On Sat, Oct 29, 2005 at 10:58:22PM -0600, Ryan exclaimed:
snip
The problem is caused on the provider side, not in your asterisk. The
packets they are sending to your asterisk have the wrong sequence
numbers. I finished some code lastnight to *help* work-around this
issue, I just need to put the
See if http://www.vid.com.au/index.php?option=com_contenttask=viewid=13Itemid=37 or
http://www.westany.com/ - they may have pounds and pence.RobOn 10/30/05, Mark Phillips [EMAIL PROTECTED]
wrote:Ah, no pence huh.I guess I'll have to add that to my list of updates.
MarkJP Carballo wrote: Obelix
We have a voicemail solution, but here's a way you can do it in your
dialplan below. This also allows you to check messages remotely and/or
disable voicemail. Disabling voicemail is desirable on real fax machine or
credit card lines.
[macro-resvoice-handlevoicemail]
exten =
Dear All :
I wonder if the Announce feature is available in Asterisk MeetMe
Room ?
Announce Feature is the feature which play some kind of
announcement when someone joins the MeetMe room
Of course it also may record his name and his
company , then announce all the members of the
On Sat, Oct 29, 2005 at 11:22:33AM -0700, JP Carballo wrote, in reply to Obelix:
O.T. Is the Asterisk the Gaul comics still in circulation? It's been
years since I read the series...
No, it is not. In fact, the current Asteri*x* is not very
penguin-friendly:
http://mobilix.org/
--
Adam Moffett wrote:
does anyone know when 1.2 will no longer be beta?
The quick answer is: When it's ready for release.
Open Source software doesn't really follow a set agenda. We have been in
code freeze for quite a while, fixing bugs. A lot of people are testing
the 1.2 beta and reporting
Mark Hulber wrote:
In recent CVS Head build when I run: sip show peers my dynamic peers
show:
Name/username HostDyn Nat ACL Port Status
sipura2_2/sipura2_2(Unspecified)D N 0UNKNOWN
sipura2_1/sipura2_1(Unspecified)D N
Hi folks,
whoever owns a Cisco phone and is unhappy about slow firmware,
incomplete XML support etc... should really have a look at Sergio
Chersovani's rewrite of chan-sccp!
It is in fact far away from any sccp-channel drivers we used to know
just half a year ago. Sergio did a complete
On 10/30/05 09:47 James Sizemore said the following:
Now the pri's do load and are signaling via national 2 but I would like
to know why they
are being ignored and how do I get it to not Ignore tone duration?
a reload will always ignore switchtype et al. you'd need to restart
asterisk,
Hi,
I would like to use pattern matching in what some call the ex-girlfriend rule:
[demo]
exten = 830449/_0721.,1,Answer()
exten = 830449/_0721.,2,Dial(SIP/stefan,20,tr)
When I dial 830449, asterisk tells me:
Channel 'CAPI/ISDN1/830449-3' sent into invalid extension 's' in context
'default',
Folks,
* newbie trying out 1.2-beta. Want to make sure I haven't missed some
dialplan invocations (or perhaps waving of chicken feet, etc.).
calling voicemailmain() works for me to the point I get to hear the
message left by someone. However, the * docs I've read don't seem to
say much, so I
do i need to install libpri? my only setup is Digium TDM400P with 2 fxo port.
-- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame...
http://www.spreadfirefox.com/?q=user/registerr=19441
___
--Bandwidth and Colocation sponsored by
On Sun, 2005-10-30 at 09:03 -0500, David Bandel wrote:
Pointers to the correct FM to RTFM appreciated. Need to incorporate
the usual press 3 to delete, 7 to save, 9 to skip to the next
message prompts. (Odd no examples in the extension.conf.samples for
this.)
Well www.asteriskdocs.org has
I have asterisk installation, using CVS-HEAD.
When call from D-Link 2004S (SIP) to _some_ numbers (e.g. mobile
numbers), after answer on other side D-Link user continue hear progress
tones.
Callee can hear, what D-Link user say, and ifter he try to send DTMF
(press any key on mobile phone),
I have asterisk installation, using CVS-HEAD.
