ki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
--
Michael L. Young
(elguero)
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- On Feb 5, 2021, at 11:18 AM, Michael L. Young wrote:
> - On Feb 4, 2021, at 4:26 PM, Social Boh wrote:
>> The problem is with this CentOS 7 glibc version:
>> 2.17-317.el7
>> After the library update and system reboog,
>> gotoif Asterisk applicati
a 'yum history' and note the transaction ID of
the update. Then try running 'yum history undo [transaction id]'. That should
roll you back to the previous glibc.
Looks like Red Hat is already working on it:
https://access.redhat.com/solutions/5778071
--
Michael
owse/ASTERISK-26143 |
https://issues.asterisk.org/jira/browse/ASTERISK-26143 ]
Not sure if this is the answer to your problem but thought that I would throw
that out there.
Michael L. Young
(elguero)
--
_
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- Original Message -
> From: "sean darcy"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
>
> Sent: Tuesday, January 21, 2020 9:22:28 PM
> Subject: [asterisk-users] permission woes on systemd
[..]
> So why would starting asterisk as user asterisk work, but fail using
>
running? How are you starting Asterisk (init script /
systemd)?
--
Michael L. Young
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- On May 4, 2016, at 8:49 AM, Mamadou NGOM n...@numericap.com wrote:
> Hello everybody,
> When I call my extension the agi script don't work well. when I look at the
> cli,
> that is what I have:
> AGI Tx >> agi_request: **.php
> AGI Tx >> agi_channel: SIP/myprovider-0007
> AGI Tx
- Original Message -
From: cov...@ccs.covici.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 29, 2014 6:42:05 PM
Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available
* ASTERISK-23754 - [patch] Use
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 2:43:30 PM
Subject: [asterisk-users] Login by AMI ok, by AJAM fails
--
root@apbx:/tmp# curl
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 3:39:35 PM
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
You're right - but I tried username too and it fails. I can't
...@lists.digium.com on behalf of Michael L.
Young myo...@acsacc.com
Sent: Friday, May 16, 2014 4:16 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
Have you taken a look at the Wiki yet?
https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access
- Original Message -
From: Michael L. Young myo...@acsacc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 4:55:30 PM
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
- Original Message
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Thursday, March 27, 2014 12:55:21 AM
Subject: [asterisk-users] Security log format / content
I've noticed that the Asterisk (v11) security log captures attempts
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
Why don't you try with
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 16, 2014 4:17:53 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
I am curious why you would say
From: Tony Mountifield t...@softins.co.uk
To: asterisk-users@lists.digium.com
Sent: Friday, November 8, 2013 10:39:25 AM
Subject: [asterisk-users] 11.5.0 - SIP registration not retrying after
failures
I had a SIP problem on an 11.5.0 system that I look after. It
registers
with a
- Original Message -
From: Noah Engelberth nengelbe...@team-meta.net
I have an Asterisk 11.5 system, using SIP Realtime and operating as a
ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system
that has a SIP trunk connection to my Asterisk box. The NV 7100 has
a public
Quoting Mike Diehl mdiehlena...@gmail.com:
Is there a list somewhere?
There is a list by state here:
http://www.call811.com/state-specific.aspx
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- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Sent: Thursday, August 1, 2013 8:41:19 PM
Subject: [asterisk-users] External sip phones register with the servers IP...
We have just updated our office server to Asterisk 11.4.0 from
- Original Message -
From: Richard Mudgett rmudg...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 2, 2013 8:24:49 PM
Subject: Re: [asterisk-users] Playing a sound file during a call
On Thu, May 2, 2013
- Original Message -
From: Leandro Dardini ldard...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 26, 2013 5:28:22 AM
Subject: [asterisk-users] rtcachefriends and rtautoclear on change password
Hello
- Original Message -
From: Jaap Winius jwin...@umrk.nl
To: asterisk-users@lists.digium.com
Sent: Thursday, March 21, 2013 12:47:57 PM
Subject: [asterisk-users] Asterisk 1.8 and dual stack support
Hi folks,
Following an upgrade to Debian wheezy, I'm now running Asterisk
1.8.13.1.
