can even make hundresd calls at a time.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd
Karlsbakk
Sent: Sunday, January 29, 2006 1:19 PM
To: Asterisk
I took a look at the asterisk-1.2.3 Makefile, seems to me that the
WARNING is just a list of all the .so files found in the modules
directory that aren't also found in a subdirectory, it isn't checking
that they were built with the current version. So it's going to
complain about the modules
Hi all,
Im trying to set up my asterisk server, but Im
having a few problems.
My server is running with a public IP address.
When I want to set up a call with a softphone in my
private network behind a router Im always having an error message.
In the CLI session we get a
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings, echo
canceller is less then ideal on long analog pstn loops, etc.
Anyone
sure, but I need to simulate the SIP REGISTER and OPTION traffic sent
by ATAs as well.
Best regards
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
---
In space, loud sounds, like explosions, are even louder because there
is no air to get in the way.
On Jan 29, 2006, at 7:26 PM, Wai Wu
yes, i do have /boot/config
so then? what should i do?
thanks again.
Maxi2006/1/29, Tzafrir Cohen [EMAIL PROTECTED]:
On Sun, Jan 29, 2006 at 01:46:32PM +, Maxi Belino wrote: Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is
Yep, tried that.
blew away all my source code, re-downloaded re compiled and re installed.
it's behaving exactly the same, calls go through but no audio in either
direction for sip-sip calls on the LAN or to-from the Internet SIP
providers tested.
I'm at a loss I feel like I have tried
At 07:09 AM 01/29/2006, you wrote:
I just tried that, but it
doesn't help. There is still one ring tone produced before asterisk
executes Answer(). And there is no caller ID being forwarded to the
destination channel, which actually I need. That is why I have
usecallerid set to yes.
I added a
On Sun, 2006-01-29 at 12:36 -0600, Rich Adamson wrote:
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings, echo
canceller is
Yep, tried that.
blew away all my source code, re-downloaded re compiled and re installed.
it's behaving exactly the same, calls go through but no audio in either
direction for sip-sip calls on the LAN or to-from the Internet SIP
providers tested.
I'm at a loss I feel like I have tried
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings, echo
canceller is less then ideal on long analog pstn loops, etc.
Just received a new Cisco 7960 (not refurb, but brand new) and it won't
tftp the initial config file (OS79XX.TXT) from an FC3 box. The 7960 does
get an appropriate dhcp response including the tftp address.
Using a sniffer, I see the tftp request being sent from the 7960 to the
FC3 box, but the
So, if anybody is interested, i think i got it!
i executed in the linux kernel source directory 'menu xconfig'
and the graphical config windows appears (nicer to me)
then in processor type i changed from PentiumPro (why was this value
here?) to K6, but after recompiling zaptel and modprobing:
On Sun, 2006-01-29 at 13:24 -0600, Rich Adamson wrote:
Seems the spa3k functions pretty well (had a few since they first came
out), but the echo can on long analog loops leaves some to be desired
as well. Short loops seem to work just fine.
Thanks for the information. Sounds encouraging.
The
No Firewalls involved, the test has been simplified down to two sip phones
on a LAN and still no audio.
For waht it's worth IAX2 still works fine.
Steve
-
Yep, tried that.
blew away all my source code, re-downloaded re compiled and re
installed.
it's behaving exactly the same,
On Jan 29, 2006, at 11:24 AM, Rich Adamson wrote:
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings,
echo
canceller is
Have you tried increasing the debug level and watching the cli?
No Firewalls involved, the test has been simplified down to two sip phones
on a LAN and still no audio.
For waht it's worth IAX2 still works fine.
Steve
-
Yep, tried that.
On Jan 29, 2006, at 10:30 AM, Warren Burstein wrote:
I took a look at the asterisk-1.2.3 Makefile, seems to me that the
WARNING is just a list of all the .so files found in the modules
directory that aren't also found in a subdirectory, it isn't checking
that they were built with the current
I got some troubles with my wifi phone.
What phone is this?
Nabeel
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Outbound proxy
Proxy IP stun01.sipphone.com
Port:: 3478
STUN servers are not outbound SIP proxies.
Nabeel
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On 13:09, Sun 29 Jan 06, Martin Joseph wrote:
I removed the following to get it starting up again:
app_enumlookup.so
app_groupcount.so
app_md5.so
app_txtcidname.so
func_cut.so
Both the README and the UPGRADE listed that those functions
became obsolete and were replaced by dialplan
Disregard... should have been smarter and looked at the wiki. Damn!
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
Just received a new Cisco 7960 (not refurb, but brand new) and it won't
tftp the initial config file (OS79XX.TXT) from an FC3
Anyone used a Cisco VG200 as FXO gateway for * ?
-Dan
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Nabeel Jafferali wrote:
I got some troubles with my wifi phone.
What phone is this?
pulver phone
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I would like to setup Asterisk as
follows:
When users make inter-office calls they can dial
the extensions, however if they want to make an external call, that they enter a
code on their phone, before they external call can go through.
