Any links on how to get a sipura 2000 to connect to
asterisk remotely over the internet ?
In my experience, getting the Sipuras (1000, 2000 or 3000) to connect
to * over the Internet is a piece of cake -- just make sure you have
the NAT related options turned on, if your SPA is behind a NAT.
We have deployed many (20+) IAXy's in the field. At a couple of
locations, the IAXy's have just stopped working after 1 or 2 days use.
No lights go on, no DHCP lease is renewed as far as we can tell, and of
course no dialtone and no registration with the server.
I bought two of them, both of them
On Fri, 21 Jan 2005, Henry Devito wrote:
Hi, I have not implemented any of the spa-2000's yet. Do they work ok with
asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is
it two fxs ports with the same extension?
They work pretty well, but I'm not impressed with the sound
Remco Barende wrote:
On Fri, 21 Jan 2005, Henry Devito wrote:
Hi, I have not implemented any of the spa-2000's yet. Do they work
ok with asterisk? Is the 2000 capable of having 2 FXS extensions off
each one or is it two fxs ports with the same extension?
They work pretty well, but I'm not
On Sat, 22 Jan 2005, Duane wrote:
Remco Barende wrote:
On Fri, 21 Jan 2005, Henry Devito wrote:
Hi, I have not implemented any of the spa-2000's yet. Do they work ok
with asterisk? Is the 2000 capable of having 2 FXS extensions off each
one or is it two fxs ports with the same extension?
At a couple of locations,
the IAXy's have just stopped working after 1 or 2 days use. No lights go on,
no DHCP lease is renewed as far as we can tell, and of course no dialtone
and no registration with the server.
I had that problem intermittently while using a 1A power supply. The
behavior
I have a an Asterisk server running asterisk 1.0.3 and a TDM04b card.
I'm having a problem with my setup. Incoming and outgoing calls are working to
95%.
When the other party hangs up their phone after I've hang up mine it starts
ringing in my phone. example:
1. I get an incoming call
2. I answer
Shouldn't you contact your vendor for support and not a different
vendors support channel?
As far I know, although Digium hosts the asterisk-users list and
supports the Asterisk development, Asterisk is still a GPL open source
project and asterisk-list is not a Digium support
Hi all,
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always different than 1.
In this moment, when I dial 123 for example, ATA waits till
timeout before dialing that number. The
Hi all,
- Original Message -
From: Steve Kann
..Quote
What would really help, though, is a packet trace of the call. The
best way to get this is to use either ethereal or tcpdump. (there is an
ethereal for windows).
If you use ethereal for Windows, have it capture all udp, make the
On Sat, 2005-01-22 at 04:38 -0600, Rich Adamson wrote:
Shouldn't you contact your vendor for support and not a different
vendors support channel?
As far I know, although Digium hosts the asterisk-users list and
supports the Asterisk development, Asterisk is still a GPL open
Hello,
I dunno if it's really needed, we should ask Mark. Anyway I created site
http://b2bua.berlios.de
where I will post all my asterisk patches and applications.
On Fri, 21 Jan 2005 11:57:52 -, Muhammad Nasim
[EMAIL PROTECTED] wrote:
Hi Mike
This is a damn useful app. Do you know if
- Originele Bericht -
Van: Jens
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Datum: Friday, 21 January 2005, 23:13
Onderwerp: Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Jens, thanks for the feedback.
No problem - but I think I
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 22, 2005 4:54 AM
Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork
X100Pinstall
On Sat,
Thomas Hger wrote:
So you can use all from mISDN supported ISDN catds in Asterisk.
Feel free to donwload and test it at :
http://www.beronet.com/download/chan_misdn-beta-0.0.3-rc5.tgz
You can report bugs and feature requests to www.beronet.com/bugs
Have fun!
Hi,
I have a few questions
I also have a SPA-2000 and it works just fine. However there a many options
available in admin mode, mainly on the Line 1/2 tabs and I've no idea what
many of then do. Is there a document somewhere that describes them?
The quick start guide that comes with the device only covers the basics
On Sat, 22 Jan 2005, Mike Dent wrote:
On Fri, 21 Jan 2005 19:25:06 -0500, Glenn Powers [EMAIL PROTECTED] wrote:
Mike Dent wrote:
What do you mean by provisioning?
loading the config files, with proxy servers, usernames, passwords, etc.
So basically its just a silly word for
Hi,
On Sat, 2005-01-22 at 11:46 +0200, Dan wrote:
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always different than 1.
