Hello,
I have several asterisk 1.2.7.1 servers connected through iax2 and often
the local asterisk would no longer see the remote one, even thought the
link is high quality and the ping is perfect.
Is there some issues to take into account about IAX2 connections?
Is asterisk's DNS resolution
On Mon, Apr 24, 2006 at 07:21:18AM +0800, Leo Ann Boon wrote:
> Louis-David Mitterrand wrote:
>
> >Should I use a T1 cross cable to connect the telco's socket to the
> >TE410P card?
> >
> >When I tried straight cat5 cables, both leds remained red at each en
On Sat, Apr 22, 2006 at 11:59:21AM -0400, Alexander Lopez wrote:
> Can't anyone stop self-promotion and tell the poor guy what he needs.
>
> A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows:
>
> 1 - 4
> 2 - 5
> 3 - NU
> 4 - 1
> 5 - 2
> 6 - NU
> 7 - NU
> 8 - NU
>
> NU = Not
On Sat, Apr 22, 2006 at 08:09:13AM -0700, Paul Mahler wrote:
> A T carrier cable is not the same as an ethernet cable. A T carrier
> cable uses a real metal shielded RJ-45 and loosely twisted pair wire.
> With most modern T carrier equipment, you can use a CAT-5 ethernet
> cable instead of a
Hello,
I am about to put an asterisk server between the telco E1 and our old
Matra PBX.
Should I use an ethernet cross cable? Something else?
Thanks,
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Asterisk-Users mailing list
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Hello,
I just received what seems to be a nice SIP<->DECT gateway but can't
make it work with asterisk. The manual is very unclear (written in
"chinese" english) and the web configurator is ambiguous as well.
Has anyone succeeded in making one of these babies work with * ?
info:
http://www.
Hello,
I just acquired a used Cisco 7920 wi-phone and it mostly works with
the newest asterisk and chan_sccp, but it reboots after most calls.
Would a kind soul send me the latest firmware for that phone?
Thanks in advance,
___
--Bandwidth and Colocat
Hello,
Has anyone used Polycom's VSX line of videoconferencing equipment with
Asterisk?
It seems some of their models, namely the newer VSX 5000, supports SIP.
--
The Internet used to be a lot of smart people sitting at dumb terminals,
but now its a lot of dumb people sitting at smart terminal
On Tue, Jan 17, 2006 at 06:07:27PM +0100, Karsten Wemheuer wrote:
> > Did I forget something in my conversion command?
>
> Are You using bristuff 0.3.0-PRE-1f?
Yes.
> I've had the same issue. Dan Austin
> wrote a notice in a mail on this list, which solved the problem.
>
> Configure the follow
Hello,
Using asterisk-1.2.1 I am trying to convert my music-on-hold files from
.wav to alaw:
% sox moh.wav -r 8000 -c 1 moh.al resample -ql
The file sounds fine when listened with:
% sox mox.al -t ossdsp /dev/dsp
But when listened through asterisk with an alaw SIP phone the sou
On Fri, Jan 13, 2006 at 02:03:20PM +0100, Armin Schindler wrote:
> On Wed, 11 Jan 2006, Louis-David Mitterrand wrote:
> > On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
> > > On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> >
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
> On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> > On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> > > [C:4] 22:0188:202 - D-X(003) 02 01 7F
> > > [C:4] 22:0189:202 - D-X
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
> On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> > On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> > > [C:4] 22:0188:202 - D-X(003) 02 01 7F
> > > [C:4] 22:0189:202 - D-X
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> > [C:4] 22:0188:202 - D-X(003) 02 01 7F
> > [C:4] 22:0189:202 - D-X(003) 02 01 7F
> > [C:4] 22:0190:202 - D-X(003) 02 01 7F
> > [C:4
On Mon, Jan 09, 2006 at 03:50:55PM +0100, Armin Schindler wrote:
> On Mon, 9 Jan 2006, Louis-David Mitterrand wrote:
> >
> > I am now using a cross cable and the green led lights up on the Diva
> > port when plugging the phone in.
> >
> > When dialing from the
On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote:
> On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
> > Hello,
> >
> > I just received a couple SX440isdn phones and was wondering if they can
> > be plugged into a Diva 4BRI port in NT mode and work
On Mon, Jan 02, 2006 at 01:01:44PM -0800, Mike Fedyk wrote:
> Administrator TOOTAI wrote:
>
> >Craig Guy a écrit :
> >
> >>Are you using raid for performance or redundancy? Software raid is nice
> >>except when the drive that fails is the one with your boot partition on it.
