Edward Halman wrote:
Can anyone recommend a good cordless user-configurable SIP hardphone
that is readily available in the states and doesn’t cost $300? There
seem to be a plethora of decent and affordable corded phones (like from
Grandstream) but the search for a cordless unit seems elusive.
I would be interested to know whether this
http://bugs.digium.com/view.php?id=8376
patch makes any difference. The problem is almost certainly not caused
by Centos (which is widely used with Asterisk) or EPIA (which I use
lots).
Regards,
Steve
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
Hello all,
When an incomming SIP call is reveived I would like to force Asterisk to
keep the SDP codec selection for the resulting outgoing call to the
destination SIP endpoint. Does anyone know how this could be acheived? I
know that the allowed codecs for each SIP endpoint can be restricted
Hi all,
is where a possibility for simply parsing and changing variables for bad
characters ?
eg. removing a '/' from a number dialed by a manager-connected application
changing 123/4567890to 1234567890
via bash you could simply use 'echo ${exten/\//}' but i couldn't find a
working
Hi,
If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it
as an complete product to an coustomer, do I have to put my developed code or
the complete product under the GPL?
best regards
Thomas
___
--Bandwidth and Colocation
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is
PC working fine in my Setup, but I want Extern1 to talk to Extern2
PC directly whitout going over Asterisk as the uplink is slow.
PC When I set for Extern1/2 canreinvite=yes it
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
For that I've set up a local DNS Cache (on the asterisk) - maradns. And
entered 127.0.0.1 as the first DNS Server in d/etc/resolv.conf.
To decrease the time asterisk is trying to do a dns lookup, I've added
this options to /etc/resolv.conf:
Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter:
Hi,
If I develope an dialplan, some AGI and AMI functions for Asterisk and ship
it
as an complete product to an coustomer, do I have to put my developed code or
the complete product under the GPL?
IANAL, but in my understanding
On 01/28/07 18:52 Florian Overkamp said the following:
Nokia seems to have done something like this in their E-series (E60 etc)
with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ?
i think that's a FMC (fixed mobile convergence) client which both avaya and
cisco wrote
On Mon, Jan 29, 2007 at 12:30:47PM +0100, Anselm Martin Hoffmeister wrote:
Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter:
Hi,
If I develope an dialplan, some AGI and AMI functions for Asterisk and ship
it
as an complete product to an coustomer, do I have to put my
Jason,
Email me off-list and I will ship you a pack of usable configs.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 26, 2007, at 3:48 PM,
Do you run asterisk through a wrapper as safe_asterisk ? (If not hi
suggest you to do so)
You can unload zaptel module from that script after a crash and reload
it when the script tries to restart asterisk
I'm using this solution on many production server whithout problems
It sounds weird
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten =
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
On Monday 29 January 2007 03:20:16 pm Steve Davies wrote:
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I
would like to be able to
somebody know how to compile the rxfax and txfax apps under asterisk
1.4.0??
i get this errors:
Generating embedded module rules ...
make[1]: Nothing to be done for `all'.
make[1]: Nothing to be done for `all'.
make[1]: Nothing to be done for `all'.
make[1]: Nothing to be done for `all'.
make[1]:
The first include references another context within extensions.conf. Contexts
are defined by words in brackets. In your example, there would be a context in
extensions.conf that would look like:
[inbound]
Contexts allow for setting up difference services and difference user
capabilities all
Hmm. Mantis says that in SVN 51223 it was implemented, im running
51363. However I may be wrong. I will apply that patch and let you
know.
Thanks for the pointer.
should I leave asterisk as -march=i586? or 386?
On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote:
I would be interested to know
Turns out this appears to be related to hald -- the hardware abstraction
layer daemon running on Centos. I had the almost identical situation occur
with a completely separate system which I loaded Trixbox up on, with a
single Digium TDM400P card in it. Struggled for several hours over the
weekend
Benny Amorsen schrieb:
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is
PC working fine in my Setup, but I want Extern1 to talk to Extern2
PC directly whitout going over Asterisk as the uplink is slow.
PC When I set for
Hi everyone
dou you know if is possible to put an Agi script in a queue?
For Example
1 - Caller joins the queue
2 - Agi script starts
...
...
Agi script ends
3 - Hangup.
Is it possible?
thanks
___
--Bandwidth and Colocation provided by Easynews.com
WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is
deprecated.
Please use ${DB(cidname/${CALLERID(num)})} instead.
[WARNING[8384]: app_lookupblacklist.c:104 lookupblacklist_exec: LookupBlacklist
is
deprecated. Please use ${BLACKLIST()} instead.