In this moment, when I dial 123 for example, ATA waits
Hi,
Would someone assist me with the configuration of
my TMD10B and X100P cards? I would ask Digium but my support was good for only
one install...( I had to rebuild my * box). I wrote down the configuration but I
must have made a mistake somewhere. I have followed the instructions on
the
Hi Florian,
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
On Sat, 2005-01-22 at 11:46 +0200, Dan wrote:
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always
I've been trying to setup asterisk with an Internet Line Jack card for
sometime. I've been successful in configuring asterisk to handle incoming
calls, make calls between sip phones, call the asterisk demo, and even
When the call comes in, what's the channel * reports handling ?
Something
Keith Burns wrote:
I think you need to look at a few other factors.
...
2. Line power - Cisco uses one standard, other phones use another... but
Cisco is the 900# gorilla in the powered switch market... your call...
I'm curious about this point..
Most if not all vendors that support PoE are not
Great service. Hope it catches on.
In time there will probably have to be some adjustments:
- ratio 10/1 - this means everyone can make 10 calls for every call donated.
This doesn't add up in the long run.
- I get credited 10 calls for every call I make to my own dialplan. This
should be fixed.
Cisco came up with PoE before the standard was set and so it differs.
The polarity is switched, so using a dumb power injector and a crossed cable
one could make it work anyway.
Quoting Julio Arruda [EMAIL PROTECTED]:
Keith Burns wrote:
I think you need to look at a few other factors.
...
Great service. Hope it catches on.
In time there will probably have to be some adjustments:
- ratio 10/1 - this means everyone can make 10 calls for every call
donated.
This doesn't add up in the long run.
- I get credited 10 calls for every call I make to my own dialplan. This
Hi,
Since around a week i have one asterisk server how stop responding randomely.
CVS HEAD with RealTime engine used.
The debug log only write Failed to grab lock, trying again... until i
stop Asterisk.
No more activity for IAX or SIP channels (no log...). CLI still responding.
When i try to stop
Thanks for answering Chad,
Actually, I just want to Switch traffic between wholesale providers (my
customers) which actually terminate
traffic (or not, some of them have just controllers-softswitches like the
one Im willing to set up)
collect CDRs and bill them =)
I have no gateways of my own
Hi folks,
I got a demo version of a Siemens HiPath RG2200 Gateway/Gateway which I
probably have to integrate with an * PBX for some time:
Some HiPath extensions should be registered to the * box using H.323 ... as
a proof of concept for a smooth migration from HiPath to Asterisk.
Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
only game in town. I know that the Farfon device will be out soon and
we'll be able to judge its quality at that time.
Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
MGCP and N2P. (I've got one on my
For what it's worth, we don't sell very many of the IAXy devices,
perhaps 250 or so have gone out in the past 6 months, but we have only
had 2-3 of them fail, that is a pretty low defect rate.
I have no reason to plug the device, our margin on it is scant, but I
have seen much higher defect
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always different than 1.
In this moment, when I dial 123 for example, ATA waits till
timeout before dialing that number. The same
Hi,
- Original Message -
From: Adi Linden [EMAIL PROTECTED]
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always different than 1.
In this moment, when I dial 123 for
Is there ant chance of the Dialogic card model D/4PCI working with asterisk
?
Best Regards
Steve Beaumont
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To UNSUBSCRIBE or update
Folks,
I'm curious to know how the volume of Asterisk-Users rates as far
mailing lists go. This list sees over 200 messages per day, which has
GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has
knowledge of any organization that might maintain such stats.
Regards,
Jim.
--
Hello All,
How are you doing today? Good I hope.
I have started working on a new softphone which will have a Visual Basic
interface with communication stacks to some C/C++ stack to speed up the
sphone along with audio, video, and text messaging as well. hope to have
it done soon.
I have done a
On 22 Jan 2005, at 17:14, Steve Beaumont wrote:
Is there ant chance of the Dialogic card model D/4PCI working with
asterisk ?
The advice I was given when I asked a similar question about a dialogic
board
was, roughly : It is possible but it will cost you money for the
license and the results
Hi:
Can any one point me in the rite direction on this?
I am using asterisk at home for learning purposes. I am trying to get the
triditional *69 working.
Has there been any success in getting it to announce the number and get it
to give you the option to call back?
Chris
- Original Message
Hello All,
I am going to try and install Asterisk on a server machine that I have and
wanted to get the opinion of experienced people as to the best method.
Initially I was going to purchase the Asterisk Starter Kit at:
I have been getting the following message in Asterisk and it shuts Asterisk
down, needing a reboot.
Power alarm on Module 2
I have
(1) TDM400P with (2) FXS (2) FXO cards
(1) X100P card
Any ideas?