> >>I
> >>guess you
On Mon, Jan 02, 2006 at 11:25:02AM +0800, Craig Guy wrote:
> Are you using raid for performance or redundancy? Software raid is
> nice except when the drive that fails is the one with your boot
> partition on it. I guess you could always tftp boot the kernel or
> something.
On our raid1 machin
On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote:
>
> The 830s are nice but limited because they do RAID on a card and have but
> one suitable PCI slot. So you can have an interface card or RAID, but not
> both.
Linux software raid is, in our experience, much better than any hardwa
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?
I know they probably work fine with mutliHFC cards with either bristuff
of chan_misdn but since I have some spare Divas, I was curious about
Hello,
Using * 1.2.1 with chan_capi CVS on a Diva server I am mostly happy.
However when a phone redirects a call (user forward) and all ISDN
channels are busy, the call goes out through an IAX connection and it
takes a few seconds to get a "ring" state from the remote * server. This
makes the inc
On Fri, Dec 02, 2005 at 12:05:08PM -0600, Kevin P. Fleming wrote:
> Louis-David Mitterrand wrote:
>
> >to Asterisk Extension Language (AEL) style.
> >I haven't found anything in the docs, wiki or examples about it.
>
> I don't believe hints are supported in
Hello,
I am trying to convert my hint priorities from the old style:
exten => 2130,hint,SIP/0146472130
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
How should I do it?
--
Sigs have been known to cause cancer in California.
_
On Fri, Nov 18, 2005 at 03:30:32PM +0100, Leif Neland wrote:
> >The gxp-2000's lack of a dialplan (or did I miss it?) led me to
> >activate its "early dial" option to avoid pressing "Send" after
> >dialing. Thus the "dialplan" is controlled by asterisk.
> >
> >It creates an extension matching probl
Hi,
The gxp-2000's lack of a dialplan (or did I miss it?) led me to activate
its "early dial" option to avoid pressing "Send" after dialing. Thus the
"dialplan" is controlled by asterisk.
It creates an extension matching problem:
exten => _00[1-9].,1,Macro(dialcapi)
If I dial 0012 the extensio
Hi,
I'm trying to get a SPA-3000 to work with a Siemens Gigaset 3010 DECT
(cordless) phone. I tried every localization scheme I could find on the
Net, including the settings recommended by the Voxilla wizard.
This Gigaset works fine and rings when plugged directly into the telco's
analog phone
Hi,
I looked at the docs and probably missed it: is there a way to set a
dialplan on the GXP-2000? (to avoid having to press "Send")
Thanks,
--
"Computers are useless. They can only give answers." - Pablo Picasso
___
--Bandwidth and Colocation sponsor
On Fri, Sep 30, 2005 at 01:11:19PM +0200, Armin Schindler wrote:
> > Also, is there a way to detect that a SIP phone has an active forward
> > number and capi-deflect any incoming calls to that number?
>
> If you can retrieve this information from extensions.conf, then you can use
> my example ab
Hi,
I've recently reinstalled a Diva in my asterisk server (alongside a
QuadBRI :-) to test the nice features Armin has been adding in
chan_capi.
The capi.conf format has changed, so my question is how do I define a
deflect= statement for different incoming MSN's?
I've tried to define a section
On Thu, Aug 11, 2005 at 07:12:32PM +0100, Mark Thorpe wrote:
> I have been trying to solve a problem wherby when I boot a cisco 7920 my
> 7940 seeks a new IP and the dhcpd log shows it released its existing IP. In
> searching for the solution I notice there were 2 messages on this list in
> Aug &
Hello,
I'd like to find a way to probe a SIP phone for forwarding information
before I actually Dial() it. For instance, if an absent user entered a
forwarding number in his (Cisco or Polycom) phone, it will anwser a
Dial() with a REDIRECT and asterisk will comply if the context allows.
However I
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
> >With the 1.5.2 firmware, have you managed to get one-touch message access
> >when
> >pressing the "Messages" button? It worked for me with 1.4.1 but no longer
> >with
> >1.5.2: I have to go through the message count screen first.
>
Hello,
With the 1.5.2 firmware, have you managed to get one-touch message access when
pressing the "Messages" button? It worked for me with 1.4.1 but no longer with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
and in sip.cfg:
Have I forgot
Hello,
I have the exact same question as you. Did you find an answer?
> We are using asterisk at the office and the incoming line is an ISDN
> (HFC-PCI card with zap_hfc driver from bristuff 0.2.0 RC3a).