When call from D-Link 2004S (SIP) to _some_ numbers (e.g. mobile
numbers), after answer on other side D-Link user continue hear progress
tones.
Callee can hear, what D-Link user say, and ifter he try to send DTMF
(press any key on mobile phone),
...but I'm gonna ask it anyway, because I can't figger it out...
Every call that is bridged in my * system begins with a console message
like this one...
-- Attempting native bridge of SIP/215-b09e and SIP/259412-5967
Now, I've got canreinvite=no in every sip definition, but it happens
no, libpri is only needed for pri trunksOn 10/30/05, Mark Quitoriano [EMAIL PROTECTED] wrote:
do i need to install libpri? my only setup is Digium TDM400P with 2 fxo port.
-- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame...
On Sunday 30 October 2005 09:44, Bill Michaelson wrote:
-- Attempting native bridge of SIP/215-b09e and SIP/259412-5967
Now, I've got canreinvite=no in every sip definition, but it happens
anyway.
That has nothing to do with reinvites.
In Asterisk terms, a native bridge between two channels
On Sunday 30 October 2005 09:48, Michael Bielicki wrote:
no, libpri is only needed for pri trunks
It's also needed for ISDN BRI, I think...
Certainly not for analog FXS or FXO though, you're right.
-A.
___
--Bandwidth and Colocation sponsored by
Solved.
On 10/30/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Sun, 2005-10-30 at 09:03 -0500, David Bandel wrote:
Pointers to the correct FM to RTFM appreciated. Need to incorporate
the usual press 3 to delete, 7 to save, 9 to skip to the next
message prompts. (Odd no
using the CLI in - mode showed the problem. Apparently, I can't
spell (or I can, but when I was typing, I transposed two letters and
made it vm-recieved vice vm-received).
Perhaps a good enhancement would be a syntax checker for the various
.conf files.
Been there... sure wish
I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the
how-to on the geekgazette as well, however, my sipura-3000 only just sits
and rings and rings and rings. I have set up the peer and the user values,
as per the configuration, and when I look at the web status info page of
Hello.
I wonder if someone cal help me to find the right way to implement the below
described TO-BE scenario (basically automatic farwarding from incoming
calls).
*** Background:
- a VoIP/PSTN gateway Mediatrix 1104 registers on [EMAIL PROTECTED] as UAs from
301 to 304. This Mediatrix is the
Thanks for the answer. Doesn't solve my problem, but that's only because I didn't state my goal. You have corrected a misconseption on my part, which ought to get me closer. I'll explain...
Indeed, I do have the "tT" options in the dial command. This is because I thought this would
On Sunday 30 October 2005 11:13, Bill Michaelson wrote:
Indeed, I do have the tT options in the dial command. This is because I
thought this would enable the use of the '#' for transfers, and it works
satisfactorily. I also have various '*N' definitions in features.conf, but
these don't
Have you read this?
http://voipspeak.net/index.php?option=c . d=
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: den 30 oktober 2005 16:12
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
ok tnx guys.On 10/30/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Sunday 30 October 2005 09:48, Michael Bielicki wrote: no, libpri is only needed for pri trunksIt's also needed for ISDN BRI, I think...Certainly not for analog FXS or FXO though, you're right.
for BRI only if you have patched it for bristuff :)On 10/30/05, Mark Quitoriano [EMAIL PROTECTED]
wrote:ok tnx guys.
On 10/30/05, Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Sunday 30 October 2005 09:48, Michael Bielicki wrote: no, libpri is only needed for pri trunksIt's also needed for ISDN
On 10/30/05, David Bandel [EMAIL PROTECTED] wrote:
Have the OReilley book. Also the new 1.2 book from asteriskdocs.org.
Pt... they're the same book :)
--
Leif Madsen - http://www.leifmadsen.com
http://www.asteriskdocs.org -- Co-Founder
http://www.oreilly.com/catalog/asterisk -- Co-Author
That link is not found
Have you read this?
http://voipspeak.net/index.php?option=c . d=
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: den 30 oktober 2005 16:12
To: [EMAIL PROTECTED]; Asterisk Users
whoever owns a Cisco phone and is unhappy about slow
firmware, incomplete XML support etc... should really have a
look at Sergio Chersovani's rewrite of chan-sccp!
