Well, for the last part of my email, I now know of AgentCallbackLogin
You see.. Asterisk is your friend!
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Lunes, 21 de Febrero de 2005 01:48 a.m.
To: 'Asterisk Users Mailing List -
Hi,
with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
asterisk-oh323 v.0.6.3b and it works fine
hope it helps
cu...
--- kolo sos [EMAIL PROTECTED] wrote:
Hi,
anybody knows what's missing or problem why i cant
compile asterisk-oh323 in my machine?
i got this compiled successfully
Check your soundcard controls... maybe it's recording what you hear
or PCM, thus sending it again to the other party.
Julianjm.
On Mon, 21 Feb 2005 09:47:55 +0200, Nic le Roux [EMAIL PROTECTED] wrote:
Good Morning,
I have a weird situation,
I'm testing with Xlite as SIP phone (is it
Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok?
also... can you make MOH random?
Also, I dont know if there is a problem with my config but when listening to
MOH, every 3 or so second I get a click sound which notices because music
gets a hickup every 3 or so seconds...
You can't change the callerid on an outgoing PSTN call (at least on
analog lines).
To modifiy the callerid on incoming calls, you could do something like
this (not tested):
[incoming-line1]
exten = s,1,setCidName(Line1: . ${CALLERID})
exten = s,2,Goto(Incoming,s,1)
[incoming]
exten =
Hi all,
I have a brand new TDMxxx card with 3 FXO modules and one FXS.
It has replaced my old 3 X100P cards.
The FXO part work as before, after some adjustments in the rxgain/txgain
part.
The problem I have is with the FXS module.
I can place calls to SIP/IAX or PSTN destinations without any
I haven't used it in a while, but I had to put subscribecontext=sip
for the phone's (in your case the snom) sip entry.
This seems like it has been removed from the wiki. Has it changed or
is this incorrect?
Hi James,
I have just found out that all you need to do is make the hint in the
Hi,
I have a problem that is biting at all my customer sites where they have
PRIs taking heavy load.
This happens both with the stable code stream and with the current CVS.
What happens is that after some running, Asterisk starts reporting strange
errors on the PRI, eventually calling the
I haven't used it in a while, but I had to put subscribecontext=sip
for the phone's (in your case the snom) sip entry.
This seems like it has been removed from the wiki. Has it
changed or
is this incorrect?
Hi James,
I have just found out that all you need to do is make
Dear folks,
I just made a release of the calling card application AreskiCC
Please check it out :
http://areski.net/areskicc-doc/
Reported bugs has been fixed.
I advice to all users to make the update.
Further informations on the release into the CHANGELOG file.
Kinds regards,
Areski
Ok your example confused me a little.
You put 690,hint,SIP/bt-karen
From this section in my extensions from your example I should have
exten = 690,hint,SIP/bt-karen
exten = 691,hint,SIP/snom-james
So set hint on the opposite extensions?
[sip]
exten = 690,hint,SIP/snom-james
--
If I put the parameter 'caller', when I execute the call 'sample.call',
the application txfax realizes two calls. One to the fax machine and
other to my own asterisk.
My Asterisk detects that the incoming call is a fax and begins to save
it with 'rxfax'. In another call, 'txfax' says:
I did change the port 4569.
Also my router forwards those packets.
If I start tcpdump port 4569 on my server I receive:
04:25:36.061292 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569:
UDP, length 24
04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569:
UDP, length
Title: Problem with ISDN Dialin via CAPI
Hello List,
because this is my first post to this list, i'd like to introduce myself.
My name is Thorsten Müller, 26 years old and live near Frankfurt/Main in germany.
Okay, now to the reason for this posting:
I just installed my first asterisk
This was set on linksys wrt54.
I turned on the forwarding to asterisk server on port 4569.
I believe that by default all outgoing packets pass through.
Bart,
Hallo
Did you allow udp outgoing on 4569 as well.. i found
udp bit different than
tcp when comming to firewalls
liaan
-
You can only set up the 220 with an extended key pad. both phones, the 220 as
well as the 200, support up to 7 SIP lines/registrations.
Regards
Nils Ohlmeier
On Monday 21 February 2005 05:53, dkwok wrote:
Snom 200 has be set up with extended key pad. The product literature
also mention
well, it seems like the 2 are communicating
correctley.. just went through
all the logs
what is the error that you recieve?
- Original Message -
From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Hey Guys
Im trying to find out how to transfer a call with ECT (explicit call
transfer) ?