We would like to give each user an access code,
this
Throw it in the trash now. There's next to no support for these. No
firmware upgrades. The are VERY SLOOW in responding to network
calls too.
All in all not a very astute purchase. I should know; I've had 5 of them.
I use the UTStarcom F1000 currently. Much better but still not good.
On 17:26, Sun 29 Jan 06, Mark Phillips wrote:
Throw it in the trash now. There's next to no support for these. No
firmware upgrades. The are VERY SLOOW in responding to network
calls too.
All in all not a very astute purchase. I should know; I've had 5 of them.
I use the
On Sun, 2006-01-29 at 18:00 -0400, Dakota wrote:
I would like to setup Asterisk as follows:
When users make inter-office calls they can dial the extensions,
however if they want to make an external call, that they enter a code
on their phone, before they external call can go through.
We
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273
Mark Phillips wrote:
Throw it in the trash now. There's next to no support for these. No
firmware upgrades. The are VERY SLOOW in responding to network
calls too.
Thanks for your suggestion, but it still did not explain how to set-up!
I figured out that if I set outbound proxy same
Dakota a écrit :
I would like to setup Asterisk as follows:
When users make inter-office calls they can dial the extensions,
however if they want to make an external call, that they enter a code
on their phone, before they external call can go through.
We would like to give each user an
Or you can use authenticate() and have it take its 'passwords' form a
text file on your machine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter aka Bret McDanel
Sent: Sunday, January 29, 2006 5:37 PM
To: Asterisk Users Mailing List -
On Sun, 2006-01-22 at 18:18 +0100, Wilson Pickett wrote:
You could also use a trick like *21* going to a new context and
waiting for digits (with a slighly longer timeout) and have it
trigger on the longest possible number.
perhaps if local extension were of the form 2nnn or 2nn and you want
In my agi-debug i get the following error-message:
AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority:
I have the same problem with all (shell) AGIs.
Not sure when it started (about two days ago) and why, i tried to restart
asterisk and my server and also reinstalling
We started out useing SPA2k but they were prone to stop talking to
the ethernet. OK after reboot for awhile but cannot keep going to
customer sites and rebooting things.
switched to spa2001 and somewhat better but they keept losing
registrations and then could not talk to them remotely.
Hello Roy,
Have you heard of Sipp? http://sipp.sourceforge.net/. I am pretty
sure it can do what you desire.
Also a commercial tool from Empirix, Hammer NXT.
(http://www.empirix.com/default.asp?action=articleID=64)
Cheers,
Omar
On 1/29/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
Carsten Bock wrote:
In my agi-debug i get the following error-message:
AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority:
Oups there something missing, the complete error message is
AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority:
Unable to set
Dakota wrote:
I would like to setup Asterisk as follows:
When users make inter-office calls they can dial the extensions,
however if they want to make an external call, that they enter a code
on their phone, before they external call can go through.
We would like to give each user an
Jean-Michel Hiver wrote:
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
Cheers,
Jean-Michel.
I have been running several asterisk xen servers for a few months now.
Problems would depend on what
Can I get some more information on this?
Are there any drawbacks?
- Original Message -
From: Alexander Lopez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 29, 2006 6:58 PM
Subject: RE:
Drawbacks are few in my opinion. The onlys issue is that users will hear
'Please enter your password They get three attempts and if they do not
enter it right the system goes to priority + 101.
Example:
Exten = _91X.,1,Authenticate(/etc/asterisk/ldusers.txt)
Exten =
Mark - The new UTStarCom F3000 should be shipping soon. I have done a bit
of preliminary testing and it seems to work very well.
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
-
On Sun, 2006-01-29 at 19:58 -0500, Alexander Lopez wrote:
Drawbacks are few in my opinion. The onlys issue is that users will hear
'Please enter your password They get three attempts and if they do not
enter it right the system goes to priority + 101.
An other drawback in my opinion is one I
I was searching thru the internet and I found a wide variety of different
web interfaces for asterisks
I was curious which one is best suited for asterisks. Thanks
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I
would set up the softphone on a public address and see if it works first. How do
you set up the sip.conf for the softphone?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
SoFieSent: Sunday, January 29, 2006 1:41 PMTo:
I see. But are you going to setup a few thousand entries in the sip.conf, one
for each of ATA?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd
Karlsbakk
Sent: Sunday, January 29, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial
Aryanto Rachmad wrote:
Thanks a lot Dean,
I think there is a way to remove that ring tone and also still have the
caller ID from the incoming call. I have been trying to find that on
zaptel.c,
chan_zap.c and pbx.c, but I could not find that. Could you please let me
know which part of the
On Jan 29, 2006, at 1:24 PM, Michiel van Baak wrote:
On 13:09, Sun 29 Jan 06, Martin Joseph wrote:
I removed the following to get it starting up again:
app_enumlookup.so
app_groupcount.so
app_md5.so
app_txtcidname.so
func_cut.so
Both the README and the UPGRADE listed that those functions
AMP hands down is STILL the best... though a few are catching up quickly
On Mon, 2006-01-30 at 01:29 +, Strain Jer wrote:
I was searching thru the internet and I found a wide variety of different
web interfaces for asterisks
I was curious which one is best suited for asterisks. Thanks
On Mon, 2006-01-30 at 10:42 +0800, Lists wrote:
AMP hands down is STILL the best... though a few are catching up quickly
The best non-web interface (I had considered copying it into a
webinterface) is the cocoa app for the mac asterisk stuff. There is a
reason you dont really see questions
Yes I have.
