I am having an odd problem that started somepoint in the last couple
days with no known config change. Asterisk will receive RTP data but
will not send it.
If someone calls my asterisk box, it will hang on any Playback() or
Background() call. No data is ever sent on the RTP stream, verified
Are the calls coming from SIP or PSTN?
- Original Message -
From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 3:56 PM
Subject: [Asterisk-Users] RTP not being sent by asterisk
Does maximum debugging show anything?
- Original Message -
From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 3:56 PM
Subject: [Asterisk-Users] RTP not being sent by asterisk
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:
Are the calls coming from SIP or PSTN?
from sip, and I can see packets going from sip - asterisk just nothing
outside of sip going from asterisk - sip phone.
Its like there is a blocking issue, although I dont know why this would
have
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:
Are the calls coming from SIP or PSTN?
further investigation shows the first RTP packet is sent and nothing
after that.
btw everything is on localnet and the IPs are all correct.
--
Trixter http://www.0xdecafbad.com
UK +44 870 340
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:
Does maximum debugging show anything?
I have -v'ed about to level 26 and it shows nothing other than:
-- Executing BackGround(SIP/trixter-03c5, beep) in new stack
where it hangs.
--
Trixter http://www.0xdecafbad.com
UK +44 870 340
From my understanding, * uses the incoming RTP stream itself as a
timing source for sending it's outgoing stream, hence the reason *
doesn't like/support silence suppression.
In other words, if there's no RTP headed back to *, then it won't send
anything.
(Someone please correct me if I'm
Hello,
I have a strange problem whenever I have 2 or more BT100s behind NAT. I am
not able to reproduce this error reliably, but it happens every 2-5 minutes.
The general setup is that there is Asterisk server sitting at a central
location. Some peers connect directly (206,205,201) but some
Hello list,
I am glad to announce that it is now possible to try XC-AST, the queue_log
file analyzer implementing most call centre metrics for the app_queue,
using a demo password.
See http://demo.xcept.it/xc-ast/xcast-live.jsp
Some people complained that it was quite too complex to set up a
I have done some further research, the first RTP packet is sent when
playback() is called. No others. The application is running, if I
press a key and goto a different item that would cause a new
playback()/background() 1 more RTP packet is sent.
To be clear If I call myself, RTP packets are
Boris,
Thanks for sharing your results.
I had to upgrade to image P0S3-06-3-00 first. Once I was at this level,
I could then upgrade to image P0S3-07-4-00. I WAS aware of the
differences in spelling in the OS79XX.TXT file and SIPDefault.cnf and
KEPT the file names DIFFERENT according to a
Hi,
try this...
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html
domé
--- urban viking [EMAIL PROTECTED] wrote:
What are the gcc and lib requirements to compile
oh323 channel (version
0.6.5) ?
your 2 cents are welcome .
Many thanks,
I am unable to
Title: [Asterisk-Users] Problem compiling 2nd AVM Fritz
I am having the exact problem. I managed to get to only 1 error by making sure the paths were correct. But yesterday 11PM the achine froze. Only this morning i will find out whats wrong.
Robson
Shane Dalgleish asterisk at
Hi.
I have lousy programming abilities, but put this together this evening. It's
a tcl script using Expect that dials a number on a Cisco 7960 from the
command line. Note you need to have privileged set on telnet_level in your
TFTP config files.
Why is this necessary, you might ask? I find it
What do i have to confiure so that a call comming in the * server through
chan_capi recognizes a normal busy line beep if the SIP phone is busy?
Kib
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What do i have to confiure so that a call comming in the * server through
chan_capi recognizes a normal busy line beep if the SIP phone is busy?
Presumably you have to fix the code and recompile chan_capi.
I did try the same without any success, I'm shure that it's an
chan_capi bug.
The only
On Thu, 2005-04-14 at 08:11 +0100, Michael West wrote:
Boris,
Thanks for sharing your results.
I had to upgrade to image P0S3-06-3-00 first. Once I was at this
level,
I could then upgrade to image P0S3-07-4-00. I WAS aware of the
differences in spelling in the OS79XX.TXT file and
hi!