- Original Message -
From: Jaap Winius jwin...@umrk.nl
To: asterisk-users@lists.digium.com
Sent: Thursday, March 21, 2013 5:27:37 PM
Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support
That's what I thought would happen. When I set bindaddr=:: and use
'netstat -lpn
- Original Message -
From: Bob Pierce westman...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: g...@westmancom.com
Sent: Monday, February 4, 2013 6:14:26 PM
Subject: [asterisk-users] Asterisk 1.8 Streaming MOH timing
- Original Message -
From: Logan Bibby lo...@keobi.com
Does anyone have a good contact for their sales? I've attempted
calling their Enterprise sales a few times and was just spun around
in circles. Having a sales rep I can just call would be awesome.
Logan,
We have an account
From: Carlos Alvarez car...@televolve.com
It may be too late for this, but in working with another RBOC who
didn't want to deal with Asterisk, I just asked what they do
support, and modified the headers sent by Asterisk to claim that it
was one of the devices on that list. Done.
Like
- Original Message -
From: Matthew J. Roth mr...@imminc.com
Your email documents the same experience we had years ago. It was
strange reading it and I was shocked that nothing has changed in that
much time. Asterisk will work with Verizon's IP trunking product,
but
they're trying
- Original Message -
From: Carlos Alvarez car...@televolve.com
Sounds like the same huge effort it takes to work with
Qwest/Centurylink, and in the long run we found it simply isn't
worth it. The few benefits of working with an RBOC are countered by
the many drawbacks of working
- Original Message -
From: Matthew J. Roth mr...@imminc.com
At least Verizon maintains a consistent customer experience. ; )
Overall, we've found the service to be reliable and stable, but when
there are problems or changes needed you're dealing with Verizon and
the
with Verizon?
Thanks,
--
Michael L. Young
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- Original Message -
From: Steven Howes steve-li...@geekinter.net
I *think* Verizon require IPSEC for the signalling, so it may be
worth reading up on configuring IPSEC in Linux (or acquiring
additional hardware) whilst you're looking at the Asterisk part.
This could have just been
- Original Message -
From: Joseph syscon...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Saturday, November 24, 2012 12:54:12 AM
Subject: [asterisk-users] * Waiting for asterisk to shutdown .
I'm running asterisk on a small box,
- Original Message -
From: Howard Leadmon how...@leadmon.net
To: asterisk-users@lists.digium.com
Sent: Saturday, November 24, 2012 3:19:10 PM
Subject: [asterisk-users] SIP Debugging Information..
I did a little googling, but didn't seem to find anything specific
to
answer the
- Original Message -
From: Felix Vazquez felix.vazq...@theboshgroup.com
To: asterisk-users@lists.digium.com
Sent: Friday, November 16, 2012 11:20:46 AM
Subject: [asterisk-users] Intruder
I am in the asterisk CLI and can see an unidentified caller trying
the make calls out of the
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 4:05:21 AM
Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object
Hi
I'm using 1.8.7.0. This morning I got an alert telling me
Asterisk
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 9:25:37 AM
Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
Thanks for
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 13, 2012 4:35:54 AM
Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber
session with ejabberd
On 11/13/2012 12:11 AM, Phil
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 7, 2012 9:20:58 AM
Subject: Re: [asterisk-users] 11.0.1: more sip registry woes
On 11/06/2012 09:45 PM, Michael L. Young wrote:
- Original Message
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 6, 2012 7:51:04 PM
Subject: [asterisk-users] 11.0.1: more sip registry woes
Upgrade to 11. This worked on 10.X.X
sip.conf:
- Original Message -
From: Ira i...@extrasensory.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 3, 2012 3:21:50 AM
Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
At 07:59 PM 10/2/2012,
of his prior email, not too many people may even be
affected by this change just like he won't be.
Michael L. Young
(elguero)
PS: If you can't tell, I am really for this change and doing so without any
configuration options
in.