I have been battling this issue since wednesday 1-25
And so far have tried many things.
Have also tried RTP debug and do not see ANY RTP when the call is made.
I will keep working at this until I figure it out but right now am very
stumped and frusterated.
The software update SHOULD
Steve:
I'm picking up the tail end of a thread, so apologies if this is offtrack...
Have you perhaps got an old set of EXECUTABLES in your path, that are
being picked up before your newly compiled ones?
Roger
Steve Gladden wrote:
Yes I have.
I have been battling this issue since wednesday
Hi everyone,
www.dialogictrader.com
dialogic and general voip hardware forum
This is
no way a plug but I wanted some user feedback on a site I had put together
which allows people that use voip and dialogic hardware to come together. They do
not necessarily have to be used together but
I have a snom 320 connected to an Asterisk server. I do some weird things
using the AccountCode as an identifier. When the snom makes a call, the
AccountCode is used successfully in the dialplan as a variable
${ACCOUNTCODE}.
When that same call is transferred using the button on the snom, I see a
Have you verified that you are actually
sending sound over the RTP streams?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
Sent: Friday, January 27, 2006
11:13 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Hi Sir,
My problem is when I click on pricelist, i have an error there's
something wrong on the pricelist database.
When I looked at the database and search for a table called pricelist
there's nothing there. I foolowed the querires on the the structure but
also found any query that creates
Signate sells a single server that can get
you to the call volumes you need.
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vic
Sent: Saturday, January 28, 2006
7:16 PM
To:
asterisk-users@lists.digium.com
Can anyone think of a way to dial 2 different numbers at the same time,
but set the callerID number differently for each channel?
The application is a simultaneous ring of an office extension and a cell
phone where the user wants to know that the call to the cell phone is a
redirected call from
You can then call up the macro like this:
[extensions]
Exten = 1120,1,Macro(call-cell,SIP/120,Local/[EMAIL PROTECTED])
Exten - 2120,1,Macro(call-cell,SIP/120)
[macro-call-cell]
Exten = s,1,Dial($ARG1ARG2)
[cellulars]
Exten = 120,1,Set(CALLERID(num)=551212)
Exten =
This doesn't really belong on the asterisk-users list. ASTPP has it's
own mailing list. This can be found @ www.astpp.org. I, or someone
else will be happy to help you either there or on the forums. On your
1st post please mention what version of ASTPP you are using.
Thanks,
Darren
I use VMWare, but will start testing XEN...I use VMWare to slice up some
nice big servers to provide dedicated hosted PBXes. We also use the VMs
for easy deployment and is a vital part of our DR Plan...
Now, we are full VoIP...not T1 or PRI cards...
-Original Message-
From: Paul
On Fri, Jan 27, 2006 at 03:20:16PM -0600, Dan Littlejohn wrote:
I was confused about the modules.
Got this warning when upgrading to 1.2.3 even when using the most
current asterisk-addons and even svn asterisk-addons.
WARNING WARNING WARNING
Your Asterisk modules directory, located
On Fri, Jan 27, 2006 at 04:03:23PM -0600, Joseph Tanner wrote:
Quick and dirty solution:
mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak
And just as a reminder: those are basically exactly the problems package
management systems are here to solve.
--
Tzafrir Cohen |
On Sun, Jan 29, 2006 at 06:54:05PM +, Maxi Belino wrote:
yes, i do have /boot/config
so then? what should i do?
if you build a custom kernel, no problems. But if you use a distro
kernel, you should point the makefile to the directory where your kernel
source is (or at least: suffucuent
On Sun, Jan 29, 2006 at 05:16:28PM -0800, trixter aka Bret McDanel wrote:
An other drawback in my opinion is one I originally said in my email
that I was corrected on ... That its a flat text file which limits
dynamic passwords, in that you have to either write something that will
allow
Jean-Michel Hiver wrote:
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
We run a number of systems with Xen, its great once you figured out the
nags of it :)
Remember, to do anything with
Roy,
Wai Wu wrote:
sure, but I need to simulate the SIP REGISTER and OPTION traffic sent
by ATAs as well.
What is the current registration time you accept on the servers ? 3600
?? One thing you can do to try this is set a number of devices to a much
shorter registration period. This
Ronald Wiplinger wrote:
Dakota wrote:
I would like to setup Asterisk as follows:
When users make inter-office calls they can dial the extensions,
however if they want to make an external call, that they enter a code
on their phone, before they external call can go through.
We would like
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