I seem to be unable to
Leo Ann Boon wrote:
It is, and is identified by wcfxo as a Wildcard FXO: Wildcard
X100P. So much for The DigitNetworks X100P is detected as an actual
X101P card.
IIRC, there were 2 Digium single FXO cards - the X100P using the
Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps
We are using a couple of Grandstream GXP2000 SIP-phones with Asterisk.
In our dial-plan, we have implemented a list of short numbers in
extensions.conf, like:
exten = 1234,1,Dial(Zap/0987654321)
So when I pickup the SIP-phone, and I dial 1234, the system dials
0987654321 and connects me to
I failed to notice that it was included in 51363 - I just checked, and
that change is indeed already in. Sorry, my mistake.
I generally do not change the -march setting, so I am probably using
an i386 default.
Regards,
Steve
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
Hmm. Mantis says
Zoilo Gomez wrote:
So when I pickup the SIP-phone, and I dial 1234, the system dials
0987654321 and connects me to that customer. Unfortunately I cannot see
the name of the customer, and I do not know if perhaps I punched the
wrong short number.
Is there a way to have Asterisk print the name
On Mon, 29 Jan 2007, Steve Davies wrote:
I failed to notice that it was included in 51363 - I just checked, and
that change is indeed already in. Sorry, my mistake.
I generally do not change the -march setting, so I am probably using
an i386 default.
I get segfaults with the VIA C3 and C7
I upgraded to the newest 1.2 Zaptel release and this is still occurring. I
checked and the digium card is not sharing an IRQ with any other devices.
I also changed busycount=8, and set callprogress=no.
The call drops are still occurring. Mid-conversation ` in 10 calls will be
disconnected.
Other phones have a defined dialplan, just like an ATA the GXP is the only
phone I've seen like that!
I had a sudden stroke of genius, I haven't tested it, but I'm sure it would
work. Define a DISA with no password at extension 011, and define a context
where international calls can be dialed
Erick Perez wrote:
both not available.
but thanks.
Email edwin ( [EMAIL PROTECTED] ), he will be able to help.
There is a newer firmware available than the one on their website
(v4.2b5) which fixes problems with freezing and introduces a phone book,
a digitmap and simple dialplan.
There
Before adding a second interface to my asterisk box I'd like to get some
feedback on having and internal interface with a private address and
external interface with a public interface. You know like pros, cons,
configuration suggestions, and anyone's true experience trying such a
design. I have
If it's using RBS then 56k is the right number.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Saturday, January 27, 2007 12:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] max tnt pri voice
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected. However, when I try to
something like (AEL syntax):
if (${DB_EXISTS(cidname/${CALLERID(num)})})
CALLERID(name)=${DB(cidname/${CALLERID(num)});
Derek Whitten wrote:
WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is
deprecated.
Please use ${DB(cidname/${CALLERID(num)})} instead.
Hello,
Do you include in your zapata.conf
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
There are any problems with hang up
Regards
On 1/29/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I
If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active
On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote:
Do you run asterisk through a wrapper as safe_asterisk ? (If not hi
suggest you to do so)
You can unload zaptel module from that script after a crash and reload
it
I have been down this path with Grandstream but they (for reasons I don't
understand) want to upgrade the firmware to have a dial plan.
So the best you can do is use early dial, for all fixed length numbers in
the * dial plan this works reasonably well. International numbers vary in
length so
Tell that to ATT who socked us with multiple $20k bills. We cant
figure out where the error was. Or why a call was established for
over 50 hours between two states with completely different PBX
hardware.
On 1/29/07, C F [EMAIL PROTECTED] wrote:
If Asterisk Is Down Then The D Channel Is Down
Wow, thanks for the awesome reply :)
On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Shane Spencer wrote:
I am trying to do a wire level tap on T1 equipment using digum
equipment. So far most call monitoring hardware for call centers try
to stay on the analog side requiring a lot of
I am very interested in the DACs capabilities of Digium cards, there
is no information anywhere on this. I could always do pri bridging
via libpri like you suggest however. But having hardware handle the
bridging onboard a single PCI card would help reduce my server
requirements for a final
Try setting AbsoluteTimeout() as the first parameter in your dialplan
entry. Check it out on voip-info.org
On 1/28/07, kjcsb [EMAIL PROTECTED] wrote:
Anyway, my question is, how do I get the offhook status to reset? So far
only a server reboot works. I tried:
- physically disconnecting the
Hello,
Check app_backticks - it is an external application which should be
compiled on your system.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
[EMAIL PROTECTED] wrote:
Hi all,
is where a
Carlos Rojas wrote:
Hello,
Do you include in your zapata.conf
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
There are any problems with hang up
I tried adding these parameters as you suggested, but then was unable to
dial out at all. Removing them allows me to dial out again, but
On 27 Jan 2007, at 16:33, Lee Jenkins wrote:
Although I wouldn't complain about a free G.729 codec, I have to be
honest in saying that $10.00 isn't that great of an expense
considering the better call quality you get.
