Question:
If I am IAX trunking between 2 Asterisk instances, and ultimately connecting
to SIP endpoints on BOTH ends of the call, can I let the ENDPOINTS do ALL
the jitterbuffering, or must the iax-trunk do its own jitterbuffering?
I'm asking because I'm ignorant to the nuanced MECHANICS of the
On Mon, 2009-10-19 at 08:02 -0500,
asterisk-users-requ...@lists.digium.com wrote:
George Farris wrote:
I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well. On that machine I have a SIP phone. I have configured a
netgear wgt634u with asterisk and a SIP phone and
Hi all,
I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well. On that machine I have a SIP phone. I have configured a
netgear wgt634u with asterisk and a SIP phone and linked the two systems
together via IAX. Audio from Ubuntu to netgear is not bad, audio from
George Farris wrote:
I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well. On that machine I have a SIP phone. I have configured a
netgear wgt634u with asterisk and a SIP phone and linked the two systems
together via IAX. Audio from Ubuntu to netgear is not
On Thu, Sep 11, 2008 at 8:10 PM, C. Chad Wallace
[EMAIL PROTECTED] wrote:
At 8:29 AM on 11 Sep 2008, John Millican wrote:
Not directly on-topic for this list, but I'd not heard of OpenSIPS
before, so I had a look at the website. It looks to be a fork of
OpenSER. Does that mean OpenSER
On Tue, Sep 9, 2008 at 3:34 PM, Darren Sessions [EMAIL PROTECTED] wrote:
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this particular application.
An OpenSIPS solution will take care of your traveler's NAT issues (and could
handle the registrations) while you
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this
particular application.
If the users in question are often in hotels abroad, something like this may
not solve the problem - I've noticed quite a few hotels are now blocking SIP
traffic (presumably so as to
Chris Bagnall wrote:
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this
particular application.
If the users in question are often in hotels abroad, something like this may
not solve the problem - I've noticed quite a few hotels are now blocking SIP
On 9 Sep 2008, at 20:19, Mattias Andersson wrote:
Hi all!
I am looking for some software that would work as a proxy between a
SIP device (SIP phones and ATA boxes) and the Asterisk system
running IAX. The reason is that I can only get IAX trow the
firewalls, and can't change the
At 8:29 AM on 11 Sep 2008, John Millican wrote:
Not directly on-topic for this list, but I'd not heard of OpenSIPS
before, so I had a look at the website. It looks to be a fork of
OpenSER. Does that mean OpenSER development has slowed/ceased, or
has the OpenSER project itself morphed
Hi all!
I am looking for some software that would work as a proxy between a SIP
device (SIP phones and ATA boxes) and the Asterisk system running IAX. The
reason is that I can only get IAX trow the firewalls, and can't change the
settings.
One solution I am using are getting several Asterisk
I would suggest using OpenSIPS with Asterisk and bypass IAX all
together for this particular application.
An OpenSIPS solution will take care of your traveler's NAT issues (and
could handle the registrations) while you used Asterisk for voicemail
and whatever else.
I've personally used
Hi all,
I've been googling for a solution here and haven't really come up with
anything yet. We're doing an Asterisk install for a local radio station,
and we're looking for a phone that they can use in their control room
hooked up to their mixer board for recording calls. So, when you phone
in
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 17, 2008 3:45:56 PM (GMT) Europe/London
Subject: [asterisk-users] Sip or IAX device with professional balanced audio out
Hi all,
I've been googling for a solution here and haven't really come up
Peter Gradwell wrote:
mmm, but as you've seen, some customers like using multiple codecs. The
cisco kit is able to support a raft of options - and it does transcoding
very nicely - so the optimum solution is to have the cisco + customer's
asterisk agree on the same codec, and then have our
Hi All,
Had anyone got an idea of there exists an LDAP backend for SIP and IAX?
I've read that there is a patch for LDAP realtime, but I hadn't seen any
type of
relevant configuration information.
Any information on the above would be highly appreciated.
Regards,
Nir S
]
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 12, 2006 9:20:29 AM GMT-0600 US/Central
Subject: [asterisk-users] SIP and IAX configuration from LDAP
Hi All,
Had anyone got an idea of there exists an LDAP backend for SIP and IAX?
I’ve read that there is a patch for LDAP realtime
On 9 Oct 2006, at 11:49, raviprakash sunkara wrote:
Hello Users.
I'm in Dilemma with the performance on SIP and IAX
Can any one help ...
1) Difference between the SIP and IAX...
which one is Best... in VOIP service
I'm using only SIP protocol for my
Hello Users.I'm in Dilemma with the performance on SIP and IAXCan any one help ... 1) Difference between the SIP and IAX... which one is Best... in VOIP service
I'm using only SIP protocol for my VOIP in OpenSER...And Also I using Asterisk in SIPwe can Communicate the SIP and IAX by
I have two boxes on the net that support local phones
in two offices.
I am not using any VOIP providers. Just local TDM04B cards then IAX
between offices.
I experience between offices drop outs, half way conversations things
like that.
Is that normal for asterisk to asterisk?
I have two 3
Jerry Geis wrote:
I have two boxes on the net that support local phones in two offices.