I wanted to ask if someone ever got the error flexible rate not heavily tested
I am not able to dial from PSTN to iaxphones(on which agents are
logged in)...I have been successfully running this for some time
now..but today all of a sudden i got this error and I can`t get
connected to the
Good day all
I just want to know if someone tried this and with out any hassles
What I want to do is take 4 extension(analog) of a current,old,pabx unit
and put them into a asterisk server with a 4port analog card,like the
voicetronix openline4 card.
(PSTN)(old PABX)---===(4 ports
Elmar,
I tried the config from Damian and works for me. The only problem is
that it is not a traditional german busy tone but an american one.
Maybe there is also a solution to this.
Kib
Elmar Haneke wrote:
What do i have to confiure so that a call comming in the * server
through
chan_capi
On Thu, 14 Apr 2005, Altus Snyman wrote:
Good day all
I just want to know if someone tried this and with out any hassles
What I want to do is take 4 extension(analog) of a current,old,pabx unit
and put them into a asterisk server with a 4port analog card,like the
voicetronix openline4
On 13 Apr 2005, at 05:42, Brian Capouch wrote:
Rick from Digium got published yesterday.
http://www.nwfusion.com/news/tech/2005/041105techupdate.html?
nltcode=nltechupdate1476
Note that a newer version of the IAX RFC is imminent.
If they are planning to go through the full RFC process, I may
be
Hi,
I found the following from the wiki:
**
HylaFax and Asterisk
Another solution is the Hylafax
software. capi4hylafax and chan_capi will gladly coexist. You just tell
asterisk to ignore the DIDs that are used for fax.
My question: How can I tell * to ignore special DIDs and let them
On Wed, Apr 13, 2005 at 02:45:18PM -0700, Brian Leyton wrote:
I installed a couple of Asterisk test machines, and have been successful in
getting them talking to one another, but I have question.
After installation, I put an x100p clone in one of the machines. From what
I understand, I no
Hi Nabeel,
I also wrote a siliar script using the same tools, I found I still had a few
problems with it (Im also a terrible programmer) and dont use it anymore.
However, on a windows system you can try the following 2 programs and get
integration with outlook and/or cut and paste dialing.
hi!
I wanted to ask if someone ever got the error flexible rate not heavily tested
I am not able to dial from PSTN to iaxphones(on which agents are
logged in)...I have been successfully running this for some time
now..but today all of a sudden i got this error and I can`t get
connected to the
Hi Eric
I started getting these 'WARNINGS' (WARNINGS are not ERRORS) when I started
using the manager interface (IPSwitchboard / AstWinManager). I sometimes
also get some extra delay when I see these but havent determined if its
related.
If I disconnect my manager program the warnings
hi!
I have already memtioned the error i am getting on my cli ieflexible
rate not heavily tested.
I am also getting a warning ,something like broken -pipe
I need urgent help plz reply
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Sounds like a message from mpg123,
Is your asterisk crashing when this happens or are you just giving the
wrong input files for moh ?
Zoa.
amna saleem wrote:
hi!
I wanted to ask if someone ever got the error flexible rate not heavily tested
I am not able to dial from PSTN to iaxphones(on which
Hello, all.
I wonder if there is a SIP stack within asterisk. I
need to build my own application based on a SIP stack. I don't know if
asterisk is able to help me on this. Could anyone give some ideas?
Best regards,
Abe
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Is there any better docs or step by steps other than what's in the Wiki for
Realtime setup?
We have been trying to get this running and it's driving us batty..
It seems that the switch command is totally being ignored as far as we can
tell.
We are basically just getting an error telling us
Good Evening,
Just a quick question to ask if blind transfers (via #) are
possible? I have an IAX2 connection to my VOIP provider. In my dial plan I
sometimes forward an incoming call back out on IAX, but when this happens I
seem to lose the ability to transfer the call. If the
Good day all
Will a voicetronix openline 4 card work with a 4port BRI card?
Please HElp/advice
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Title: Message
Guys,
where
, can I register to this forum ?
Rene
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
GeisSent: Tuesday, April 12, 2005 3:05 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] multiple
In what sense ? voicetronix is analog BRI is ISDN digital
On 4/14/05, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
Will a voicetronix openline 4 card work with a 4port BRI card?
