the realtime peer and
create a peer in sip.conf this works !!
On Tue, May 3, 2011 at 11:15 AM, Justin Case
nogoodnameswereavaila...@gmail.com wrote:
On Mon, May 2, 2011 at 1:09 PM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
I have
On Tue, May 3, 2011 at 2:50 AM, Ernie Dunbar maill...@lightspeed.ca wrote:
I'm kind of at a loss to diagnose problems like this, yet we get them a lot.
- The ATA (Thomson 784 in this particular case) is logged into the
Asterisk server. 'sip show peer' shows their IP address, port, and
On Mon, May 2, 2011 at 1:09 PM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
I have context=defcontext set in sip.conf. For each peer I have
context=outcontext in the peer definition since I want outgoing calls
from registered SIP peers
It's not as easy as getting a switch and buying and selling minutes. You
have to learn what issues there are out there, day to day problems etc. You
will want to learn the protocols that you will be using (SIP/H.323 etc.). I
have two carriers that know less about SIP than I do. Before you jump in
Sure. If you write the dial plan correctly and your legacy PBX supports it.
On Mon, Jul 12, 2010 at 7:31 AM, Malvin Rito
mr...@mail.altcladding.com.phwrote:
Hi List,
We’re planning to use Asterisk as our backend PBX for our legacy PBX
where-in received calls from legacy PBX can be
What would the power have to do with the D Channel ? Isn't which channel
used a logical setting (as opposed to physical). I am not saying your wrong
I am just trying to understand why it happens.
On Mon, Jul 12, 2010 at 7:56 AM, C F shma...@gmail.com wrote:
I have found that sometimes shutting
On Wed, May 5, 2010 at 7:10 PM, Adrian Marsh adrian.ma...@ubiquisys.comwrote:
Anyone have any experience with a Japanese local VoIP termination
supplier?
I’ve emailed a few companies looking to setup some PSTN to SIP and SIP to
PSTN termination, but no luck so far.
Thanks,
Adrian
On Thu, Nov 13, 2008 at 9:56 AM, Matt Gibson [EMAIL PROTECTED]wrote:
Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who has some used to
sell or point me in the direction of a seller.
Hi
You don't want to remove the INBOX folder. Also just rm will want you to hit
y to confirm. Try doing rm -rf
/var/spool/asterisk/voicemail/default/100/INBOX/*.*
On Fri, Jul 4, 2008 at 2:57 PM, voip crazy [EMAIL PROTECTED] wrote:
Hello,
I want to create an script which remove all the old
Hi List,
I made the mistake of having auto payments via PayPal. Just had some one put
in payments and have them all denied. So far this person send in funds from:
julie tosh - [EMAIL PROTECTED]
David Somerville - [EMAIL PROTECTED]
Gaetane Fortier - [EMAIL PROTECTED]
ray stewart - [EMAIL
Tell me when to stop laughing. Multiple channels and unlimited minutes ? No
sane person will give that to you.
On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I have a budget to work with and was wondering if there are any folks
providing SIP/IAX2 trunking for
What comes up in the Asterisk CLI ? Set debug and verbosity to 9 and see
what comes up. Also it can be a NAT issue ? Have Asterisk register every 3-4
minutes.
On Dec 24, 2007 4:00 PM, Jaap Winius [EMAIL PROTECTED] wrote:
Hi all,
Perhaps someone here could help me with this. I'm new to
The Snom 360 has worked the best for me as far as bridging two calls from
the phone. As far as quality of sound I would go with Polycom. I have the
601 but the 501 should do just as well.
On Dec 24, 2007 4:57 PM, Dean Collins [EMAIL PROTECTED] wrote:
Hey Chris,
As you know I'm not an Asterisk
Hi,
I am looking for some one to make a test call for me to a toll free number
in Australia (from a land line in Australia) and to a toll free number in
Argentina (from land line in Argentina). The numbers are set to echo test at
the moment.
Thanks.
/J
I am having the same issues when asterisk gets a call and then sends it to
an Avaya system. Anyone have an idea as to what would be causing it ?
On Nov 12, 2007 3:03 PM, Mark Bell [EMAIL PROTECTED] wrote:
Hey Guys,
I have something that just started happening. When my users call each
other
I have the same issue and I cant fix it :(
On Nov 21, 2007 9:56 PM, Vincent [EMAIL PROTECTED] wrote:
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
Hi,
I am using OpenSer + Asterisk. I am using a Audiocode MP112 over a
satellite link. The ping time to the server is about 700ms. When
connecting to another carrier there is no delay what so ever. When I
connect it to my test server there is a 3 second delay. From what I
heard my test carrier is
If I am behind the computer I end up just working. I need to get away and
read the book. Only way I will really learn ;)
On 10/7/07, Steve Totaro [EMAIL PROTECTED] wrote:
Justin Case wrote:
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
Anyone know of any other books that are worth reading ?
Thanks.
Justin
___
--Bandwidth and Colocation
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