>
> And I have a problem, when both ISDN B channels are in use (i.e. 2
> calls in progress)
Hi Kape,
Life is generally good with bristuff and the quadBRI cards. However I've
got a concern: how does one return a busy signal to the telco when all B
channels are busy? Right now, when all channels are in use, the remote
caller is kept waiting until the telco times out and finally get a busy
Hi,
After upgrading to 1.5.2 I no longer can directly access to my voicemail
by pressing the "Message" button, I have to go through the
"urgent,new,old" report first. The oneTouchVoicemail parameter is set to
1 but not taken into account apparently.
Anyone noticed that problem?
_
On Fri, Jun 17, 2005 at 10:34:25PM +0200, Conrad Beckert wrote:
>
> ... probably one of those RTFM kind of questions (while I'd be happy to know
> where a good reference "FM" is :-) )
>
> Has anyone an idea on how to disable the console sound driver. My problem is
> that a running asterisk is
Hello,
I looked everywhere in the docs and in google but couldn't find an
answer.
Is it possible to localize the output of ${VM_DATE} (say, in french) ?
--
Only half the people in the world are above average intelligence.
___
Asterisk-Users mailing l
On Fri, Jun 03, 2005 at 02:39:48PM +0100, Gavin Hamill wrote:
> On Friday 03 June 2005 14:28, Nardis Dome wrote:
> > --- "Brett, Gary" <[EMAIL PROTECTED]> wrote:
> > > Is the Eicon that much better ?
> >
> > sorry, i have only experience with Eicon... maybe
> > someone else is able to give a feedba
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote:
> On Fri, 13 May 2005, Louis-David Mitterrand wrote:
> > On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
> > > On Fri, 13 May 2005, Paul Hales wrote:
> > > > I battled with chan_capi d
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
> On Fri, 13 May 2005, Paul Hales wrote:
> > I battled with chan_capi during the week, and it was not fun.
>
> Since I'm working on chan_capi, I would like to know what problems exist.
> Can you please be more specific on what proble
Please find attached a patch I made to app_queue.c to add distinctive
ringing support. So the following works:
exten => 2131,1,SetVar(ALERT_INFO=)
exten => 2131,2,Queue(standard|r)
I took code in app_dial.c and lightly adapted it.
I hope this gets included in * as it is really useful. I faxed m
Hi,
Is it possible to have distinctive ringing in a queue?
I've tried:
exten => s,2,SetVar(ALERT_INFO=)
exten => s,3,Queue(standard|r)
without success.
However the SetVar(...) works fine when just dialing a SIP device.
Any ideas?
___
Asterisk-User
On Fri, Apr 01, 2005 at 04:12:05PM +0200, Julius Vindex wrote:
>
> When I Dial(SIP/1&SIP/2&SIP/3) if any of these phones has a forward to
> another destination ("302: moved" response) then the simultaneous ring
> stops immediately and the incoming call goes to wherever the forward
> points to.
>
Hi,
When I Dial(SIP/1&SIP/2&SIP/3) if any of these phones has a forward to
another destination ("302: moved" response) then the simultaneous ring
stops immediately and the incoming call goes to wherever the forward
points to.
We are using simultaneous ringing as a fallback when the receptionist
Hi,
If several phones register to the same sip.conf section what will happen
with a "Dial SIP/shared" in asterisk?
All phones ringing and the first one to answer gets the call?
Undefined behavior?
Thanks,
--
Jesus is coming! Everyone look busy!
___
On Tue, Feb 15, 2005 at 08:26:42PM +1100, Adam Goryachev wrote:
> On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote:
> > You are right, this activity is related to logging.
> >
> > After consulting the admin manual I am unsure as to what settings
> > rel
On Tue, Feb 15, 2005 at 07:56:10PM +1100, Adam Goryachev wrote:
> On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote:
> > Hi,
> >
> > I am mostly happy with my Polycom IP600 but it apparently needs to check
> > the FTP server every minute. I couldn't
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
to that behavior in the configuration files.
Any idea how to curb the IP600's spurious network activity?
Thanks,
--
Lord, protect me from your f
On Tue, Feb 01, 2005 at 09:41:25AM -0500, Noah Miller wrote:
> >I faithfully followed the instructions from:
> >
> >http://www.voip-info.org/wiki-
> >Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
> >
> >but still the message waiting indicator doesn't flash when a message is
> >waiting. There
Hello,
I faithfully followed the instructions from:
http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
but still the message waiting indicator doesn't flash when a message is
waiting. There is a brief intermittent chirp but nothing more.
Using latest firmware 1.4.
On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote:
> If you have it, can I get a copy please, or possibly can you send it to the
> keeper of http://www.freedomphones.net/polycom/files/
> I am looking for the latest boot image too.