Please HElp/advice
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On Thu, 14 Apr 2005, Abraham WEI wrote:
Hello, all.
I wonder if there is a SIP stack within asterisk. I need to build my own
application based on a SIP stack. I don't know if asterisk is able to help
me on this. Could anyone give some ideas?
Hi,
There is a SIP stack in Asterisk - but
On Thu, Apr 14, 2005 at 01:51:51PM +0500, amna saleem wrote:
hi!
I have already memtioned the error i am getting on my cli ieflexible
rate not heavily tested.
I am also getting a warning ,something like broken -pipe
I need urgent help plz reply
If you need urgent help, I humblly suggest
If you either e-mail me your current .conf files, or post your .conf files
on the web somewhere I will take a look. I hope to release a web
interface to the realtime database on Sunday at 1900BST.
I will test your .conf files with a scratch database, and then come back
to you. Would you also
On Thu, 14 Apr 2005, Paul Seymour wrote:
Just a quick question to ask if blind transfers (via #) are possible? I
have an IAX2 connection to my VOIP provider. In my dial plan I sometimes
forward an incoming call back out on IAX, but when this happens I seem
to lose the ability to transfer
I know that the syntax for registration is
register = user[:secret[:[EMAIL PROTECTED]:port][/contact]
but then most of the TSP allow me to register without specifying the contact field. This allows me to use the s extension to process all incoming calls. Now, with this specific TSP
Voicetronix will only be used for the gsm cell router and BRI for
outgoing-incoming calls
On Thu, 2005-04-14 at 11:26, Michael Bielicki wrote:
In what sense ? voicetronix is analog BRI is ISDN digital
On 4/14/05, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
Will a voicetronix
[EMAIL PROTECTED] wrote:
On Thu, 14 Apr 2005, Paul Seymour wrote:
Just a quick question to ask if blind transfers (via #) are possible? I
have an IAX2 connection to my VOIP provider. In my dial plan I sometimes
forward an incoming call back out on IAX, but when this happens I seem
to lose the
On Thu, 14 Apr 2005, Altus Snyman wrote:
Voicetronix will only be used for the gsm cell router and BRI for
outgoing-incoming calls
Hi Altus,
They will probably work OK together. I guess you'll want to test it and
find out. I understand that the Voicetronix boards use their own channel
Hi all,
I am writing a program (using version 1.05 and 1.07 of asterisk) which
accepts an incoming call, plays some messages, accepts some DTMF digits
(which make up an outgoing phone number) and then dials the provided
number. However, the problem I am suffering is the loss of DTMF digits when
I have an Asterisk system that seems to be randomly dropping calls. The
system is currently running on a Junghanns octo-bri card and one Digium
TDM400b card. The calls seem to be dropped during peak call hours and I
assume that it probably has something to do with congestion (this box
On Thu, Apr 14, 2005 at 10:06:37AM +0800, Eddie wrote:
I cannot dial two phones using zap at the same time.
One will ring but the other one hangs up.
Are those phones on an FXS or through an FXO to a PSTN to an outside number?
zapata.conf
[channels]
context=default
I tried the config from Damian and works for me.
I do not intend to use an Voicemail here.
The Initial Answer results in the caller to be charged for the call
even if no connection is made. That's not really MY problem, but I
would prefer to have an correct solution.
Elmar
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They can, its called cross-shipment, but
Selon Rich Adamson [EMAIL PROTECTED]:
Funny, they sell these old cards.. it seems like they are selling
refurbs
as new.. ... anyways RMA is on its way, would be nice if they would
send
one as a replacement first, so that we could continue our work and
don't
have to delay it.
Hi list!
I'm using bristuff on a box with one BRI line. I have echo cancel on.
However incoming faxes (which routed to a Sipura 2000) fail after about 2
pages and the fax will report a communication error.
I switched off echo cancel but ofcourse now I have an extremely annoying
echo.
Is there
there is an application in bristuff that lets you selectivelly turn
off echo cancellation for a call. Use that one for all fax calls :)
On 4/14/05, Remco Barende [EMAIL PROTECTED] wrote:
Hi list!