Thanks
Martin
___
Asterisk-Users mailing list
Hi all,
i am stuck with the configuration of
asterisk
- modules are loaded ( zaptel and wct4xxp
)
- i have zaptel.conf configure, output
of ztcfg -vv
--- snip --
rapid:~# ztcfg -vv
Zaptel Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db
(CSU)/0-133 feet (DSX-1)
SPAN 2:
Stig,
The 10/1 ratio is really just to get things started... over time,
we plan to lower the ratio, but in the mean time, it is kind of like
Reaganomics.
/ed
[EMAIL PROTECTED] / [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stig Hess
hi guys...
im a new asterisk user. i dont understand why asterisk doesnt start.
pls take a look at the output below.
i could start it a couple of times ..
#asterisk -gc
Asterisk CVS-v1-0-01/21/05-14:58:31, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
I am not sure about the image part, but for the settings thing, try
Settings button, option 9
Dan
On Fri, 21 Jan 2005, Manjit Riat wrote:
Just got my 7960 that I picked up from ebay. It looks like it has a SKINNY
image instead of SIP.. where can I get a SIP image ?
And how do I unlock the
Tim,
Thanks for the advice, the card was donated by a friend so I guess I better
let him have it back.
I've been sucessfully dabbling with asterisk for a while now, I saw this
card as a way of expanding my knowledge for little outlay :-) guess I will
have to bite the bullet and buy one of the
You really need to get hold of the SIP image and load it in to the phone.
I'm not sure but I seem to remember that there was something similar to the
unlock config of the SIP image in the skinny image.
If you have trouble getting hold of the image drop me an email.
Steve b
- Original
Hi!
According to this only the BT102D supports conferencing aka 3-way
calling:
http://www.grandstream.com/Product_Spec.pdf
This is vaporware, the product was cancelled. Anyway, with a recent
firmware you can do attended transfers with the BudgeTone, but no phone
based
Hi!
What are the advantages in using mISDN over other solutions?
- it replaces the (rather terrible) isdn4linux of the 2.4 kernels
- works with many many ISDN cards (no need for CAPI drivers or a HFC
chipset)
- most likely implements more ISDN features than bristuff
Haven't used mISDN
Hi,
If you still are in the Skinny image Settings --- Network config in that menu press **# and you will get the phone unlock.
Otherwise, if you are in SIP you need to do the following:
Once the telephone has booted -- Settings -- 9 Unlock config --- Enter password
The default password is cisco
Ok you want a ALL-OVER-IP softswitch,, asterisk may help in fact. but
not 100% sure about the accurrate billing.
regards
Humberto
On Sat, 22 Jan 2005 13:03:07 -0300, Diego Ventrice [EMAIL PROTECTED] wrote:
Thanks for answering Chad,
Actually, I just want to Switch traffic between
Hi, I am configuring a solution. The Asterisk with public IP outside NAT firewall and there is a private LAN behind any NAT/PAT firewall. What arethe key points in sip configure file I need to pay special attention? And except allowing the ports for UDP and RTP, do I need to do anything else in
Folks,
I'm curious to know how the volume of Asterisk-Users rates as far mailing
lists go. This list sees over 200 messages per day, which has GOT to put it
in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any
organization that might maintain such stats.
Regards,
Jim.
I
Before I start, I just want to say this is not necessarily a problem
with ASTCC, but may be a problem the way I have set up ASTCC (and
possibly the way others have set it up as well). The issue is that ASTCC
tries to match the pattern *anywhere* in the called number, not
necessarily only at the
On Sat, 2005-22-01 at 12:19 -0500, Jim Van Meggelen wrote:
I'm curious to know how the volume of Asterisk-Users rates as far
mailing lists go. This list sees over 200 messages per day, which has
GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has
knowledge of any
I had the same problem, and it's a database issue, not a code problem.
Use the character ^ in front of the pattern in the routes table, and I
think you will have better luck. E.g., ^1416... will match only
numbers that start with 1416.
Bruce Komito
High Sierra Networks, Inc.
Is there ant chance of the Dialogic card model D/4PCI working with
asterisk ?
Word of caution: Even if you can buy the drivers and make this card work
with *, it is not meant to plug directly into a CO -48vdc talk battery and
90-130vac ring voltage delivered by your phone company. These
Just checking, but if you're in Germany, don't you want E1 rather than
T1 settings? Not sure what a PMX is.
If you do indeed want E1, then something like the following would be
used in zaptel.conf.. Note also that you have to change the hardware
jumpers on the board (4) to the E1 position.