Im currently transferring a call as following:
exten=2,1,Dial(capi/720:07,18)
exten = 2,2,Goto(2-${DIALSTATUS},1)
exten = 2-NOANSWER,1,Dial(capi/720:07979)
exten =
hallo all,
i have a quicknet LineJACK card and it seems to work ok, the only problem
is, that when i use this in extentions.conf,
exten = _[1-9]., 1, Dial(IAX2/krath:[EMAIL PROTECTED]/${EXTEN},50,Ttr)
exten = _[1-9]., 2, Congestion
it dials only 2 digits, e.g when i dial 1234 it dials only 12,
Hi all,
I'm having a weird problem. The setup is Asterisk A with a
TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another
DSL line.
Both boxes are behind their own NAT. Asterisk B forwards calls from it's
four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using
I'm having a weird problem. The setup is Asterisk A with a
TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another
DSL line.
Both boxes are behind their own NAT. Asterisk B forwards calls from it's
four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using
Rich Adamson wrote:
I'm having a weird problem. The setup is Asterisk A with a
TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another
DSL line.
Both boxes are behind their own NAT. Asterisk B forwards calls from it's
four PSTN ports to Asterisk A over an IAX2 trunk, which works
I am going to now sit in a corner and go quietly insane while playing
the banyo with no strings.
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone did not come
on. I dialed out of extension 691 to an outside line, yet
On Mon, Feb 21, 2005 at 11:36:45AM +0100, Müller, Thorsten wrote:
Hello List,
because this is my first post to this list, i'd like to introduce myself.
My name is Thorsten Müller, 26 years old and live near Frankfurt/Main in
germany.
Okay, now to the reason for this posting:
I just
Anyone having problems compiling the current cvs head this morning?
New cvs checkout on RH9, followed by appropriate make clean and make
install. System was running cvs head from Nov 23 with TDM card, PRI,
SIP phones on local wire, and IAX.
Appears zaptel and libpri compiled correctly, however
Hi,
Using latest cvs.
A comment-line begins with semicolon ;
However - if the line contains
;--
or like this
; -- blabla bla --
You get this error and * stops reading that file:
Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load:
Unterminated comment detected beginning
We have the following scenario:
Incoming call to a queue, Agent A answers. Agent A determines after
about 20 seconds that agent B needs to deal with this call. A puts
call on hold, calls and speaks to B, and then transfers the call to B. B
speaks to the incomming caller for 5 minutes.
That's
Hi,
Using latest cvs.
I (as many otheres it seems) can't get Attended transfer to
work with Cisco ATA186 (using SIP)
Has anyone else had any luck?
Same with 3-part calling, if one drops off, all are disconnected...
/Stig
___
Asterisk-Users mailing
I've got a nagios plugin making sure the * box is up, but I would like
to do more than that.
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
-Daniel
I've used a Nokia 32 unattended (remote) for the past year or so.
David Uzzell [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
| Ok I have a question. Seen it come and go around the mailling list for a
| while but never really seen an answer that seems to sort it out.
|
| What is
I am going to now sit in a corner and go quietly insane
while playing
the banyo with no strings.
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone
did not come
on. I dialed out of extension 691 to an
Is there a way to make a join conference between 2 lines? like when you have
2 incoming calls and you merge them together with you? how can you do this
on * if its possible?
Transfert them both to a conference room, then join that conference.
At least, that's how I would do it.
Send us your DIAX configuration.
Denis.
Em Seg 21 Fev 2005 07:29, Bartosz Wegrzyn - asterisk escreveu:
I did change the port 4569.
Also my router forwards those packets.
If I start tcpdump port 4569 on my server I receive:
04:25:36.061292 IP 192.168.1.253.4569
On Mon, 21 Feb 2005 02:42:41 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok?
also... can you make MOH random?
Yes, this if fine. Please read the archives. Use google. Use the wiki.
Again, on the random, read the samples, use
Daniel Corbe wrote:
I've got a nagios plugin making sure the * box is up, but I would like
to do more than that.
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
-Daniel
I corresponded with Voicetronix around Christmas last year. Jim, there
is a dealer in Ottawa although I got better answers from emails to Aus.
There are two things that they don't do that the Zap cards do:
Distinctive Ring Detection and fax detection.
They went out of their way to say they were
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
There aren't any specific tools that do exactly what
carrier via a PRI, they will dictate what
the DID looks like. Some will be the last 4 digits, others
will be all 10. (assuming US). They do this, because it would
be to difficult to maintain your extension mapping on their side.