I'm using bristuff on a box with one BRI line. I have echo cancel on.
However incoming faxes
On Thu, 14 Apr 2005, Paul Seymour wrote:
Just a quick question to ask if blind transfers (via #) are
possible? I
have an IAX2 connection to my VOIP provider. In my dial plan I
sometimes
forward an incoming call back out on IAX, but when this happens I
seem
to lose the ability to
Note: forwarded message attached.
__
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
http://smallbusiness.yahoo.com/resources/ ---BeginMessage---
Hi,
We have configured Asterisk to allow calls among IAX
clients and a GSM
On Thu, Apr 14, 2005 at 06:04:20AM -0500, [EMAIL PROTECTED] wrote:
I'm using bristuff on a box with one BRI line. I have echo cancel on.
However incoming faxes (which routed to a Sipura 2000) fail after about 2
pages and the fax will report a communication error.
I switched off echo
It depends on whether Sprint is the local exchange carrier. If they
are, they will install the inside wiring and jacks (probably for a
price). If you don't ask, they probably will not install them. It
really is phone company dependent; some do, some don't, and some
charge if its part of the
bristuff does not work with head
On 4/14/05, Edwin Groothuis [EMAIL PROTECTED] wrote:
On Thu, Apr 14, 2005 at 06:04:20AM -0500, [EMAIL PROTECTED] wrote:
I'm using bristuff on a box with one BRI line. I have echo cancel on.
However incoming faxes (which routed to a Sipura 2000) fail after
On Thu, 14 Apr 2005, Michael Puchol wrote:
Thanks for that explanation, it's very useful, I'm also having transfer
problems on IAX2 bridges. In my case, I have a remote * with a couple of
PSTN lines that are bridged over IAX2 to a local * which in turn handles
a bunch of SIP phones. When
Hello all,
I need to setup some dialing rules and I've had nothing but problems
with them.
I am using an IAX terminated service for my PSTN calls. To access the
service from my asterisk box I dial 6 then the number. The part I'm
having problems is that since the company is based in the US if
Looks normal to me.
What Dial with the '' means is that both lines ring, but the first one
to answer is connected on the call.
From you trace it looks like Zap/3-1 which is your number 206 answered
the call, so the other line goes to hangup.
The Dial with '' is used to implement call teams where
On Thu, 14 Apr 2005, Edwin Groothuis wrote:
On Thu, Apr 14, 2005 at 06:04:20AM -0500, [EMAIL PROTECTED] wrote:
I'm using bristuff on a box with one BRI line. I have echo cancel on.
However incoming faxes (which routed to a Sipura 2000) fail after about 2
pages and the fax will report a
My question: How can I tell * to ignore special DIDs and let them through
to Hylafax?
Take a look at capi.conf. There is a entry like incomingmsn. Modify it like
incomingmsn=12345,23456,34567
So you can define which msn are for asterisk and which msn is used for other
purposes.
HTH
Frank
On Thu, 14 Apr 2005, Paul Seymour wrote:
Thanks Steve, have done as you suggested and it works perfectly. Would
this be considered a bug since the T or t directive in the dial plan
probably should preclude native bridging if the end result is to prevent
a transfer?
Yeah - I think I'd
I am having issues using realtime, and it is my understanding that you
still have to perform a reload for the changes to take effect?
No, I am using realtime for everything, basically nothing in
*.conf's, and you dont need a reload for things to work thru realtime.
Does a sip reload
Kib Eki wrote:
My question: How can I tell * to ignore special DIDs and let them
through to Hylafax?
Don't put them in /etc/asterisk/capi.conf as incomingmsn. You can still
use them for outgoing if you want to though.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
Mystery Glitch wrote:
In my [incoming] context I have something like this:
exten = 8885861575,1,Macro(vrforward,${EXTEN},8136361451)
And thie Macro contains this:
[macro-vrforward]
exten = s,1,GotoIF($[${CALLERIDNUM} = 954555]?40:2)
exten = s,2,SetGroup(${ARG1})
exten = s,3,CheckGroup(3)
exten
Can you capture Ethernet traffic with ethereal or similar tools and show
what is happening?