Hi all,
I'm trying to get Asterisk do AAA against a Cisco VSA compatible
RADIUS server/billing platform. I decided to give a try to appradius
(since it was written in C and didn't require all the modifications to
Asterisk that PortaOne's client does) played with it the whole day,
after lots of
Did you ever figure a way around this? It would be a good time to test
since LiveVoip is having some issues today.
On Sat, 8 Jan 2005 14:44:23 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
Hello.
I am having an issue where sometimes the cheapest provider for certain
international
OK, I have done all the stuff at my end and at Bellsters end to add 21
new area codes (all of california) to the Bellster dial plan. Pretty
cool deal! I hope others go for this quickly - as it could be a really
nice co-op.
I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk
Hi,
I have my IAX peers setup in iax.conf and use a qualify statement to see
whether the peer is up or not. I then use a macro for each peer. This way
I can take appropriate action depending on the DIALSTATUS variable.
exten = _9011.,1,Macro(gw-voipjet,${EXTEN:1}); VoipJet.com WORLD
Kanwar Ranbir Sandhu wrote:
On Sat, 2005-22-01 at 12:19 -0500, Jim Van Meggelen wrote:
I'm curious to know how the volume of Asterisk-Users rates as far
mailing lists go. This list sees over 200 messages per day, which has
GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has
On Sat, 22 Jan 2005 15:54:21 +0100, Kristian larsson [EMAIL PROTECTED] wrote:
Cisco came up with PoE before the standard was set and so it differs.
The polarity is switched, so using a dumb power injector and a crossed cable
one could make it work anyway.
Did you try it? and what are the
I love belster, I added a route for the 518 area code, (that covers most
of upstate NY), only thing I wish I could do is get rid of the message
that says how many credits I have left.
I would rather it just report congested is the call can't go though
(doto lack of credits), that way I could
I have a grandstream 101 that is calling an extension on Zap/1 of a TDM20B.
The grandstream 101 can call another grandstream 101 at a different
extension- that works fine.
The two phones on TDM 20B can call each other.- no problem.When I call
the TDM20B Zap/1
from the grandstream phone it rings
I just hope that the next one will shares his philosophy on VOIP.
On Fri, 21 Jan 2005 15:40:55 -0700, Brandon Patterson
[EMAIL PROTECTED] wrote:
This is not burn out. This is time to jump to a higher paying job
Brandon Patterson
I'd guess he's tired of it. It is fairly rare to see
Agreed - the announcement is not needed - although it's kind of neat -
perhaps it could be something that was optional/configurable from the
bellster web page?
Nathan Goodwin wrote:
I love belster, I added a route for the 518 area code, (that covers
most of upstate NY), only thing I wish I
Im having problems using the following
[sip]exten = _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})[macro-test]exten = s,1,Answerexten = s,3,Flashexten = s,3,Dial(SIP/${ARG2},30,t)exten = s,4,Dial(SIP/${ARG1},30,t)exten = s,t,Hangupexten = s,i,Hangupexten = s,h,Hangup
I know I must be
MJ wrote:
I'm having problems using the following.
[sip]
exten = _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten = s,1,Answer
exten = s,3,Flash
exten = s,3,Dial(SIP/${ARG2},30,t)
exten = s,4,Dial(SIP/${ARG1},30,t)
exten = s,t,Hangup
exten = s,i,Hangup
exten = s,h,Hangup
You
A couple of proposals for this excelent idea :
- Credits accounting based in minutes and not in number of calls.
- The possibility for the owner of the route to add a specific multiplier
factor for the route, for instance between 1 and 10. This way cellular
or expensive routes can give more
Sorry, what I actually have is...
[macro-test]
exten = s,1,Answer
exten = s,2,Flash
exten = s,3,Dial(SIP/${ARG2},30,t)
exten = s,4,Dial(SIP/${ARG1},30,t)
exten = s,5,Hangup
exten = s,i,Hangup
exten = s,h,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
As Eric pointed out you have priority 3 listed
twice here, but from the debug output I am guessing that's a typo. but in the
second priority 3 line, I think ${ARG2} is being replaced by ${CALLERIDNUM}
where he wants $EXTEN instead.
I don't think you need to dial your SIP extension
back
I like that idea, and if it worked that way I could gladly add a few
international routes.
Gustavo Russo wrote:
A couple of proposals for this excelent idea :
- Credits accounting based in minutes and not in number of calls.
- The possibility for the owner of the route to add a specific
Henry Devito wrote:
Is there ant chance of the Dialogic card model D/4PCI working with
asterisk ?
