Am 18.12.11 20:19, schrieb Carlos Rojas:
> Hello everybody
>
> I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine?
>
> I've been find any information and saw heatbeat + cysnc2 and heartbeat +
> rdbd, any one has worked any these aplications fine?
>
>
> Best regards
Hello,
Hi Sammy,
Actually we have 2 voip trunk at our server 1 of *Voipon* and 2nd of *
Gradwell*. When our balance goes down then they don't auto-refill it, I
don't know the reason behind it.
Ans some time goes down means Call will not go through from VoIP trunk.
So want to make a script in AMI / AGI
Thank you.
*José Pablo Méndez
*
On Sun, Dec 18, 2011 at 8:23 PM, Kevin P. Fleming wrote:
> On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote:
>
> Embarrassingly enough, I just tried the nat=no again both in the
>> general and peer sections and the blessed phone registered My
>
Is it possible to load some parts of the extensions.conf file via static
realtime?
For example,
extensions.conf
[some_context]
#include abc.conf
extconfig.conf
abc.conf => mysql,asterisk,ast_config
So far, I wasn't able to get it going - Asterisk crashes at startup. Maybe
abc.conf doe
On Sat, Dec 17, 2011 at 03:55:55PM -0600, Warren Selby wrote:
> Why not try set a variable under each device in sip.conf to the same as the
> endpoint name then Dial(SIP/${CustomVar})?
Thanks. That keeps the device-extension association in one place, at the
cost of a little redundancy if done ma
Hello,
Do you saw this solution?
http://linuxnotes.us/
Regards
On Sun, Dec 18, 2011 at 12:26 AM, virendra bhati wrote:
> Hi List,
>
> I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip
> trunk for making outgoing and DID for incoming to server.
>
> My question is how
On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote:
Embarrassingly enough, I just tried the nat=no again both in the
general and peer sections and the blessed phone registered My
apologies, again, I wrote the thread late at night probably this blinded me.
No problem, we've all done that
Thanks for answering Kevin.
I guess my eyes were tired the night I started this thread, and yes, it
would be ridiculous that Cisco phones couldn´t do rport. I actually found
that its not the rport parameter, but the UDP ports usage. nat=no receives
the REGISTER with source port 5400 for example, a
Hello everybody
I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine?
I've been find any information and saw heatbeat + cysnc2 and heartbeat +
rdbd, any one has worked any these aplications fine?
Best regards
--
On 12/18/2011 01:42 AM, José Pablo Méndez Soto wrote:
I have been testing with Cisco phones and have been able to register
them with new firmware 9.2.1 (7911/7945/7970). All worked until I
realized that from version 1.8.7.2, the VIA header contains the rport
parameter, which breaks the phone reg
On 12/18/2011 06:43 AM, Andreas Sikkema wrote:
On 12/18/11 12:55 AM, William Scott wrote:
Notice there is no "rtpmap:18 G729/8000" in the reply.
The call continues fine.
Is it right that there is no codec info in the reply and the call continues?
The value for 18 is defined in some RFC as be
On Sat, Dec 17, 2011 at 6:16 AM, Carlos Rojas wrote:
> Hello
> Did you use callerid(num) in your dial plan?
>
Thanks Carlos it worked after looking at the link ->
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
Thanks and Regards,
Kaushal
--
__
Hi List,
Which will be the appropriate variable to get called peer IP address?
I tried following channel variables
peerip, recvip, URI, from
and following SIP channel variables:
SIPURI,SIPDOMAIN
They all return calling peer IP but not the destination/called peer IP.
unfortunately set(CDR(calle
On 12/18/11 12:55 AM, William Scott wrote:
> Notice there is no "rtpmap:18 G729/8000" in the reply.
>
> The call continues fine.
>
> Is it right that there is no codec info in the reply and the call continues?
The value for 18 is defined in some RFC as being G729/8000 so there's no
real need to
Hi Warren
According to the book I'm using as well as the documentation on Asterisk 1.8
you should remove the musiconhold.conf file from /etc/asterisk if you want to
read it from a database. From what you say it looks as if you can have both.
Maybe not with the same classes?
You are right about
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