1) I have the 1.4.1 firmware. To whom should I send
Hi,
At the * console I periodically get these messages:
Dec 9 10:58:11 WARNING[-1248765008]: chan_iax2.c:5021 socket_read: midget
packet received (1 of 4 min)
Which seem pretty inocuous.
Google say (almost) nothing about that subjet.
What does it mean?
--
Field Artillery lends dignity to
On Wed, Aug 25, 2004 at 04:22:26PM -0700, [EMAIL PROTECTED] wrote:
> I finally have my 7920 working though I'm seeing this bizarre
> behavior. As soon as the 7920 boots and authenticates with the AP my 7960
> release's its ip.
Hi,
I have exactly the same problem. Have you found a solution or wo
On Tue, Sep 07, 2004 at 02:24:06PM +0100, Nick Barnes wrote:
>
> Hi all,
>
> I have just received the following e-mail from an Asterisk user:
>
> "I just made a call via BT to a mobile. Then an incoming call came in and
> Ann else answered it - it made my call go completely fuzzy and I could hea
Hello,
We have been using a Diva 4BRI with our Asterisk PBX through the capi
interface for almost a year now with good results. However, recently we
started to hear heavy clicking sounds in our phones when two
simultaneous incoming calls are processed by the card. The clicking does
not originate w
Hi,
I am in the process of setting up call forwarding through capiECT with
the 7960's CFwdAll button.
When the phone redirects the call to an outside number (through a 302
SIP redirect) then the CAPI[contr1/xxx] channel becomes Local/[EMAIL PROTECTED] as
the call is reinjected into the dialplan.
When I access voicemail remotely, from a gsm phone say, some extra
characters get inserted in my dtmf tones: when I type , *
understands 88f8f8 (it always seems to be 'f'):
-- Incorrect password '88f8f8' for user '2130' (context = )
And the 'f' always starts after the second digit. Might it b
On Sun, Jan 25, 2004 at 08:48:20AM +0100, Jan Czmok wrote:
> Hi all,
>
> we are happy to announce the new test-release of chan_sccp.
> The Cisco 7920 support is working now, however, some call handling stuff is
> hardcoded for testing.
Thanks a lot for this work. Indeed the wifone 7920 now sorta
On Mon, Dec 15, 2003 at 09:25:05AM -0500, John Todd wrote:
> Paul -
> Yes, your description is correct.
>
> - moving the phone (no ethernet passthrough) results in no symptoms
You might have a virus on that XP box that totally saturates the poor
7960 switch with bogus IP packets.
--
May the L
On Sat, Nov 29, 2003 at 11:28:56PM -0600, Mark Spencer wrote:
> I'm coming to Paris Dec 19. I was wondering if there was any interest in
> having an Asterisk get together in Paris sometime near there. Any one out
> there interested? Anyone in Paris who could help organize something like
> that?
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
>
> I second that, and I think I remember hearing Mark talking about it too. But.
>
> What type of encryption can you do that does not introduce latency?
>
> That said, I would like it to support hardware encryption cards.
>
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
> Hello,
>
> I have searched google, read everything on the mailing list, read
> /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
> the IRC channel and I cannot figure out what is wrong with my IAX2 trunk.
>
On Mon, Nov 03, 2003 at 11:32:11AM +0100, Roy Sigurd Karlsbakk wrote:
> > If your DSL link is the bottleneck, rather than earlier hops back
> > through the providers network, the provider could also prioritize VOIP
> > packets going up the DSL line. That requires a cooperating provider,
> > of cour
On Thu, Oct 23, 2003 at 07:55:09PM +0200, Olle E. Johansson wrote:
> Louis-David Mitterrand wrote:
>
> >On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
> >
> >>Also trunking requires that some sort of timing device (digium card or
> >>ztdummy)
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
>
> Also trunking requires that some sort of timing device (digium card or
> ztdummy) be in place for trunking. Otherwise trunking is disabled.
What does ztdummy require to work? kernel compile options? Does it work
on SMP systems?
> Typ
On Mon, Oct 20, 2003 at 08:49:42AM -0700, John Todd wrote:
> At 2:54 PM +0200 10/20/03, Louis-David Mitterrand wrote:
> >> Missing a microphone to work handsfree (or i didn't find it.) but
> >> strange enough their is a speaker ...
> >
> >Yeah, that'
On Mon, Oct 20, 2003 at 08:29:49AM -0700, Anthony Minessale wrote:
> I run into that # issue sometimes too
>
> All I can do is hit ## so the lady tells me there is no ext really
> fast and i may not miss any of the call the # still makes it to the
> real call too.