You purchase a DID. When a call comes in it says, This is the
Title: [SOLVED] Problem with ISDN Dialin via CAPI
Hi,
i was able to solve my problem.
During my playing around with * and capi i changed several options in
config files.
I did this while my * was running. To test if my changes where successful i entered reload on * console.
This didn't
Hi All,
As my previous mail was not posted on the list for more than 10 hours now,
I'll try to resend it.
Thank you,
Dan
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February
I'm trying to disable musiconhold, simply because I don't need it.
But then chan_iax2 is complaining : undefined symbol: ast_moh_stop.
Is there a way to completely disable moh (maybe a compilation option) ?
Thank you.
Yves
___
Asterisk-Users mailing
Hello,
I have done some
browsing through the wiki and on Google and havent
been able to find anything that looks like what is happening to me. When I start Asterisk by typing asterisk
vvvc, I get Illegal instruction
and nothing else. Nothing
before and nothing after.
This is a Via Cyrix
Hi
This is my script for my local forecast for SE England. I have had
problems getting festival to work integrated so I have cron run this
script every 3 hours and use Playback to play it in Asterisk:
Script
--
#!/bin/sh
cd /var/lib/asterisk/sounds
curl
Sorry if this subject has been covered, but, my boss claims to have a working
IAX2 trunking set-up without a timing source on one side of the connection. In
all the posts and documentation regarding this subject, this appears to be
impossible. My questions are: 1) could this be true? 2) if it
Okay
here's a quick and dirty little perl script to monitor the PRI Status
and mimic nagios plugin output.
-Daniel
On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy [EMAIL PROTECTED] wrote:
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs
Just to bug you all (feel free to rant at me),
a client wants to set his caller*ID number for outbound calls though us
to PSTN.
the client is using SIP to us, he can set the caller*ID name fine.
if he sets his caller*ID number to anything other than his account
number (8440101), the
Asterisk wrote:
We have the following scenario:
Incoming call to a queue, Agent A answers. Agent A determines after
about 20 seconds that agent B needs to deal with this call. A puts
call on hold, calls and speaks to B, and then transfers the call to B. B
speaks to the incomming caller for 5
On 21 Feb 2005, at 01:21, Lyle Giese wrote:
Hmmm, maybe you need to re-read the instructions? You missed a major
step.
make clean; make install *is* the recommended way to compile and
install.
See http://www.asterisk.org/index.php?menu=download.
As for your error - I don't think there really
Andrew Thompson wrote:
Asterisk wrote:
We have the following scenario:
Incoming call to a queue, Agent A answers. Agent A determines
after about 20 seconds that agent B needs to deal with this call. A
puts call on hold, calls and speaks to B, and then transfers the call
to B. B speaks to the
On Mon, February 21, 2005 8:47 am, David Cook said:
There are two things that they don't do that the Zap cards do:
I've received nothing but positive and rapid support to any issues I've
had with my Voicetronix card. We should make the distinction between the
low-level VPB device driver
Hi There,
Thanks for your reply.
Where can I read up on doing this or maybe you could point me in the right
direction.
I don't believe that I have recording enabled.
| Julian Wrote:
|
| Check your soundcard controls... maybe it's recording what you hear
| or PCM, thus sending it again to the
I am trying to get the name and number to show up for an
incoming calls on my Polycom IP 500.
Right now only the name shows up, but in the call list both name and number
show up. Any help on what to change in the config file would be greatly
appreciated.
Thanks
Mark
Hi,
Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk?
I'm especially interested if you've used it with a TE405P or TE410P.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
Mark Floyd wrote:
I am trying to get the name and number to show up for an incoming calls on
my Polycom IP 500. Right now only the name shows up, but in the call list
both name and number show up. Any help on what to change in the config file
would be greatly appreciated.
Watch the display.
I've finally got my Adit 600 and are configuring it right now.
But I have to say, there aren't much documentation for it.
I've setup MGCP and Asterisk seems to find it.
But all channels (40 FXS channels) are Down!
But the MGCP itself is Up according to the statistics.
I can't find any documents
I am trying to get the name and number to show up for an incoming calls onmy Polycom IP 500. Right now only the name shows up, but in the call listboth name and number show up. Any help on what to change in the config filewould be greatly appreciated. Watch the display. Once you answer the
On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote:
I've got a nagios plugin making sure the * box is up, but I would like
to do more than that.