Is there a good resource out there for people who don't have a lot of
experience with Cisco phones? I picked up a 7960 earlier
Is this release under the GPL?
I see no mention of this windows based program on your web site.
::)
Paul
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Saul Diaz
Sent: Saturday, October 29, 2005 11:08 PM
To: Asterisk Users Mailing
http://voipspeak.net/index.php?option=com_contenttask=viewid=24Itemid=27
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: den 30 oktober 2005 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Have a look here, :
http://www.asteriskguru.com/tutorials/cisco_7960_skinny_chan_sccp.html
If you find any other suggestions, remarks after installing, please post
them as a comment to the page.
Zoa
Chris Bagnall wrote:
whoever owns a Cisco phone and is unhappy about slow
firmware,
Just updated cvs-head this morning, and now on a 'stop now' and restart,
* doesn't know about the previously registered sip phones (as shown with
sip show peers) on fc3.
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet... anyone seeing the
Andrew Kohlsmith wrote:
On Sunday 30 October 2005 11:13, Bill Michaelson wrote:
Indeed, I do have the tT options in the dial command. This is because I
thought this would enable the use of the '#' for transfers, and it works
satisfactorily. I also have various '*N' definitions in
Chris Bagnall schrieb:
whoever owns a Cisco phone and is unhappy about slow
firmware, incomplete XML support etc... should really have a
look at Sergio Chersovani's rewrite of chan-sccp!
Is there a good resource out there for people who don't have a lot of
experience with Cisco phones? I
Rich Adamson wrote:
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet... anyone seeing the same?
Mark just committed some changes in this area... it's possible those
changes unintentionally broke writing of SIP registry information into
On Sunday 30 October 2005 12:09, Rich Adamson wrote:
Just updated cvs-head this morning, and now on a 'stop now' and restart,
* doesn't know about the previously registered sip phones (as shown with
sip show peers) on fc3.
Once the phones register again, they can be called, but not until
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet... anyone seeing the same?
Mark just committed some changes in this area... it's possible those
changes unintentionally broke writing of SIP registry information into
the astdb.
Chris,
I wrote a post that contains the information (files) you need for the
asterisk tftpboot directory to load a 7960 Sip 7.5 image from the server.
See this post
https://sourceforge.net/forum/message.php?msg_id=3374221
As far as obtaining the SIP 7.5 software, I see it on EBay all the time
Stefan Gofferje wrote:
Hi folks,
whoever owns a Cisco phone and is unhappy about slow firmware,
incomplete XML support etc... should really have a look at Sergio
Chersovani's rewrite of chan-sccp!
Ok - I'll give it a go :) - Just one problem... My phones have been
converted to SIP - and
Hi,
I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune
utility, which solved my echo problems , my zttest results are low, but
no echo on ZAP lines...
Marco.
Chris Miller wrote:
Mojo with Horan Company, LLC wrote:
The recent suggestion on the list was to not use
Hi,
I'm running asterisk 1.0.9 with an FXO card. People can call me on my pstn line
and that gets transferred to my laptop (on 192.168.62.100). That all runs fine.
If, though, I want to dial out (a pstn line) I always get a call rejected: 404
not found error (in sjphone) and in the asterisk
Hello,
I have just acquired my first 7960 from a business sale. It
has already been preloaded with the 7.3 SIP image which works flawlessly with
my Asterisk box. I want to experiment with chan_sccp and therefore I would need
the skinny firmware = 7, I
guess. Could someone tell me an
Quoting Obelix [EMAIL PROTECTED]:
I did some searching and I found them in the Asterisk 1.09 Sounds distribution.
I simple searched google for asterisk pounds.gsm. So much for forgetting about
the obvious.
