Hi,
What I did is first to dig a bit into the app_dial.c. I saw how the
ANSWEREDTIME variable
is created (end_time - answer_time). Then I added some lines to export the
answer_time variable
as a channel variable. I added these lines right after the answer_time
decleration (line 1426 in ver 1.4.4
Hi
I've two boxes connected via IAX2 Trunk were working fine from few days
suddenly today one box is got crashed with this message
2007-06-03 12:25:37 WARNING[26511]: chan_sip.c:2612 sip_write: Can't send
4113608 type frames with SIP write
my version of * is 1.2.14 on FC4
thanks
arun
_
In short, the 's' extension is not a catch-all.
The use of 's' can be confusing. The best example I have of the use of
's' is when a ZAP call comes in on an analog line. IIRC, the book says
something to the effect that 's' is for when, upon arrival in a
context, the call has no other place to g
HI
Im getting strange message on asterisk console
WARNING[26853]: app_queue.c:2321 try_calling: Announcement file
'custom/announce-adslsetupnatrate' is unavailable, continuing anyway...
thanks
arun
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Hi,
> I setup auto dial on my asterisk server. The problem
> is asterisk does not wait for called party to answer
> the call but proceed to process the extension specifed
> in my .call file
No problem with Auto Call
> exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try c
Hi,
> I setup auto dial on my asterisk server. The problem
> is asterisk does not wait for called party to answer
> the call but proceed to process the extension specifed
> in my .call file
No problem with Auto Call
> exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try c
And you don't find that sufficiently self-explanatory?
On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote:
> HI
>
> Im getting strange message on asterisk console
>
> WARNING[26853]: app_queue.c:2321 try_calling: Announcement file
> 'custom/announce-adslsetupnatrate' is unavailable, continuing
Hi
my * box is giving me these warning and b'coz of second warning line my
agents are not able to hear the announcement in the queue some time it
happen many time
2007-06-03 13:40:30 WARNING[28016]: chan_sip.c:2612 sip_write: Can't send
4113568 type frames with SIP write
2007-06-03 13:40:30 WAR
Hi
sorry for not details. when ever I see this message on * console my agents
are not able to listen to announcement.
thanks
arun
On 6/3/07, Mattt <[EMAIL PROTECTED]> wrote:
And you don't find that sufficiently self-explanatory?
On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote:
HI
Im g
Hi,
Steve Totaro írta:
> It sounds like the Meridian GSM devices connect via analog
> line so if you connected the ports on the Meridian currently
> terminating into the GSM devices go into asterisk via a four
> port FXS (guessing the GSM devices supply dialtone) then the
> routing in the Meridian
Hi,
Does any in the Czech republic had an issue with Telifonica blocking port 5060
? I have a client there that is unable to register with our servers in the US.
Thanks.
Dovid
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asterisk-use
If your getting an error about invalid Octet key or something make
sure you are doing the load from a "Windows Computer"
it's been noted that a lot of people have had problems loading that
version from various OS's. I went through a few computers
before it took the load (for the record it w
Thank you.
Seems the rtptimeout did the trick.
Sphinx2 doesn't always manage to work out what i'm saying
but when it does, everything works.
Thanks again.
-
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for
On Sun, Jun 03, 2007 at 02:50:17AM -0700, Khaled Chehab wrote:
> I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
> ... error appears >
>
> >From where I can get the missing rpms .or kernel source
you need kernel-devel or kernel-smp-devel, depending on your running
kerne
I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there
is someone did that, Please in need to have the installation procedure step
by step. Its too urgent for me .