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Thursday, April 14, 2005 1:56 AM
To: asterisk-users@lists.digium.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all, i installed [EMAIL PROTECTED] v0.8, very clean install (great
piece of software!).
I have successfully configured incoming calls, but i still have some
problems.
My setup is a X100 connected to a POTS, one analogic phone, 2 SIP's
(X-Lite) in
I have seen the same problem as well. If don't think this is a problem with
BT100. I think the problem is with public STUN server. I think sometimes,
the server is too overloaded and can't provide the translation. That is when
you are getting the problem with your clients behind NAT. the only
This is the asterisk output:
-- Executing Answer(SIP/202-8236, ) in new stack
-- Executing Dial(SIP/202-8236,
SIP/203|100|tTr) in new stack
-- Called 203
-- SIP/203-3c5d is ringing
-- SIP/203-3c5d answered SIP/202-8236
-- Attempting native bridge of SIP/202-8236 and
The demo does not seem to be working, I am doing something wrong. It is
constantly complaints that file placed in 'File' field is not found. Please
let me know how to resolve this.
Thanks
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent:
trixter http://www.0xdecafbad.com wrote:
I have done some further research, the first RTP packet is sent when
playback() is called. No others. The application is running, if I
press a key and goto a different item that would cause a new
playback()/background() 1 more RTP packet is sent.
To be
This message is to announce a bounty for the following:
If ztdummy is already loaded, generate an error to the console and
syslog when modules for Digium cards are loaded.
If a modules for a Digium card are already loaded, generate an error
to the console and syslog when ztdummy is loaded.
You
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
The bounty is US$10 and will be paid via Paypal. The patch must be
accepted into CVS-HEAD before the bounty will be paid.
--Eric
--
Always do
On Wed, Apr 13, 2005 at 05:10:06PM -0400, Michael George wrote:
I am working on my dialplan, and I have come across many cool uses of DISA()
internally to generate dailtone at specific places where I want it. Works
quite well.
However, now I'm adding stuff to the dialplan that requires me
On Thu, Apr 14, 2005 at 07:28:44AM -0500, Eric Wieling wrote:
This message is to announce a bounty for the following:
If ztdummy is already loaded, generate an error to the console and
syslog when modules for Digium cards are loaded.
Asterisk is normally not running when modules get loaded.
Manish Sapariya [EMAIL PROTECTED] writes:
Hi,
I was going through some of the list postings...and I felt
like if want to do voip within a LAN, I might have to install
Asterisk on every machine.
I hope it is not the case.
What I understand is (or what I want is)
- Install asterisk on one
More bug fixes. *69 works now. Cisco stuff works. Lots
of other fixes.
is phpconfig fixed ?
when editing a file, it doesn't show the list of sections, it only list Header
What needs to be modified :
In the function OC_readConfFile around line 131 change :
$this-_OC_the_file[] = fgetc($file);
Here is my situation:
- running [EMAIL PROTECTED] 0.8
I can do the following:
- Make phone calls to POTS lines using my termination company (VoIPJet)
- Make calls to other users on other asterisk servers
- Receive calls from users on my asterisk server
- Receive calls from users on other asterisk
--- Qiao Yuansong [EMAIL PROTECTED] wrote:
At the beginning of a call, the latency is not very
long, but it becomes more and more obvious along
with time. If the call keep 10 minutes, the delay
will be about half or one second.
Anyone knows the reason, and any suggestion?
Are you running
I have a Sipura SPA-3000 for access to my standard analog PSTN line. I
have the SPA-3000 answering and then directing all calls into Asterisk.
This setup is working fine for everything except voicemail. Most, about
2/3 or so, of messages left come across very quiet when the voicemail is
Ben probably better to post to the [EMAIL PROTECTED] sourcforge list seeing
this is a specific [EMAIL PROTECTED] problem, having said that please post a
console
output of the problem occurring.
If you don't know how to do this please email me and I'll explain how to
use putty to view the
On April 14, 2005 08:31 am, Eric Wieling wrote:
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure
the dialplan
There have been a lot of bug fixes because of the change to the event driven
model.