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
Morgan Gilroy wrote:
To get around this i updated CVS HEAD and changed the sip entity from
type=user to type=peer (yes peer!) (type=friend works too but im making
a point) the client now must register to set his outbound caller*ID Number.
Yes, that is normal. SIP has difficulty separating the
El 21/02/2005, a las 12:30, James Bean escribió:
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone did not come
on. I dialed out of extension 691 to an outside line, yet still the
light did not come on.
Snom190 has firmware
Mark Floyd wrote:
Yes that works, but it would be nice to see who is calling before I pick
up. Is there a way to make that happen, have both name and number, or
just number show up for incoming calls before I answer.
You identify incoming callers by number, and not by name? Odd.
In any case,
On February 21, 2005 10:25 am, Tony Mountifield wrote:
Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk?
I'm especially interested if you've used it with a TE405P or TE410P.
I'm actually using a 7043P-8R with a TE405P (hacked it so it ran in a 3.3V
slot) -- it works but I
Hello All,
I'm having problems with international calling via Global Crossing. I'm
told I need to send a true ani versus a sudo ani. What is the difference and
how can I configure asterisk to do this. Global Crossing is denying calls
with sudo anis.
Thanks,
Keith
On Mon, 2005-02-21 at 09:26 -0500, Tommy Vielkanowitz wrote:
Hello,
I have done some browsing through the wiki and on Google and
havent been able to find anything that looks like what is happening
to me. When I start Asterisk by typing asterisk vvvc, I get
Illegal instruction and
To get around this i updated CVS HEAD and changed the sip entity
from
type=user to type=peer (yes peer!) (type=friend works too but im
making
a point) the client now must register to set his outbound caller*ID
Number.
Yes, that is normal. SIP has difficulty separating the remote
On Mon, 2005-02-21 at 16:33 +0100, Daniel Nyström wrote:
I've finally got my Adit 600 and are configuring it right now.
But I have to say, there aren't much documentation for it.
I've setup MGCP and Asterisk seems to find it.
But all channels (40 FXS channels) are Down!
But the MGCP itself is
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote:
Anyone having problems compiling the current cvs head this morning?
New cvs checkout on RH9, followed by appropriate make clean and make
install. System was running cvs head from Nov 23 with TDM card, PRI,
SIP phones on local wire, and
I don't believe the adit 600 has an up/down for channels.
Are the channels connected to something. You might
look at the 'connect' command and see if that helps.
To bring the FXS channels up on my box I needed to
connect them to the T1 (in your case it would be the MGCP)
The t1 syntax is I
Hello list, I have been working with asterisk for a coupleof months and nowI have run into aproblem, I have the following setup
PSTN ==Asterisk(remote behind nat)===IAX==Asterisk(local public ip)OH323Gateway
I want to terminate incoming calls from the gateway in
Hi folks,
Does anyone know if there is a small test board that has a mike and
speaker? Board should run an OS that supports asterisk. I want to load
asterisk on to it and test out.
Thanks in advance,
Kiran
___
Asterisk-Users mailing list
Worked Great! Thx Julian..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Lunes, 21 de Febrero de 2005 02:46 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID
You can't change
Hi all,
I have a brand new TDMxxx card with 3 FXO modules and one FXS.
It has replaced my old 3 X100P cards.
The FXO part work as before, after some adjustments in the rxgain/txgain
part.
The problem I have is with the FXS module.
I can place calls to SIP/IAX or PSTN destinations without any
Morgan Gilroy wrote:
you mean amalgamating user and peer so there will eventually only be one
type for both incoming and outgoing calls, (hopefully have an option to
disable enable in/out bound calls).
Yes, exactly (and there will be other settings as well, to identify the
type of peer (network,
I'm doing something like that on my system --
http://muware.com/asterisk
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 1:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] CallerID
Has anyone implemented callannounce?
Here is an example.
1. Caller dials into asterisk
2. Call chooses Sales Extension
3. SalesPErson picks up the line and asterisk says you have a call from
Sales Press 1 to accept the call Press 2 to send to voicemail. press 3
to hear caller ID
Is something
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote:
Anyone having problems compiling the current cvs head this morning?
New cvs checkout on RH9, followed by appropriate make clean and make
install. System was running cvs head from Nov 23 with TDM card, PRI,
SIP phones on local wire,
Anyone tried this yet?
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
o713-861-4005
o800-905-6412
c713-201-1262
Is there a way to prioritize calls in multiple queues based on hold time? I
have three queues set up on my Asterisk PBX with agents logged into all
three queues. I've noticed that sometimes calls in one queue will make it
through in a couple minutes while another queue will be backed up with
I have two * boxes running two differnet versions of *.