Just my thoughts on the above concerns presented.
Michael L. Young
(elguero)
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are case sensitive.
If someone is moving from a GUI interface to CLI, then they would/should know
that case sensitivity is important and therefore the change shouldn't pose a
problem.
Just some thoughts in regards to the concerns brought up.
Michael L. Young
(elguero
- Original Message -
From: Thorsten Göllner t...@ovm-group.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 18, 2012 11:52:15 AM
Subject: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging
(mysql,
- Original Message -
From: Jayesh Labade jayesh.lab...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 25, 2012 2:09:58 AM
Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44
bytes file
- Original Message -
From: Jayesh Labade jayesh.lab...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 24, 2012 4:10:29 PM
Subject: [asterisk-users] Asterisk MixMonitor starts recording 44
bytes file
Hello
CSeq: 102 INVITE
Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250
Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO)
Content-Length: 0
I think the 404 Not Found being returned from the server is a clue as to what
the problem is.
Michael L. Young
(elguero
, that in the configuration files, the signaling option used is
opposite of what the module is.
Regards,
Michael Young
(elguero)
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- Original Message -
From: Mike Diehl mdi...@diehlnet.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2011 5:13:22 PM
Subject: Re: [asterisk-users] Polycoms rebooting themselves
Well, we've taken the time to check out the wiring. It's only 3
years old and
looks
- Original Message -
From: Chris Maciejewski ch...@wima.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 19, 2011 9:39:57 AM
Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to
satisfy
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 27, 2011 3:34:03 PM
Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
Friends,
We
--
Michael L. Young
Administrative Claim Service, Inc. | IT Manager
600 Main Street, Suite 5, Winchester, MA 01890
www.acsacc.com
Phone 781-721-1998
- Original Message -
From: Andrew Stewart astew...@notre1.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
that helps but it is worth a shot in mentioning to you.
Regards,
Michael Young
(elguero)
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Hello:
New to asterisk and hoping to use for http://summitcamp.org research
station.
While trying to use with Inphonex I find that incoming calls drop after
about one minute--
-- Got SIP response 420 Bad Extension back from 208.239.76.169
== Spawn extension (incoming-inphonex, 210, 1)
Quoting Tim Nelson tnel...@rockbochs.com:
Do you have any sort of site/mailing list/etc setup to facilitate
this group? I'd be interested in attending such a meetup in the
future.
http://www.tcaug.net/
Quoting Thczv F. Thczv thczv.th...@gmail.com:
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I
Quoting Fred Posner f...@teamforrest.com:
Starting around 10:00 AM EST.
All services from them whether I connect by IP or DNS (both east coast
and west). Anyone else?
Yes, I'm experiancing the same problem.
Their www.voicepulse.com and connect.voicepulse.com seem to be offline
as well.
International numbers are variable length, so the timeout applies for those.
North American National numbers are a fixed length.
Generally, the phone company will collect 7, 10 or 11 digits for North
American numbers.
For example, I live in Minneapolis, MN.
My number is 612-xxx-.
I have
Quoting Gustavo A Gonzalez [EMAIL PROTECTED]:
I am looking for a sample sip configuration of a SIP provider that runs
Broadsoft VoIP switch.
This is what I use:
register = 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368
[broadworks]
type=peer
host=1.2.3.5
dtmfmode=rfc2833
Quoting Doug Lytle [EMAIL PROTECTED]:
C F wrote:
Then there is basicly no way to do this besides for cracking it? I
Not that I am aware of, no. This subject went around several years
back. They also talk about brute forcing the password as well. As far
as I recall, nobody came back
saw this problem
while running CentOS 5.1 XEN kernel and if you search their bug tracking
system you will see some reports about this bug. A search on google
revealed some possible solutions.
This was the first thought that came to my mind when I saw this.
Regards,
Michael L. Young
(elguero
;Include genzaptelconf configs
;#include zapata-auto.conf
;Include AMP configs
;#include zapata_additional.conf
Kind Regards,
Richard Young
Intrintech Limited
[EMAIL PROTECTED]
111 Cannon Street
London
EC4N 5AR
Phone: 0845 644 2918
All orders placed or confirmed via email
It would be possible if Asterisk sent a remote-party-id back to the
calling phone.
