How is your system configured?
Debug output of faild faxes?
This kind of information is needed to help you!
Regards,
Juan
khalid touati wrote:
can anyone help me out in this, a big number of my faxes
are lost everyday! i would really appreciate any help on how i can
tweak asterisk (rxfax)
Try using sendrpid=yes on sip.conf
Regards,
Juan
Ondrej Valousek wrote:
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on his phone.
Note that name of A gets displayed on
or directrtp set
to yes can help you.
Saludos,
Juan E. Rodríguez
-Original Message-
From: Kenneth Noisewater noisewater...@gmail.com
Date: Thu, 1 Apr 2010 16:50:47
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Connection Question
trunk you can have as much calls as *
supports.
Saludos,
Juan E. Rodríguez
-Original Message-
From: Dr. Kenneth Noisewater noisewater...@gmail.com
Date: Thu, 01 Apr 2010 19:35:54
To: jerdg...@gmail.com; Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
/mailman/listinfo/asterisk-users
Saludos,
Juan E. Rodríguez
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When you say 'a2billing' won't pass the number, you mean you are calling to an
IVR or something like that.
And when did you dial you destination number twice???
Saludos,
Juan E. Rodríguez
-Original Message-
From: Nathanial Allan nathanial.al...@gmail.com
Date: Tue, 30 Mar 2010 13:08
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Saludos,
Juan E. Rodríguez
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New
Try setting canreinvite and nat to no for those extensions.
Saludos,
Juan E. Rodríguez
-Original Message-
From: Alyed al...@vivoxie.com
Date: Fri, 26 Mar 2010 10:56:50
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Saludos,
Juan E. Rodríguez
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Johann:
Do you know how is the SMS sent over the IP, does it use SIP INFO
message or somthing like that?
Regards,
Juan
Johann Steinwendtner wrote:
randulo schrieb:
2009/10/9 "Juan E. Rodrguez" jerdg...@gmail.com:
Does any one know about a SIP hard phone capable
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wrote:
What do you mean I should use a global function. I'm kind both well versed
and a newb to *
James Shigley
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E.
Rodríguez
Sent: Monday
A2billing forum has a lot of information and questions are answered
very fast. Try searching on the forum before posting, cause the answer
may be there already.
forum.asterisk2billing.org/
Regards,
Juan
Bruce Nik wrote:
Hi Sucan,
A2Billing doesn't have a mailing list but you may ask
Is ddwhome defined in global context?? If so, then you should use global
function.
Paste asterisk log to check.
Saludos,
Juan E. Rodríguez
-Original Message-
From: James A. Shigley j...@answeringserv.com
Date: Mon, 28 Dec 2009 12:11:35
To: Asterisk Users Mailing List - Non-Commercial
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Saludos,
Juan E. Rodríguez
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Juan E. Rodríguez
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Right now, I think it does not.
Look out for it at: asterisk-...@lists.digium.com
Regards,
Juan
Khaled W Chehab wrote:
Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP)
protocol ?
And how to integrate
Regards
Khaled
Chehab
NGN Eng.
Hello,
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.
As I see here, you do not have to include the big10 context inside the
default context, as you have an extension defined to reach that context
and its extention is start extension.
If the cleveland-IVR is based on the start extension too, the same
applies.
Besides that, it would
If you already mangle packets with IPTABLES, then you should comment
the line[s] tos_* on sip.conf.
Regards,
Juan
Bart Fisher wrote:
I don't understand this message:
[2009-10-29 16:31:51]
WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184
From what I
Hello,
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.
If the trunk is a dynamic IP you need the other end to register to
Asterisk, so letting Asterisk know the new IP.
Regards,
Juan
B.Masoud @ SH wrote:
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP
Check out ClarkConnect and SmoothWall.
Regards,
Juan
Steve Totaro wrote:
On Tue, Oct 13, 2009 at 2:41 PM, SIP s...@arcdiv.com
wrote:
David Wathen wrote:
Hi,
My customer has a outdated firewall that is also presenting a NAT
nightmare for getting the Asterisk server
A2billing (Star2Billing, I think, for commercial support) is a good
choice and it's very mature software.
Astercc is very fast and has a very good callshop solution.
Regards,
Juan
voip crazy wrote:
Hello all,
I want to instal a Billing solution in the same asterisk's box. I have
browse for
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
Or at least with J2ME support, to run a little program?
Regards,
Juan
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Use CALLERID(name).
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
Steve Totaro wrote:
On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote:
Hi List
I've a CID lookup hooked onto an inbound route (i m using trixbox) it
runs well but it returns the value as
Maybe you could use the Asterisk Database.
In 1.4 you can do it with DBGet and DBPut:
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput
In 1.6 use DB() function.
Regards,
Juan
David Backeberg wrote:
On Mon, Jul 6, 2009 at 11:37 AM,
that starts the AMI instance will override the values
making it unusable for what I want to achieve.
It was really useful anyways. :)
2009/7/6 Juan E. Rodríguez jerdg...@gmail.com
mailto:jerdg...@gmail.com
Maybe you could use the Asterisk Database.
In 1.4 you can do it with DBGet and DBPut
Try running your script with /usr/bin/php5 script.php to test it
Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q
Leah Newmark wrote:
Thanks.
I didn't change anything in my dialplan. I am aware of reloading configuration
:)
My AGIs are copied from a working asterisk install -- the
I do, I am planning to have little more than 1000. Right now I had
managed little more than 700 SIP channels + 100 IAX channels.
Do you think this can cause any problem?? --I mean, having this RTP
ports range--
Tzafrir Cohen wrote:
On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez
Neal:
Try having on sip.conf:
srvlookup=no
Regards,
Juan
[EMAIL PROTECTED] wrote:
Hello,
Thanks for your replies.
We checked our sip.conf and we have canreinvite=no already. I agree
it could be a firmware issue. I will get another vendors phone hooked
up to the pbx before going crazy
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