Box A is running:
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
Box B is running:
Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD
I can make a IAX call from B to A but not
Hi,
I have two asterisk machines, chomper and otao.
otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no
PSTN connections.
chomper is at my house, behind NAT, but has a single X100P PSTN connection.
I would like to establish two way calling between otao and chomper.
Right
On Sun, 20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED] wrote:
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
FSX modules) and is connected to the PSTN. B has same, but is NOT
connected to PSTN. I
Mark Floyd wrote:
Yes that works, but it would be nice to see who is calling before I pick
up. Is there a way to make that happen, have both name and number, or
just number show up for incoming calls before I answer.
You identify incoming callers by number, and not by name? Odd.
In any
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora Core1
on my local network.B refuses any network connection attempts from
A, i.e. I can't even telnet or FTPto B from A, but I canto A
from B. This makes B refuse
I was looking at the exercise as a bit of Linux lerning for myself, so I
guess Mandrake 10.1 and mISDN? Does anyone have working examples?
Ray
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: 20 February 2005 23:57
To: Asterisk Users
Eric Wieling wrote:
Yes. There are lots of messages in the mailing list archives regarding
this problem, some of them even include things to try. You didn't see
these messages when you searched the mailing list archives?
Yes, I've read then.
They say it can be caused by interruptions.
I
Yes 7 lines on the SNOM 200 SIP phone.
Use a web browser to connect to your phone's IP address. There is a
world of things it can do via its built-in web server. Just don't change
the setting that says where to get the photos from, leave it as from
the phone.
Each line can be configured to
On 13:27, Mon 21 Feb 05, Thorben Jensen wrote:
I am going to now sit in a corner and go quietly insane while playing
the banyo with no strings.
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone did not come
on.
Anyone here technical enough to design a voice recognition voice
xml interchange for asterisk please email me; Ive been speaking with a
contact of mine that is in the voice recognition space and he is interested in donating
some technical support to the Asterisk community to assist with
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora
Core1 on my local network.B refuses any network connection
attempts from A, i.e. I can't even telnet or FTPto B from A, but
I canto A from B. This makes B refuse
On Mon, 2005-02-21 at 12:11 -0600, Rich Adamson wrote:
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote:
Anyone having problems compiling the current cvs head this morning?
New cvs checkout on RH9, followed by appropriate make clean and make
install. System was running cvs head
Hi Folks,
I installed [EMAIL PROTECTED] on my PC. It went through the installation
and all. But now i get a command line login window. Doesn't it has a
KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED]
web GUI)? After i login, just the command line interface comes out.
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing
context
;add function here to
What are the advantages/disadvantages of using
a ZAP FXS port versus using one of the many
small ethernet FXS devices on the market. The
ZAP FXS talks directly to asterisk over PCI. Is this
an advantage? The ethernet devices I assume
speak either iax2 or sip, does this cripple the
Sorry for the cross post, but I'm still trying to find a Seoul DID. I
received an email from LiveVoip.com that said they have service in
South Korea, but when I called them they said they didn't offer such
service.
If you have the capability to offer a DID please let me know what your
pricing
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Looks like a hardware problem as you had failures in
different locations
but both where a gcc seg fault. This means either your CPU is hot and
starting to spit out randomness or your memory is failing and
Sergey Kuznetsov wrote:
This is happens because of imperfect HDLC code.
Do you mean the software? The source code?
[]s
--
Alex G Robertson
NOC - Microlink
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Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing context
;add function here to
Alex G Robertson wrote:
Eric Wieling wrote:
Yes. There are lots of messages in the mailing list archives regarding
this problem, some of them even include things to try. You didn't see
these messages when you searched the mailing list archives?
Yes, I've read then.
They say it can be
On Mon, 2005-02-21 at 12:11 -0600, Rich Adamson wrote:
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote:
Anyone having problems compiling the current cvs head this morning?
New cvs checkout on RH9, followed by appropriate make clean and make
install. System was running
Guys
Ive setup FWD using IAX according to all the docs and I tried the give me a
call url on FWD webpage and I do get the call but when asked to say my
name, I hear a voice saying it didnt get it.. seems my voice is not getting
thru to FWD... anybody had this problem while setting up FWD with
Yeah, I'd be interested in porting your work so it runs under nagios.
Please post your results when you're finished.
-Daniel
On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote:
I've got a nagios plugin making
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