Does G.729 work by pushing up the compression, therefore moving from
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected.
Lee Jenkins wrote:
Lee Jenkins wrote:
After playing around a bit, it appears that this is just random as far
as I can see. It may allow me to dial a few times, but then hangup.
After rebooting my server, it may let me dial once and then start
hanging up.
I really hope it's not a fight
Lee Jenkins wrote:
Lee Jenkins wrote:
Lee Jenkins wrote:
After playing around a bit, it appears that this is just random as far
as I can see. It may allow me to dial a few times, but then hangup.
After rebooting my server, it may let me dial once and then start
hanging up.
I really hope
Hi, Lee:
Lee Jenkins wrote:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on the caller first.
Then it
Carlos Rojas wrote:
Hello,
Do you include in your zapata.conf
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
This doesn't work everywhere. I don't think Verizon does disconnect
signalling with a polarity switch, though I'd be happy to be corrected.
What part of the world are you
Eric ManxPower Wieling wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I
Stephen Bosch wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on the caller
Hi, Lee:
Lee Jenkins wrote:
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought (and posted incorrectly
Snom's do 729 just fine 190,320,360 all have worked well for us
On 27 Jan 2007, at 16:33, Lee Jenkins wrote:
Although I wouldn't complain about a free G.729 codec, I have to be
honest in saying that $10.00 isn't that great of an expense
considering the better call quality you get.
Does
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on the caller first.
Then it will
Lee Jenkins wrote:
Well, not really -- when the extension hangs up, Asterisk knows the
channel has been abandoned and clears it.
When the remote party hangs up first, the card doesn't tell Asterisk
that the channel is clear (because it doesn't know the caller has hung
up), so Asterisk has no
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought (and
Hello everyone! In my humble try of creating a Blog, I've made this:
http://fameal.blogdns.org.
By now, it's hosted in my own server but shortly it'll be moved to a
serious hosting. All post are written in spanish, so it's only for
spanish talking people, I will try to make it grow and have
Eric ManxPower Wieling wrote:
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension
Thanks for the response, I 've already matched codecs. I have no
problems with that. Do rxgain and txgain have something to do with R2
protocol errors?
Regards.
On 1/28/07, Angel Heart [EMAIL PROTECTED] wrote:
Hi Facundo,
Were you able to match your phone's codec with the asterisk codec? Try
Stephen Bosch wrote:
Lee Jenkins wrote:
Well, not really -- when the extension hangs up, Asterisk knows the
channel has been abandoned and clears it.
When the remote party hangs up first, the card doesn't tell Asterisk
that the channel is clear (because it doesn't know the caller has hung
On Friday 26 January 2007 23:40, Leo Ann Boon wrote:
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
SNIP
busydetect=yes
You may need to add these 2 values to help the busydetect
busycount=3
busypattern=375,375
busypattern tells asterisk how your
-Original Message-
John Marvin,
Thank you very much. The CYT35 utility worked like a charm, though I feel a
bit like a criminal. Not at all intuitive to set up, but the VTech 8100-2
is performing marvelously with my asterisk setup. I just got my grandstream
budgetones in the mail,
I was able to update the password through the dialplan with this:
exten = ,1,MYSQL(Connect connid 127.0.0.1 pbx pbx pbxdb)
exten = ,2,MYSQL(Query resultid ${connid} UPDATE\ voicemail\ SET\
password=\ where\ mailbox=52007)
exten = ,3,MYSQL(Clear ${resultid})
exten =
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected.
Kyle Gordon wrote:
snip
Hi Leo,
That appears to have done the trick. fxs_ls does seem to detect it hanging up
more reliably. I don't know what the difference is, but it works :-)
If there's any change, I'll be sure to let you know :-p
No problemo. Glad to know it worked for you. Like
Hi, Lee:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is
OK, I think I may have found the problem for myself at least. Actually,
a friend of mine suggested it. Apparently, Asterisk is a little too
fast for the card.
Placing a w in front of the number to insert a pause looks like it did
the trick!
Dial(ZAP/1/w555)
Looks like it gives the
Lee Jenkins wrote:
OK, I think I may have found the problem for myself at least. Actually,
a friend of mine suggested it. Apparently, Asterisk is a little too
fast for the card.
Placing a w in front of the number to insert a pause looks like it did
the trick!
Dial(ZAP/1/w555)
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion,
Yuan LIU wrote:
From: /Lee Jenkins [EMAIL PROTECTED]/
[...]