Polycom and Sipura phones (possibly Cisco phones) Support this with
SIP on Broadworks and it works great.
--Shane
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
It is not possible to do this the way
. I never investigated further.
--Shane
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
Have you ever actually done this with Asterisk?
Shane Young wrote:
It would be possible if Asterisk sent a remote-party-id back to the
calling phone.
Polycom and Sipura phones (possibly Cisco
It's all priced by quantity of each feature you license, number of
users, number of concurrent calls, things like that.
Previously it only ran on Solaris. It now also runs on Linux.
I wasn't involved with our initial purchase, but I couldn't imagine
you could have a working system for less
Quoting C F [EMAIL PROTECTED]:
Anybody here having any problems with nufone?
Calls are not going thru, when trying to call their customer service
number it doesn't go thru.
When trying to resolve www.nufone.net I get (sourec:
http://www.dnsstuff.com/tools/lookup.ch?name=nufone.nettype=A ):
Quoting Curt Shaffer [EMAIL PROTECTED]:
I wanted to see if there was anything reasonable in price out there yet that
performed an RF to IP bridge via asterisk. What I mean by this is callers
from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is
an option available for the
.
Michael L. Young
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of François Delawarde
Sent: Monday, May 14, 2007 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel huge irq problem
Hello,
I had noticed
Quoting Savoy, Kevin - Williston, ND [EMAIL PROTECTED]:
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
your requirement don't really make sense.
to try again:
speex and/or gsm
- put into /etc/init.d/___ - phone enabled on boot up
Huh? IS that phone a client program? If so: why should it be run as a
server?
There are plenty of ways to run a
is strictly prohibited. If you are not the intended recipient (or
authorized to receive for the recipient), please notify the sender by reply
e-mail and delete, or destroy all copies of this message immediately.
-Original Message-
From: Shane Young [mailto:[EMAIL PROTECTED]
Sent: Monday, February 19
phase 1 requirements:
- sip and/or iax2 using g729 and/or gsm
- put into /etc/init.d/___ - phone enabled on boot up
- all parameters in /etc/___
- automatically navigate around gnome and kde sound
- automatically navigate dhcp (if any)
- gnu has something sort of close(?)
- must install through
in your dialplan:
[context]
...
h,1,AGI(...)
David Ruggles [EMAIL PROTECTED] wrote: I'm working on writing some test IVR
code in AGI. I can't get my FXO port to
detect a hang-up, but I'm going to deploying this using Digital cards so I
decided to just skip that problem for now. However this
please send me more info
thanks!
Tim Panton [EMAIL PROTECTED] wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can
Need to deploy between 50 to 300 lightweight Linux - only browser and softphone.
Any recomendations?
-
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have a Grandstream and SJPhone SIP phones going to asterisk.
with SJPhone (on Linux) getting. any ideas??
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.100;branch=z9hG4bKc0a80264001045c3c2c52331d4920678;received=24.10.123.39;rport=60754
From: sip:[EMAIL
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I have to be able to add my own context.
from API use Command to run the CLI command add extension
-
The fish are biting.
Get more visitors on your site using Yahoo!
On a SIP phone is it possible to enter the dialplan when the user picks up the
phone without having to wait for the user to press an extension?
Is is possible to do something like
[sip-test]
s,1,Answer
s,2,Playback(welcome)
s,3,WaitExten(30)
1,1,Noop(exten 1)
...
t,1,Goto[s,2]
snip
zaptel.conf
---
loadzone=uk
defaultzone=uk
span=1,1,1,ccs,hdb3,crc4,yellow
span=2,0,1,ccs,hdb3,crc4,yellow
bchan=1-15,32-46
dchan=16,47
bchan=17-31,48-62
---
where span 1 is to the provider and span
its notransfer=yes in iax.conf not transfer=no :)
this is getting close!
however, it takes about SEVEN seconds after the called party hangs up
before the next priority is executed - same as with the T option.
as contrast to h option, when called party hits asterisk, the next
priority is almost
[EMAIL PROTECTED] wrote:
On Tuesday 16 January 2007 2:31 pm, chester c young wrote:
however, it takes about SEVEN seconds after the called party hangs
up
before the next priority is executed - same as with the T option.