I am not using any VOIP providers. Just local TDM04B cards then IAX
between offices.
I experience between offices drop outs, half way conversations things
like that.
Is that normal for asterisk to
I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.
I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.
If you know of a project
I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.
I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.
If you know of a project
hi
We take calls inbound via SIP from our Cisco PSTN gateways, and pass it
to customers using IAX (they run their own asterisk servers).
We've noticed that asterisk is transcoding the call into a different
codec, if the customer prefers a codec different to that which our cisco
gw prefers.
Hi Peter,
I don't see any codec allow=blah statements. If your end user has
something like
[gradwell]
disallow=all
allow=gsm
Then you'll be forced to send them a GSM coded call.
Why not force the codec at your end by only supporting one? If the
customer then transcodes the call when it gets
Hi Peter, as one of your customers I would ask you not to dissallow g729
on IAX2 as we currently use it extensively.
Bails
Mark Phillips wrote:
Hi Peter,
I don't see any codec allow=blah statements. If your end user has
something like
[gradwell]
disallow=all
allow=gsm
Then you'll be
Mark Phillips wrote:
Hi Peter,
I don't see any codec allow=blah statements. If your end user has
something like
[gradwell]
disallow=all
allow=gsm
Then you'll be forced to send them a GSM coded call.
Why not force the codec at your end by only supporting one? If the
customer then transcodes
yeah yusYu Safin [EMAIL PROTECTED] wrote:
On 10/19/05, Steve Totaro <[EMAIL PROTECTED]>wrote: YES - Original Message - From: "Frank Kostin" <[EMAIL PROTECTED]> To: <ASTERISK-USERS@LISTS.DIGIUM.COM> Sent: Wednesday, October 19, 2005 8:58 AM Subject: [Aste
Hello everybody,
Is it possible to route any incoming SIP call
(without authentication - register) from an Asterisk A
to a remote Asterisk B(throught IAX2), transparently ?
Otherwise said, I would like to pass any incoming SIP
call from Asterisk A to Asterisk B without SIP need to
be registered,
YES
- Original Message -
From: Frank Kostin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 19, 2005 8:58 AM
Subject: [Asterisk-Users] SIP to IAX
Hello everybody,
Is it possible to route any incoming SIP call
(without authentication - register) from
On 10/19/05, Steve Totaro [EMAIL PROTECTED] wrote:
YES
- Original Message -
From: Frank Kostin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 19, 2005 8:58 AM
Subject: [Asterisk-Users] SIP to IAX
Hello everybody,
Is it possible to route any
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one call at a time ?
Thanks
Sandeep
Sandeep A.S wrote:
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
I can't say anything based on experience, but guessing that IAX2
trunking will
On Thursday 02 June 2005 02:48, Sandeep A.S wrote:
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one
Hello,
Could someone give me clues where to figure out this problem?
If I call from a Sip client to an Firefly client running IAX, the call
connects fine, no problems.
I can connect to asterisk using any codec (excepting g.729) on firefly
to voicemail and music-on-hold, other sip extensions and
use ethereal or iax2 debug to see what capabilities are been set in your
NEW message
Ernie Ankele wrote:
Hello,
Could someone give me clues where to figure out this problem?
If I call from a Sip client to an Firefly client running IAX, the call
connects fine, no problems.
I can connect to
On a sip to iax :
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 0ms SCall: 19170 DCall: 1 [xx.xxx.xxx.xxx:20406]
FORMAT : 4
-- Call accepted by xx.xxx.xxx.xxx (format ulaw)
--
Can you paste the full NEW frame please. Could be Preference vs capability
thanks,
Adam
Ernie Ankele wrote:
On a sip to iax :
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 0ms SCall: 19170 DCall: 1
Adam, I think I got it worked out...
I changed disallow=723.1 to disallow=all and then accepted back in
ulaw,alaw,gsm and ilbc and
it started accepting the calls. I do not know why, but its working now.
FWIW, here is the full frame as it was before:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000
I have a customer I'd really like to be able to call on his DSL
connected Mac but he has an old OS. Has anyone had any succes with
older systems? Are there even any SIP/IAX clients at all for two or
three year old Macs?
___
Asterisk-Users mailing list
Hi !
I would like to know how to choose SIP IAX login for customer account.
We will provide them DID, the good idea will be to said, login=phone number.
But in fact not, since a user can get call from different DID and may change
one day is phone number.
Also some user may not have a DID but a
Hi !
I would like to know how to choose SIP IAX login for customer account.
We will provide them DID, the good idea will be to said, login=phone number.
But in fact not, since a user can get call from different DID and may change
one day is phone number.
Also some user may not have a DID but a
I am testing the following, and have no G729 codecs installed on my asterisk
-
Firefly [G729] - asterisk -- firefly [G729] which works fine,
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX2/3005|20|t) in new stack
-- Called 3005
-- Call accepted by 192.168.2.20 (format
Hello,
This might be a wild chase, but I'm looking for a driver / daemon
/ asterisk app which on one side speaks SIP or IAX, and on the other
side is a modem emulator.
Or to explain with an example: On my *nix box I want to start a
terminal program (minicom, kermit, whatever), connect to a
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