If I call out to a party on that Zap line and hangup first, I do
not experience that problem. It looks like Asterisk is not getting
the termination signal from the telco (Verizon) when the other
party hangs up first.
Running
Hi,
I'm not sure, but I experienced it before with our Nortel Meridian I MFC/R2.
When set to both zero(0), calls drop once answered. I tried to vary its values
until I finally got it stabled. I'd been in the Datacoms/Telecoms for 16 years
now, only with Asterisk I experienced beyond technical
Hi:
Is there any way to adjust the detection threshold for kewlstart
signalling on the TDM-400 cards?
Example:
The telco provides a 100 ms open loop or battery drop to indicate
remote party hangup. If the zaptel driver expects to see a 350 ms drop,
it will never detect the hangup and sit on the
I'll try it during weekend then.
Thanks for the help. I appreciate it.
On 1/29/07, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
I'm not sure, but I experienced it before with our Nortel Meridian I
MFC/R2. When set to both zero(0), calls drop once answered. I tried to vary
its values until I
Shane Spencer wrote:
I am very interested in the DACs capabilities of Digium cards, there
is no information anywhere on this. I could always do pri bridging
via libpri like you suggest however. But having hardware handle the
bridging onboard a single PCI card would help reduce my server
I wanted to know if there was a peekaboo factor to it all. You can
flow data under a glass window :)
On 1/29/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Shane Spencer wrote:
I am very interested in the DACs capabilities of Digium cards, there
is no information anywhere on this. I could always
Ed W wrote:
Using a TDM400P in the UK nearly works fine, but I have a last remaining
problem in that if the incoming is ringing and then the caller hangs up,
asterisk takes another couple of rings before it spots the hangup.
This is annoying in that I sometimes get phantom calls late at
Shane Spencer wrote:
I wanted to know if there was a peekaboo factor to it all. You can
flow data under a glass window :)
Well - you can always use a logic probe :). Bridging does add a little
latency to the whole thing. Why don't you consider a passive tap
solution like the hi-z OpenPRI
you got that while doing SIP/ZAP and parking?
On 1/29/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Mon, 29 Jan 2007, Steve Davies wrote:
I failed to notice that it was included in 51363 - I just checked, and
that change is indeed already in. Sorry, my mistake.
I generally do not change
Hello,
Anyone managed to control a Cisco voice gateway (2,811 in my case) using
MGCP? I cannot make the PRI going on-line (while with SIP I can).
If you ask why I want to use MGCP and not SIP: it is because Cisco uses
different Q.sig signalling when you manage it with different protocols,
n asterisk 1.2 branch SVN 51363
zaptel svn 1980
libpri svn 393
addons svn 332
Asterisk is connected via tdm400p to an avaya system to reach PSTN.
When a pstn phone hangs-up asterisk seems unable to detect the busy
tone and i keep hearing like 20 busy tones until the zap channel get
closed. I'm
From: Lee Jenkins [EMAIL PROTECTED]
Hi Yuan,
Thanks for chiming in. I accidentally posted the fix to a wrong thread
above. Been a very long day ;) Here is what I posted:
OK, I think I may have found the problem for myself at least. Actually, a
friend of mine suggested it. Apparently,
What avaya system is this, if the avaya is configured on the ports to
use a 2500 set, then it should do CPC and should work as is.
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
n asterisk 1.2 branch SVN 51363
zaptel svn 1980
libpri svn 393
addons svn 332
Asterisk is connected via tdm400p to
Shane, are you trying to say that the PRI was actualy down (the D
channel was NOT up) for the time that ATT is billing you?
On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote:
Tell that to ATT who socked us with multiple $20k bills. We cant
figure out where the error was. Or why a call was
When Asterisk dials an IAX destination with no registration, it very quickly
comes to the conclusion that it can't make the call
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1,
IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
[Jan 29
This is a G3. And I'm not the avaya operator. What do you mean with
2500 set and CPC?
On 1/29/07, C F [EMAIL PROTECTED] wrote:
What avaya system is this, if the avaya is configured on the ports to
use a 2500 set, then it should do CPC and should work as is.
On 1/29/07, Erick Perez [EMAIL
It was either down or asterisk was frozen. Either way a heartbeat
could fix that.
On 1/29/07, C F [EMAIL PROTECTED] wrote:
Shane, are you trying to say that the PRI was actualy down (the D
channel was NOT up) for the time that ATT is billing you?
On 1/29/07, Shane Spencer [EMAIL PROTECTED]
Using tdm400. While transfering a call from outside to another
extensions, while this outside call is waiting with music, the
another extension call hangs up suddenly, and the call is back to the
outside call suddenly.
Wathcing logs:
Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes
91 matches
Mail list logo