What kind of last leg are these calls? to POTS (even CAS T1) or
PRI
$25 to openvpn.org - thanks to Anselm Martin Hoffmeister
--- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young:
the answer sucks, but is apparently correct.
If your application involves the caller (e.g. an employee of your
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
When the calling channel hangs up, Asterisk will send the remaining
leg of the call to exten =
Silly question: how are the calls going out? If they're going out
through an analog line without the ability to detect hang-ups, then,
that's the problem.
calls are coming in and out thru an Asterisk server using iax2. have
tried two different DID providers and have same problem.
--- Paul [EMAIL PROTECTED] wrote:
Anselm Martin Hoffmeister wrote:
Curious - is this still a $50 thread?
yes.
Never miss an email again!
Yahoo! Toolbar alerts you the instant new Mail arrives.
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
extensions.conf:
[general]
static=yes
writeprotect=no
--- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
cannot make Dial(...,,g) work correctly, although Dial(...,,gh)
works
just fine. (to make matters worse, it does seem to work sometimes,
although once working
--- bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
To create the symbolic link, I read in the documenation that I have
to type this command:
# ln -s /usr/src/'uname -r' /usr/src/linux-2.4
1) What it means by 'uname -r'?
2) Why I have to create such symbolic link to do pointing for
bridge-1.2.12.1.patch, there are other
ones that say trunk, obviously only work with the trunk version of
Asterisk.
Kind Regards
On 1/3/07, chester c young [EMAIL PROTECTED] wrote:
(my pstn calls are coming in thru an upstream asterisk server, so
the
called and calling phone number
after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status. is this possible?
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
use a simple agi - php is easy to do.
--- O.Kamal [EMAIL PROTECTED] wrote:
I just need to retrieve a value from a field in a postgres database,
and
playback this value when someone dial a specific extension.
On 1/4/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
O.Kamal wrote:
I need to
be the primary timing source. The other span should
either be at 0 (not used as a timing source) or set as a secondary timing
source.
Hope this helps.
Michael L. Young
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(my pstn calls are coming in thru an upstream asterisk server, so the
called and calling phone number is passed as an extension.)
when caller comes in on 555, he will go to extension 1234 where he
will wait for the API to make a call to 999 for him. how do I
bridge the two calls?
-1.2.12.1.patch, there are other
ones that say trunk, obviously only work with the trunk version of
Asterisk.
Kind Regards
On 1/3/07, chester c young [EMAIL PROTECTED] wrote:
(my pstn calls are coming in thru an upstream asterisk server, so
the
called and calling phone number is passed
does anyone know if ztdummy is requires under 1.6 or are they using
Linux' rtc?
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how is this fitting into 1.4?
- can it be compiled against 1.4 or only 1.2?
- if not, are there leanings in that direction?
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--- Mark Greene [EMAIL PROTECTED] wrote:
Hey guys,
In your experience what is the best way to go for a production
asterisk box in your offices?
(In the US) I have had very good luck with Opterons in Tyson rackmounts
bought from Newegg.
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Do
Dial(...|30|g) does not seem to work
whereas
Dial(...|30|gh) works just fine
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and some steps in Asterisk for reducing
echo: http://www.xorcom.com/pdfs/AB007_Echo.pdf
Michael L. Young
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Hi All,
I have a Linksys SPA-3000 [Hardware version 3.0.0(1178), Software
version 3.1.10(GWd)], with both the FXO and FXS interfaces
registering with asterisk via SIP seperatley. I also have a
Cisco 7940 and 7960 using the sccp2 (chan_sccp) driver, and a
couple of IAX softphones
Both inbound
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