How are you trying to do it? ChanSpy or ZapBarge?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Azzopardi
Sent: Wednesday, May 10, 2006 9:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Stefan Agethen schrieb:
I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones.
I want to post the Solution for this, so others can find it and help
themself.
[...]
I have updated my Kernel and the first day it seems like my problem was
solved, but its not.
Problem : DTMF Tones
Hello,
Is there anyone using sip jitter buffer with callingcard application? it
seems like there is a memory leak that dosent let the app_prepaid_call to
insert acc_start informations in database and send the call so the
asterisk segfaults.
Best Regard,
Hekuran
Are you looking for an web interface that write to asterisk config files? if yes, you can look at freepbx.org . On 5/11/06,
Kerry Garrison [EMAIL PROTECTED] wrote:
You could install any number of interfaces but it does not
come with one.
Kerry
GarrisonDirector of Technical
Hello everyone. I've got this really annoying HFC Cologne card (or
however I should call it - a single channel ISDN card based on the HFC
chipset).
It wrongfully detects lots and lots and lots of incoming DTMFs, to the
point the card is not usable.
Here's a sample out of CLI:
P[ 1] I IND
Ok.. Here we are again - with more input ;-)
Probably I have sorted out the source of the problem. I think it is OVERLAPDIAL.
For example: OVERLAPDIAL is set to YES - The DTMF recognition at mobiles on
outgoing calls is not available
OVERLAPDIAL is set to NO - Every DTMF tone is deteced VERY
Hi! I found, that there is 4 options for nat:
-no
-never
-yes
-always
no and never is ok
but sometimes yes, and sometimes always worked for me :-o
I am having problem diagnosing a call problem. On both a Cisco phone and a
Linksys 942 I am only getting one side of the call when connected over
Rodney G. McDuff wrote:
Is the TE411P just a TE410P with hardware echo cancellation?
Yes. Same for a TE405P and TE406P.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Would any one advice how implement Diva Server BRI or
PRI card to support fax and data modem? In Eicons website, it says that they
support these. But there is no FXS port on the card, how it can be connected to
Fax machine or data Modem?
Thanks in advance.
Isaac Xiao
Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1
install and did an svn update of asterisk-addons and followed the readme
in asterisk-ooh323c and I get through the .configure with no errors. But
make causes:
rpath /usr/local/lib -L./ooh323c/src
Daren J. Howell DTCommunication wrote:
I have restricted the asterisk server to G711 to match the choice on the
PBX, and still same result.
I have read that either endpoint have to be either a master or slave to
communicate to each other. I see in the logs that both are shown to be a
Does digium provide a snmp solution to monitor their
telephony cards ?
Harry
--- [EMAIL PROTECTED] a écrit :
Hello,
I 've installed both cacti and res_snmp for
monitoring.
Does res_snmp is able to send snmp traps when
hardware
is out of service or others status ?
Harry
[EMAIL PROTECTED] wrote:
Does digium provide a snmp solution to monitor their
telephony cards ?
Not at this time, no.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi list,
I want to setup a stun server. I tried stun server from vovida and mystun, but
my voip phone said that logon failed. When I use stun.xten.com works fine.
This is vovida stun in verbose mode:
***received on A1:P1
Got a request (len=28) from xxx.186.145.226:5060
Received stun
On 11/05/2006, at 6:03 PM, Isaac Xiao wrote:
Would any one advice how implement Diva Server BRI or PRI card to
support fax and data modem? In Eicon’s website, it says that they
support these. But there is no FXS port on the card, how it can be
connected to Fax machine or data Modem?
It
I had a very similar issue just today with some Linksys SPA-941's...
Of a collection 15, 5 of them had consistent 'one-sided-audio' on
INBOUND calls, but worked fine on OUTBOUND calls.
In the end, a flash upgrade to 4.1.12(a), a factory reset, and a
reconfig fixed the problem... Same settings
Is it a solution to add some code in those cards in
order to a snmp agent could get/query some
informations about the state of the cards ?