Thanks alot
Regards
*
No employee or agent is authorized t
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: Sunday, June 03, 2007 8:22 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Centos kernel source
>
> On Sun, Jun 03, 2007 at 02:50:17AM -07
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Khaled Chehab
> Sent: Sunday, June 03, 2007 6:58 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc: [EMAIL PROTECTED]
> Subject: [asterisk-users] zaptel on CENTOS s
On Sun, Jun 03, 2007 at 09:10:19AM -0400, Steve Totaro wrote:
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Khaled Chehab
> > Sent: Sunday, June 03, 2007 6:58 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussi
Steve Totaro wrote:
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
>> Sent: Sunday, June 03, 2007 8:22 AM
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Centos kernel source
>>
>> On Sun, Jun
I already did what you said,please see the log results in zaptel.rar
attached when I compile zapltel using
make
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without exp
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: Sunday, June 03, 2007 9:22 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] zaptel on CENTOS servercd
>
> On Sun, Jun 03, 2007 at 09:10:19A
I already did what you said,please see the results when you compile using
make
ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type
/usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencing pointer to
incomplete type
/usr/src/zaptel-1.4.1/pciradio.c:1535: error: dere
It would be good if you did not trip the part of "I already did what you
said." I am not sure what you did.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Khaled Chehab
You already did what WHO said? What did you already do?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Khaled Chehab
> Sent: Sunday, June 03, 2007 8:08 AM
> To: 'Asteris
On Sun, Jun 03, 2007 at 10:14:10AM -0400, Steve Totaro wrote:
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> > Sent: Sunday, June 03, 2007 9:22 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [aste
On Sun, Jun 03, 2007 at 05:07:49AM -0700, Khaled Chehab wrote:
> I already did what you said,please see the results when you compile using
>
> make
>
>
>
> ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type
> /usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencin
Thanks for the detailed procedure. One update:
On Sun, Jun 03, 2007 at 03:31:53PM +0200, Philipp Kempgen wrote:
> Here's what I did:
>
> yum update
>
> reboot
>
> yum install kernel-devel-`uname -r`
>
> yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \
> libtool make automake
Understood, it is not the "catch all" but, what if I am designing a server
that needs to accept calls from 15 or more 1800 numbers?
How would you now channel it to a "catch all"?
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Sunday,
Tzafrir Cohen wrote:
> Thanks for the detailed procedure. One update:
>
> On Sun, Jun 03, 2007 at 03:31:53PM +0200, Philipp Kempgen wrote:
>
>> Here's what I did:
>>
>> yum update
>>
>> reboot
>>
>> yum install kernel-devel-`uname -r`
>>
>> yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++
Rob Schall wrote:
> Are you able to access the phone via a web browser? And did asterisk
> register the phone? If both are true and you set the always reboot flag
> to 1, then rebooted the phone by hand, there shouldn't be anything
> standing in the way.
It seems that it will only reboot if certai
Hi:
[EMAIL PROTECTED] wrote:
> Hi All,
>
> I setup auto dial on my asterisk server. The problem
> is asterisk does not wait for called party to answer
> the call but proceed to process the extension specifed
> in my .call file
This will not work with this channel driver. Explanation follows.
>
Ricardo Martins wrote:
> Thinking this way, I invite those who think about the open source
> communities just as a "zero price" and its mailing lists as a space to
> wait passively for answers, to rethink its own ideas. Before asking for
> something and adding "trash" to communities mailing lists,
Gordon Henderson wrote:
> On Fri, 1 Jun 2007, Gavin Henry wrote:
>
>> Dear all,
>>
>> I think this is common, or at least how it is supposed to be, but
>> whening dialing over a ZAP channel, it's taking around 5~ seconds to
>> ring on the over end, likewise inbound.
>>
>> This is just with a norma
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk trunk.
I tried the latest patch which failed in many
Hi Everybody,
I'm experiencing this kind of issue.
One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel
card. Everything seems to work but sometimes the third party caller when
listening to MOH listens some "SSH!" instead of MOH, this is not
continuos, MOH plays ok for,
exten => _X.,1,
On 6/3/07, BSumrall <[EMAIL PROTECTED]> wrote:
Understood, it is not the "catch all" but, what if I am designing a server
that needs to accept calls from 15 or more 1800 numbers?
How would you now channel it to a "catch all"?
Brad
-Original Message-
From: [EMAIL PROTE
- Original Message -
From: "Mauro Zanin" <[EMAIL PROTECTED]>
To:
Sent: Sunday, June 03, 2007 12:32 PM
Subject: [asterisk-users] Loud noise instead of MOH
Hi Everybody,
I'm experiencing this kind of issue.
One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel
card.
Hi
I have FC6 system in the office running SVN-trunk-r63567
It is behind a NAT router which I have configured to do port forwarding etc.