Download here: http://ipswitchboard.thorben.dk
IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a
FREE
Andrew Kohlsmith wrote:
On April 14, 2005 08:31 am, Eric Wieling wrote:
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure
I was following a discussion on this list about the TDM400P
revisions.
It is my understanding that the current revision that one
should have
is the Rev. H and not the E/F. I have not yet been able to
verify the
rev stamped on the board, but zaptel is reporting that I
have
I've found that a TDM400P card in our * box is sharing IRQ's with two other
devices. The server doesn't support assigning IRQ's through the BIOS and the
pig only has three PCI slots, so swapping cards between slots hasn't fixed the
problem (it just ends up sharing IRQ's with other devices).
i have asterisk on my system and when making a call a
delay problem in talking appears,that means when i
talk to somebody he will listen me after almost a
second (the ping on my voip provider's IP is 700ms to
800ms)so i dont know if the problem is in the nternet
connection or another problem
Please ignore
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On April 14, 2005 09:42 am, Eric Wieling wrote:
exten = h will not be called unless the channel has ALREADY hung up.
I understand that, which is why I'm still suggesting a WARNING and not an
error.
Something like No need to execute Hangup from the h exten, line is already
hung up
-A.
On April 14, 2005 10:44 am, Rich Adamson wrote:
Sounds like its time to swap motherboards. :(
I just wish that the PCI bridges on the TDM and TExxx cards would allow you to
utilize INTA,INTB,INTC or INTD... if the mobo's fucked up at least let the
card route around the damage. I'm not sure
I have asterisk 1.0.6 with cisco 7912/7960 phones (sip) and a isdn card
with capi drivers, everything works fine, except for music on hold, even
when you transfer a call (which is the most annoying part, since the
caller thinks the line is down and hangups).
With transfer I don't mean direct
Andrew Kohlsmith wrote:
On April 14, 2005 09:42 am, Eric Wieling wrote:
exten = h will not be called unless the channel has ALREADY hung up.
I understand that, which is why I'm still suggesting a WARNING and not an
error.
Something like No need to execute Hangup from the h exten, line is
Hello,
I can not find anything on this, so it may not be possible.
I would like to dial one number which then rings at least two extensions
at the same time. Not a hunt group, but ringing at the same time as if
they were plugged into the same physical port.
Does anyone know if this can be
Quoting Julien Goodwin [EMAIL PROTECTED]:
My biggest task is getting in some of the big bugfixes and bad behavior
fixes that have been major issues. In testing at the moment is a fix to
allow speeddials to work at any time (meaning you could in theory create
a speeddial that auto-navigated a
cool thanks for the update. next time please submit a
bug to the [EMAIL PROTECTED] source forge project. Then it
will get fixed in the next release. I had no idea this
was broken.
--- Time Bandit [EMAIL PROTECTED] wrote:
More bug fixes. *69 works now. Cisco stuff works.
Lots
of other fixes.
G.Marshall wrote:
Hello,
I can not find anything on this, so it may not be possible.
I would like to dial one number which then rings at least two extensions
at the same time. Not a hunt group, but ringing at the same time as if
they were plugged into the same physical port.
Does anyone know
I've had a question related to this: what's the deal with frame
slippage on the Digium TDM analog cards? What would cause this? How
can one correct for this? I've recently seen a bad buzz every 6
seconds or so, heard by callers when calls are bridged with my TDM card
analog phones.
Hi all,
I think I've seen this somewhere, but I can't remember where; Is it
possible to steal a call from a sip extension? Let me explain what we
are trying to do:
Parking calls is a good thing, but having to remember an extension may
be a bit much to ask my user base who is used to seeing
I currently run an asterisk server with cvs from May 2004. I'm planning
to upgrade to the latest stable version but want to segregate a test
version first. I know I can do this by editing the install_prefix field
in the makefile.
I can also change the install prefix and load the new zaptel
Apr 14 08:52:44 NOTICE[13947] app_dial.c: Unable to create
channel of type 'ZAP' (cause 0)
Apr 14 08:56:52 NOTICE[13947] app_dial.c: Unable to create
channel of type 'ZAP' (cause 0)
I just started seeing this anytime I try to call out on my TE110XP.
I can still receive calls, but no
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