I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
overseas IP connection, and somehow SIP seemed to work better.
Peter
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Hello,
I have been testing a trunk IAX and another SIP, using sipp to
generate SIP calls to a Asterisk box.
The testing dialplan just connects to another Asterisk box, who
answers the call and playback some files.
I noticed that the cpu load is higher when I use an IAX, about 90% for
25 simulta
I have no NAT
issues. My PBX is multihomed and the outside IP is locked down for all
except IAX and SIP ports.
With the current
version of asterisk, which transport is better right now?
I am looking at 6-10
simultaneous calls over a half T1.
I am not asking
about codecs here, I am as
> I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> overseas IP connection, and somehow SIP seemed to work better.
>
> Peter
>
Then try making two or three or more calls at the same time.. :)
If you setup IAX in trunk mode it uses the same connection for multiple voice str
Does this thread help?
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote:
> I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> overseas IP connection, and somehow SIP seemed to work better.
>
On Fri, 19 Sep 2003, WipeOut . wrote:
> > I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> > overseas IP connection, and somehow SIP seemed to work better.
> >
> > Peter
> >
>
> Then try making two or three or more calls at the same time.. :)
>
> If you setup IAX in tr
How do you set up IAX in Trunk mode?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: Friday, September 19, 2003 3:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX vs SIP
> I wonder how IAX compares to SIP bandwidth-w
> How do you set up IAX in Trunk mode?
> Uriel
>
Add "trunk=yes" to your definition in iax.conf..
Later
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>
> FYI: trunking only works in IAX2 and it requires you to have a zaptel
> interface on both endpoints
>
I have heard that but in my setup I only have Zaptel hardware on one side and trunking
appears to work fine..
Initially I tried using ztdummy on the side which didn't have zaptel hardware
Hello,
On 19-09 19:48, WipeOut . wrote:
> Also IAX does not care about NAT so a situation like..
> AST<-->NAT<-->Internet<-->NAT<-->AST
> ..will work fine.. SIP will have problems in a setup like this without the use of
> specialised NAT routers..
I am wondering how setup like this could work w
>
> I am wondering how setup like this could work with IAX (or any other
> protocol) when symmetric NATs are used.
>
> If you have two different NATs then direct connection is not possible
> between hosts behind those two NATs. You have to do some kind of
> provisioning of the NAT boxes (i.e
> Does this thread help?
>
> http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
>
Thanks, this is exactly what I was looking for. I tried experimenting with
different codecs myself, and GSM seems to be the only one that works...
neither iLBC or Speex went thru. I'm using XLite
> Thanks, this is exactly what I was looking for. I tried experimenting with
> different codecs myself, and GSM seems to be the only one that works...
> neither iLBC or Speex went thru. I'm using XLite v1.x & Asterisk 0.5.0,
> wonder if it's a softphone's problem?
>
I have got X-Lite to work with
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote:
> Hello,
>
> I have been testing a trunk IAX and another SIP, using sipp to
> generate SIP calls to a Asterisk box.
>
>
> The testing dialplan just connects to another Asterisk box, who
> answers the call and playback some files.
>
> I noticed that t
Hi,
Just to review the test I did:
---SIP extension-- Trunk
-
| SIPp |<--->| Asterisk 1 |<>|
Asterisk 2 |
------
--
Both Asterisk b
We used iax for more than a year and moved to sip about 6 months
ago. The quality from termination providers seems much better now with sip.
Tom
At 09:38 PM 8/30/2006, you wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary="_=_NextPart_
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros a
We've been using iax with teliax.com for a couple of years, and it seems
the quality of calls varies with time. Sometimes it is good and next
time its not so good. There has been changes occurring to iax and the
jitterbuffer stuff over the last two years, and I'm reasonably certain
that some po
Does IAX support music on hold? It seems only my SIP phones do. Is this
correct?
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I hope I didnt get your question wrong, but if you are asking whether
Asterisk can play MOH to an IAX client, then the answer is yes. We
have a couple of IAX clients connecting into the queue and are being
played MOH while waiting for an operations.
Hope this helps :)
Cheers
On Tue, 29 Mar 2005
[EMAIL PROTECTED] wrote:
Does IAX support music on hold? It seems only my SIP phones do. Is this
correct?
As I understand it, once the call is delivered to asterisk, it becomes
abstracted into a channel. And you can do anything to one channel that
you can do to other channels ( with a few nota
In order to work around some authentication issues I am considering
connecting two asterisk boxes with IAX instead of SIP. The original
reason for choosing SIP was to reduce the need to translate SIP
signaling to IAX, since all origination, termination, and UAs are SIP.
Can anyone comment on th
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with sip phones on several different servers and
I
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with si
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
> On Thu, 4 Jan 2007, Noah Miller wrote:
>
> >Hi Damon -
> >
> >>Can anyone comment on the overhead added when a SIP call comes into one
> >>asterisk box, is routed to another with IAX instead of SIP, and is then
> >>sent
> >>to th
Brad Templeton wrote:
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
Unless bandwidth between the * servers is a concern, then you're better
off keeping the link between servers as IAX. (preferably trunked)
It is worth rememberin
Unless bandwidth between the * servers is a concern, then you're better
off keeping the link between servers as IAX. (preferably trunked)
As I understand it video will NOT work if you use an IAX trunk between
* boxes, it must be SIP. Just food for thought in case you are
planning on using video.
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote:
> Brad Templeton wrote:
>
> >
> >For SIP phone calling * box, relay to other * box and out to SIP
> >phone, you definitely want SIP all the way.
> >
> Unless bandwidth between the * servers is a concern, then you're better
> off keepi
Brad Templeton wrote:
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote:
Brad Templeton wrote:
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
Unless bandwidth between the * servers is a concern, then you're better
off
If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.
I have had great success getting this to work using IA
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