>
> If you knew in advance yo
On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote:
> >This morning I found myself stumped when a remote interactive system
> >asked me to enter some identification followed by the # key, and my
> >local Asterisk interrupted with "Transfer?".
> >
> >Is there a way to escape the pound key, shor
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with "Transfer?".
Is there a way to escape the pound key, short of disabling transfers?
Cheers,
--
"Make it idiot proof, and s
On Mon, Oct 20, 2003 at 09:21:45AM +0200, Michael Devenijn wrote:
> Justy to let you all know
>
> that i tested 7905G phone with a SIP image (latest download) and it
> works great ! for a reasonable price but with a good quality and a
> brand ... which inspires trust and helps selling better
>
Having purchased a license for 5 g729 channels on Digium's web shop I
thought registration and installation would be a snap. NOT.
I followed registration instructions to the letter but it failed with
that message:
ERROR! Your Internet connection is probably behind a proxy and the
Hi,
To push voice through a long thin wan (dsl) there are two choices:
(1) have the cisco's (7912G) talk g729a to each other (reinvite=yes), or
(2) have the cisco's talk to their local * in ulaw (reinvite=no), which
talk to each other through a more advanced low-bandwidth codec (ilbc or
speex)
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote:
> My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of
> service. They were deployed for about 6 months. These include the AC power
> adapter and station license. We also have some other related equipment. If
> someone
On Mon, Sep 22, 2003 at 03:41:43PM -0400, Jeremy McNamara wrote:
> cvs update the zaptel source and make clean install it.
>
That did it, thanks a lot.
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http://lists.digium.com/mailman/listinfo/asterisk-
Suddenly after recompiling my 2.4.22 kernel I can no longer load
chan_zap:
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span
status: Inappropriate ioctl for device
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to
register chan
On Tue, Sep 16, 2003 at 11:10:33AM +0200, Jean-Marc V. Liotier wrote:
> On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote:
> >
> > i dont think that the Eicon Diva Server 4BRI's NT mode feature will
> > work with linux/capi. I think the feature in the driver is for their
> > PRI cards (wher
On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote:
> On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote:
> > Anyone have a good source for BT-101 phones?
>
> Yes.
>
> But it may not work for you because I've no idea on which of the 5
> continents you are.
I am looking for Grandstrea
On Mon, Sep 15, 2003 at 10:48:00PM -0400, [EMAIL PROTECTED] wrote:
> I upgraded my 7960 firmware to ver 4.4. I now can't make any calls and
> I get errors "(retrans_packet) on call" on the console maximum retries
> exceeded. And ideas?
Check that the bindaddr in sip.conf is set to a reachable ad
If like me you run * on a VPN (or multihomed) gateway and want to serve
remote SIP clients, make sure you have
bindaddr = 192.168.0.1 ; or whatever is your box's private IP
otherwise * might bind to its public IP and send it as return address in
the SIP call setup, which will (should) be rejecte
On Tue, Sep 09, 2003 at 07:57:20PM -0400, Jeremy McNamara wrote:
> Travis Johnson wrote:
>
> >I've called NuFone and was not impressed by their voicemail answering
> >system (choppy) and was unable to even leave a message before the phone
> >call was disconnected (in the middle of the
> >recordi
On Tue, Sep 09, 2003 at 08:23:49AM +, WipeOut . wrote:
> OK you are correct..
>
> *8 picks up the call..I wonder why *8# does not work??
>
> I also had the same problem that the phone that I collected the call
> from did not stop ringing..
I have the same problem. Mark Spencer is working on
On Thu, Sep 04, 2003 at 10:56:10PM -0700, Andrew Gillham wrote:
>
> Unless you're hoping to load Linux or some pirate image in the future,
> there is no
> reason to stay with the old code.
> At least I have not experienced any new issues I can attribute to the
> update to 5.3 code.
Hello,
I bo
Hello,
As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco
7960 with *8 but the extension then keeps ringing indefinitely, even
though I picked up the call.
Is this a known issue? Thanks,
--
There are no Debian developers in any part of Hell, because the good
karma incurre
Hi, I am testing a 7960 in this context:
[SIP] --- > VPN ---> [*] ---> [ANY]
(ANY == any type of phone: isdn, SIP, IAX, etc.)
the call goes through and is dropped after 5 seconds with this message
in the log:
"File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on
call for se
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip entries have:
callgroup=1
pickupgroup=1
However I am unable to remotely pickup a ringing phone using *8#. I get
fast busy tone. Is there some flag to add in extensions.conf ?
Thanks in advance,
_
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