Asterisk connects and registers correctly to my SIP service (Sipgate.co.uk)
and I can make and receive calls from any SIP phone on the office LAN.
The problem
Hello I just settup a realtime mysql table for sip_peers. All peers
(friends) is autenticateing but when i want to initiate a call between them
i got the following error. Someone have some ideea? Thank you.
--
pbx*CLI>console dial 1014
== Console is full duplex
-- Executing [EMAIL PROTE
On Sun, 3 Jun 2007, Ian Clough wrote:
The problem comes when I try to use a SIP phone at home (also behind a
NAT router). The phone registers correctly and I can see the SIP OPTONS
packets being sent to the phone (SNOM 190). I can see an OK reply being
received by Asterisk (using SIP DEBUG).
I've tested a few different wifi SIP phones for office/factory use, and
generally have been underwhelmed. Before I grab another few and test, I'd
like to ask around here about the candidates.
My requirements are relatively simple:
- WEP/PSK should be supported WITHOUT dragging the phone down
-
Stephen Bosch wrote:
[outbound-reminder]
exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)
Zaptel considers the call answered the moment it is bridged. You need
call progress detection of some kind. Are you using an analog card?
This is not fully correct.
Calls going out FXO signaled ports are cons
To catch "1 or more of any character" you would use _.
To catch "2 or more digits" you would use _X.
You almost NEVER EVER EVER want to use _. as it will also match
extensions like "s", "a", "o", and "h".
Tom Lynn wrote:
exten => _X.,1,
On 6/3/07, BSumrall <[EMAIL PROTECTED]> wrote:
Under
BSumrall wrote:
Understood, it is not the "catch all" but, what if I am designing a server
that needs to accept calls from 15 or more 1800 numbers?
You put all 15 or more 800 numbers in as specific individual extensions
just like most every one else does.
If your numbers have a pattern like
Did you look at this one?
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
We're thinking of buying a couple for communication between our IT team
members across 5 floors in 2 buildings.
If you've tried it I'd be interested to
On Sun, 3 Jun 2007, Alex Crow wrote:
Did you look at this one?
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a UT StarCom F1000G ...
I have one of those phones and it's not very good - certianly can't handle
roa
Khaled Chehab wrote:
I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there
is someone did that, Please in need to have the installation procedure step
by step. Its too urgent for me .
Thanks alot
These scripts are working pretty well for me. They are based on a
scr
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
> No frills, specs look good, price seems excellent!
> http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen is pretty much useless,
the buttons are *TINY*, the battery life horrib
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Saturday, June 02, 2007 11:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto Dial Problem
[EMAIL PROT
I have a 2 sangoma cards that need to be configured on a same server, one is a
T1 and another is a for PSTN line. Is this possible, if so please help.
Regards,
Sanjay Rajdev
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asterisk-users mai
Firefox under Linux seems to have loaded it in ok.
No browser under OSX worked (PPC or Intel), including Camino, Firefox,
or Safari.
On 6/3/07, Bryan Laird <[EMAIL PROTECTED]> wrote:
If your getting an error about invalid Octet key or something make
sure you are doing the load from a "Windows Co
Sangoma offers EXCELLENT Technical support
Why not try them first?
They have never failed me yet, even for our peculiar requirements
Email address on their website
John Novack
Sanjay Rajdev wrote:
I have a 2 sangoma cards that need to be configured on a same server, one is a
T1 and another is
Hi Gordon
So, mexuar solution was that java softphone that you talked about?
Any other small softphone type solution around, something on the same lines
of what you described, something that the user could download but could be
preconfigured or passed parameters to so they user wont have to mess
True, maybe I didnt make myself clear on that point, what i meant was, Im
not looking for an app that would let people click a sip: URL type to
make a call using their already installed softphone but rather allow any
user that visits our website to click on something and either open a web
softp
Thank you for the explanation Dean, you are right on the money and could be
more precise.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Sábado, 02 de Junio de 2007 04:34 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-u
On 6/3/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: 03 June 2007 18:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Options Reply Ignored
On Sun, 3 Jun 2007, Ian Clough wrote:
> The proble
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