Silly me :-)
Is there a .gsm file for announcing UK pounds and pence after the credit
Wayne schrieb:
Stefan Gofferje wrote:
Hi folks,
whoever owns a Cisco phone and is unhappy about slow firmware,
incomplete XML support etc... should really have a look at Sergio
Chersovani's rewrite of chan-sccp!
Ok - I'll give it a go :) - Just one problem... My phones have been
Paul schrieb:
Chris,
I wrote a post that contains the information (files) you need for the
asterisk tftpboot directory to load a 7960 Sip 7.5 image from the server.
See this post
https://sourceforge.net/forum/message.php?msg_id=3374221
As far as obtaining the SIP 7.5 software, I see it on
Bobby Lacey schrieb:
Hello,
I have just acquired my first 7960 from a business sale. It has already
been preloaded with the 7.3 SIP image which works flawlessly with my
Asterisk box. I want to experiment with chan_sccp and therefore I would
need the skinny firmware = 7, I guess. Could
In article [EMAIL PROTECTED],
Mohamed Farid [EMAIL PROTECTED] wrote:
I wonder if the Announce feature is available in Asterisk MeetMe Room ?
Announce Feature is the feature which play some kind of announcement
when someone joins the MeetMe room ...
It's the 'i' option to MeetMe (CVS-HEAD /
On Oct 30, 2005, at 1:39 AM, Ryan wrote:
On Sat, Oct 29, 2005 at 10:58:22PM -0600, Ryan exclaimed:
snip
The problem is caused on the provider side, not in your asterisk. The
packets they are sending to your asterisk have the wrong sequence
numbers. I finished some code lastnight to *help*
I am running CVS HEAD. How can I tell which software echo canceller I am using?On 10/28/05, Matthew Fredrickson
[EMAIL PROTECTED] wrote:On Oct 27, 2005, at 12:38 AM,
[EMAIL PROTECTED] wrote: My question is, what is the direction in relation to analog boards and such?Right now, it looks like the
On Sunday 30 October 2005 16:08, Eric Bishop wrote:
I am running CVS HEAD. How can I tell which software echo canceller I am
using?
Look at zaptel/zconfig.h and see what is uncommented.
e.g.
/*
* Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :)
*
*/
/* #define ECHO_CAN_STEVE */
Getting chan_sccp from Sergio to work is really easy. The
distro contains a well documented sample config and - as I
wrote before - there are lots of info in the chan-sccp-users
mailing list archive.
Yep, I've just tried chan_sccp with the 7960 I have here and it appears to
work fine
Can I please get a definitive answer on this from the list...
I have read through lots of material, and I am 90% sure the answer is
no, but the occasional offhand statement has left this little
nugget of hope in my mind.
Is there any echo cancelation that I can enable or add for pure
I've seen very small echo on two boxes connected with IAX2 and SIP phones.
Polycom 500's.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Grey
Sent: Sunday, October 30, 2005 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 10/30/05, Leif Madsen [EMAIL PROTECTED] wrote:
On 10/30/05, David Bandel [EMAIL PROTECTED] wrote:
Have the OReilley book. Also the new 1.2 book from asteriskdocs.org.
Pt... they're the same book :)
OK, well, I have two and they are definitely different books. For one
thing, one has
On Sunday 30 October 2005 16:45, Dave Grey wrote:
Is there any echo cancelation that I can enable or add for pure
IPIP calls?
No. There is no need because IP--IP calls are what is known as four wire
circuits -- there is no mixing of the received audio and the send audio, and
thus zero need
Hi Everyone,
I have a problem withasterisk-at-home beta 4.
Whenever I do an attended transfer (softphone or IP phone),once the 2
parties have finished talking the asterisk switch reboots with the following
error;
usr/sbin/safe_asterisk:line 42:14265 aborted
${astsbindir}/asterisk
Sorry for the top post.
If you have three remote offices and can control the routers, use a hardware
vpn router. I use netgear FVS318's and FVL328's. They are inexpensive and
functional for small offices and your server is not exposed.
Paul
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Look at zaptel/zconfig.h and see what is uncommented.
[...]
I am using the KB1 echo canceller.