Word of caution: Even if you can buy the drivers and make this card
work with *, it is not meant to plug directly into a CO -48vdc talk
battery and 90-130vac ring voltage delivered by your phone
Should I take out the CallERIDNUM? The
following is where I got my config
http://lists.digium.com/pipermail/asterisk-users/2004-July/056878.html
Not in any way a good solution, but what I've done is create an extension that flashs the line, and then returns the call to my sip phone.
BTW, the second incoming call never gets
answered by *.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Saturday, January 22, 2005
8:13 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
flashing zap using
C F wrote:
Did you try it? and what are the pinouts for such a crossed cable?
I have been using this to power our 7940's with a 3COM injector.
http://www.voip-info.org/tiki-index.php?page=Cisco%20POE
Calvin
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Asterisk-Users mailing list
Any thoughts? Could this be a jitterbuffer problem?
On Fri, 21 Jan 2005 19:08:59 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls. During the call, the calling parties
voice sometimes sound
I would suggest you go with the easy road :
- install CentOS : http://www.centos.org/
- then download Asterisk 1.0.4 (latest stable) :
ftp://ftp.asterisk.org/pub/asterisk/
- install it by following this document :
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
then play with
On Sat, 22 Jan 2005 22:03:45 -0500, Brian Dingman wrote:
Any thoughts? Could this be a jitterbuffer problem?
On Fri, 21 Jan 2005 19:08:59 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls.
I'm a firefighter and we have a lot time off. So I help out with the other
firefighters. They have small businesses and I install Linux Servers. They
seem to like Webmin. I have one firefighter that runs a small pool clean
business and needs a small PBX.(3 lines) Anyways, its just easyer on me to
I am seeing the following
asterisk*CLI iax2 show registry
217.160.244.186:4569 usernamexx.xx.xxx.xx:4569 60 Registered
66.234.228.170:4569 username xx.xx.xxx.xx:4569 60 Registered
65.39.205.121:4569username xx.xx.xxx.xx:4569 60 Registered
On Sat,
[EMAIL PROTECTED] wrote:
I would suggest you go with the easy road :
- install CentOS : http://www.centos.org/
- then download Asterisk 1.0.4 (latest stable) :
ftp://ftp.asterisk.org/pub/asterisk/
- install it by following this document :
Did you ever figure a way around this? It would be a good
time to test since LiveVoip is having some issues today.
No, I'm afraid I never found a solution. I posted a feature request on
Mantis but I guess there isn't enough interest.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1
I have my IAX peers setup in iax.conf and use a qualify
statement to see whether the peer is up or not. I then use a
macro for each peer. This way I can take appropriate action
depending on the DIALSTATUS variable.
That doesn't really solve the problem of when the peer is up but not
able to
I'm having this exact problem (blown speaker audio quality on
incoming) with iax.cc / sixTel.
I'm on Asterisk 1.0.3, White Box Linux, no load on server, one Sipura 1001.
Any ideas on solving this?
--
Lee
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Asterisk-Users mailing list
Hi there,
I am experiencing some issue with X-Lite.
When I am calling over the phone thru my PSTN-to-VoIP gateway
internationally using G.729 the quality is just perfect.
When I am using X-Lite to connect the same box, and then to call
internationally - I am experiencing some issues.
I have
Hi Sergey,
Have you tried phoning from X-Lite to your PSTN line, or your PSTN line
to X-Lite? How is the audio quality then? Does it vary depending on the
codec you have used?
Andrew
On 23/01/2005, at 4:31 PM, Sergey Kuznetsov wrote:
Hi there,
I am experiencing some issue with X-Lite.
When I am
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 22, 2005 8:15 PM
Subject: Re: [Asterisk-Users] Dialogic D/4PCI
Henry Devito wrote:
Is there ant chance of the
On January 22, 2005 11:16 pm, Nabeel Jafferali wrote:
I have my IAX peers setup in iax.conf and use a qualify
statement to see whether the peer is up or not. I then use a
macro for each peer. This way I can take appropriate action
depending on the DIALSTATUS variable.
That doesn't really
On January 23, 2005 12:14 am, Lee wrote:
I'm having this exact problem (blown speaker audio quality on
incoming) with iax.cc / sixTel.
I have *no* issues on inbound quality with sixTel. They *had* a problem where
the first second of audio was cut off upon connect (Wait() did not help) but
I can run iaxcomm by itself...and I start up Asterisk
on it own...
But if I start Asterisk first, then launch iaxcomm
I get this error:
bash-2.05b$ ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX
Client .
bash-2.05b$
and if I start iaxcomm first then launch asterisk, I
get this
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