The 'new' echo canceller is MG2. Please test. This is _extremely_ good
on PRI, but it's been reported to have some issues on 2 wire circuits.
--Rob
___
I would also be very interested in what hardware you used.
On 10/29/05, Robert Augustyn [EMAIL PROTECTED] wrote:
Darren,Can you elaborate on what echocan did you use and how?Thanks.robert -Original Message- From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Darren Wright
Chris Bagnall schrieb:
Getting chan_sccp from Sergio to work is really easy. The
distro contains a well documented sample config and - as I
wrote before - there are lots of info in the chan-sccp-users
mailing list archive.
Yep, I've just tried chan_sccp with the 7960 I have here and it
Our configuration is as follows:
SIP phones - TE410P - PSTN
When a SIP handset makes a call to other ISDN numbers - no problem.
When a SIP handset make a call to analogue numbers - echo.
I know for certain that the problem is at our end. Why?
If I call our Asterisk box via Disa and then place
We can get it for you.Contact me offlist.
-Mike
Michael Crown Managing Partner
www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED]
From: Bobby Lacey
[mailto:[EMAIL PROTECTED] Sent: Sunday, October 30, 2005
3:15 PMTo: asterisk-users@lists.digium.comSubject:
On Oct 30, 2005, at 4:56 PM, Andrew Kohlsmith wrote:
On Sunday 30 October 2005 16:45, Dave Grey wrote:
I frequently get bad echo on IAX2/ulaw connections between my CVS
HEAD and a friend's 1.2 Beta, using analog phones connected through
IAXys on both ends. Neither PBX has a zap interface. We
On Sunday 30 October 2005 18:25, Dave Grey wrote:
Could you please point me to further info on echo cancellation in the
IAXy? Is it configurable, or just an as is and always on aspect
of the firmware? The IAXy documentation (such as it is) available
from Digium does not mention echo
OpenVPN is pretty great I have learned (also free).
Sorry for the top post.
If you have three remote offices and can control the routers, use a
hardware
vpn router. I use netgear FVS318's and FVL328's. They are inexpensive
and
functional for small offices and your server is not exposed.
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
I noted this on Friday. I don't think I had problems using the devices
but sip show peers took some time to show the registrations.
MARK.
Kevin P. Fleming wrote:
Rich Adamson wrote:
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet...
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in
Look at zaptel/zconfig.h and see what is uncommented.
[...]
I am using the KB1 echo canceller.
The 'new' echo canceller is MG2. Please test. This is _extremely_ good
on PRI, but it's been reported to have some issues on 2 wire circuits.
The two-wire (eg, TDM card) version with MG2 seems
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet... anyone seeing the same?
Mark just committed some changes in this area... it's possible those
changes unintentionally broke writing of SIP registry information into
the
Hello,
I am wondering if it is possible to get Asterisk to distinguish between
the situation where you place a call to a PSTN line via a SIP telephony
provider and nobody answers, and where you place the same call but the
line is busy.
Watching the Asterisk console on verbosity=5 reveals that
ok.. followed
instructions with the apps but I keep getting this error:
[app_rxfax.so]Oct 30 21:08:48
WARNING[15290]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
fax_set_phase_d_handlerand
[app_txfax.so]Oct 30 21:09:16
WARNING[15317]:
On Sunday 30 October 2005 21:12, Rusty Dekema wrote:
Watching the Asterisk console on verbosity=5 reveals that Asterisk receives
a SIP 486 Busy Here when the called line is busy, and says Nobody picked
up in 45000 ms when nobody answers (with the SIP dial timeout set to 45
sec). However,
David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
Hi,
Am having a problem with Zap group channels. Wondering if anyone has
seen the same and if there is a quick solution.
I have bonded multiple zap channels to a group for outbound calling,
e.g. channels 3 and 4 are in Zap/g2.
We have an intermittent problem with our PSTN provider in
has anyone had any luck connecting * to IPOFFICE via h323 trunk
I can call * from IPO but don't get a connection the other way
the * box is sending packets to the ipoffice I see the Call hit the IPOFFICE
as an H323 event but it doesn't actually connect a call
thanks
On 10/31/05 05:56 Andrew Kohlsmith said the following:
No. There is no need because IP--IP calls are what is known as four wire
circuits -- there is no mixing of the received audio and the send audio, and
thus zero need for an echo canceller. You only need echo cancellation when
you go
On Sun, Oct 30, 2005 at 04:03:34PM -0500, Dave Grey exclaimed:
Hey, this is great. I'm running * on a mac, though, so I can't apply
it. Ah well, my loss.
The thing about the problems I am experiencing, though, is that they
are not in connections from other * servers. I am getting this on
On Oct 30, 2005, at 10:12 PM, Dinesh Nair wrote:
On 10/31/05 05:56 Andrew Kohlsmith said the following:
No. There is no need because IP--IP calls are what is known as
four wire circuits -- there is no mixing of the received audio
and the send audio, and thus zero need for an echo
On Sun, 2005-10-30 at 21:13 -0700, Ryan wrote:
I don't use IPKall but it is likely to be the same issue if your using
SIP to connect to them. I have the same issue with telasip.
If you contact IPKall and have them change you to IAX it may resolve the
issue I've never dealt with them so
On Oct 30, 2005, at 11:13 PM, Ryan wrote:
On Sun, Oct 30, 2005 at 04:03:34PM -0500, Dave Grey exclaimed:
The thing about the problems I am experiencing, though, is that they
are not in connections from other * servers. I am getting this on
SIP connections from IPKall. Is it still likely
://lists.digium.com/pipermail/asterisk-users/attachments/20051030/45c8e619/attachment-0001.htm
--
Message: 8
Date: Sun, 30 Oct 2005 11:57:37 -0500
From: Leif Madsen [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
To: Asterisk Users Mailing List
On Oct 30, 2005, at 11:21 PM, trixter aka Bret McDanel wrote:
I am starting a provider that gives free service
Nice! You went from listing free providers to becoming one. ;-)
I would love an invite if you could use another tester.
Either way, good luck with this and thanks for providing
On Sun, Oct 30, 2005 at 08:21:42PM -0800, trixter aka Bret McDanel exclaimed:
snip
Odds are they wont, they only recently let you use direct from them to
you via SIP rather than going only through free world dialup. They are
clear on their page they are free and they dont give support for their
On Sun, 2005-10-30 at 23:42 -0500, Dave Grey wrote:
On Oct 30, 2005, at 11:21 PM, trixter aka Bret McDanel wrote:
I am starting a provider that gives free service
Nice! You went from listing free providers to becoming one. ;-)
I had been planning on offering free service for a while,
On Sun, 2005-10-30 at 21:43 -0700, Ryan wrote:
As I am starting a provider that gives free service I agree with this,
you do actually get what you pay for. The only difference is that I am
doing outbound (right now anyway) and have about 30 countries.
http://www.trxtel.com/ - its in beta
My objective is to write an AGI Script to monitor channel status on an
originated call prior to passing it to a queue.
Current approach:
1. Originate via AMI...
set msg(Channel) Local/[EMAIL PROTECTED]/n
set msg(Exten) 0021$numberDial
set msg(Account) $agentid
Hi all,
I currently have this configuration.
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060)
exten = _X.,102,Set(PRI_CAUSE=42)
exten = _X.,103,Hangup()
I have an Asterisk box connected via E1/PRI to Siemens PBX. The siemens
PBX sends my Asterisk box all cell phone call, and from the
Hi the list,
You can check your IP-phone handset by removing it during a call and
listening from the other phone set.
If the echo disapear, it is the fault of your disconnected phone handset.
Goof luck !
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL
Am having a problem with Zap group channels. Wondering if anyone has
seen the same and if there is a quick solution.
I have bonded multiple zap channels to a group for outbound calling,
e.g. channels 3 and 4 are in Zap/g2.
We have an intermittent problem with our PSTN provider in
hello evryone
can somone help me get asterisk to work with outgoing calls to
a voip operator
i have tried many stings, but i cant triger the outgoing
calls, calls on the same pbx are working fine
what did i mis out ?
in advance thanks :)
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