Do you know cards with snmp support ?
Harry
--- Kevin P. Fleming [EMAIL PROTECTED] a écrit
:
[EMAIL PROTECTED] wrote:
Does digium provide a snmp solution
There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines. This might be causing confusion?
Here is my config for a BT ISDN2e line here in UK. I don't think I have
the problems you report.
;TE mode - for ISDN line
nocid=Unavailable
Hi,
I know that this is a more strictly FOP related question than Asterisk but
I'd like to know if regexp buttons support a '-' char, i.e.:
[_Zap/1-.*]
...
In fact, I have:
Zap/1 to Zap/10 as incoming channels
Zap/11 to Zap/15, Zap/17 to Zap/21 as outgoing channels
(it is an E1 PRI)
and I'd
Hi all,I've the current scenario:User A - Zaptel call incoming in my Asterisk to my SIP user B.B gets the Call.A says : B i would like to call PSTN user C
B places a call to user C and asks if C wants the call from A.C says yes i want, then B needs to bridge the between A and C.
The only way i've
On Wed, 2006-05-10 at 21:54 -0400, Dave Morrow wrote:
Hi all. I posted this earlier but never got any advice that helped.
If anyone knows how to get this going, I'd appreciate some advice.
I am attempting to setup Asterisk to allow me to press *1 while in a
call to use automon to record
On Wed, 2006-05-10 at 22:53 -0500, Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 install and did an
svn update of asterisk-addons and followed the readme in
asterisk-ooh323c and I get through the .configure with no errors. But
make causes:
rpath /usr/local/lib
Isaac Xiao schrieb:
Would any one advice how implement Diva Server BRI or PRI card to
support fax and data modem?
You can use any CAIP-fax software to send and recieve Faxes with an
Diva-server card.
The chan-capi driever has such functionality, using capisuite would be
another option.
Hi all.
Iam with the following problem:
I have one perfectly queue functioning, however when the agent receives a call and effects a transference the blind people, blindxfer, asterisk functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended
Hi!
I've one card over from my last asterisk project.
The card is about 3 months old, a copy from the invoice for warranty is
available.
Location: Vienna, Austria.
If anyone is interested - send me a private mail.
cheers,
Tom
(i hope this mail is okay for this list)
I rolled back to 208 and it went right through, I'm headed to mantis to see if I can get that fixed.On 5/11/06, Patrick
[EMAIL PROTECTED] wrote:On Wed, 2006-05-10 at 22:53 -0500, Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 install and did an svn update of asterisk-addons
Hello,
anyone has tested the TigerNetwork IPH202A or IPH202B Ip Phone ? I'm
very interested in known if the quality of this phones is OK, and if are
any problem with asterisk with this Phone.
Sorry for my very bad english !!
Thank you in advance !
Juan Carlos Valero.
Ok... but won't that timeout throw them out of the queue? I want them
to stay in the queue, but I don't want them to get the menu until
after they have been waiting 5 minutes.
On 5/11/06, Anthony Rodgers [EMAIL PROTECTED] wrote:
Done with timeout=600 and queue-thankyou=path/to/sound/file in
Kerry,
You have a good point... about little companies who want termination..
still CEO may be willing to give them good termination... doesn't hurt
to call.
I'm a little confused though... where did I recommend voipjet?
On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote:
You always recommend
Hey, thanks for your reply.. ;)
I'm also using asttapi from website you posted - omniis.com.
Version is 0.10 (newest)
Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't
even implemented in AstTAPI driver so that could be the reason why
Outlook+AstTapi doesen't know what
I have an issue with call parking and hope there is some undocumented feature
for this. ;-)
We are replacing our legacy PBX with asterisk, but to save money over time
(handsets and network), I am trying to maintain the use
of our legacy PBX.
Asterisk extensions can not use the call parking
Jerry Jones wrote:
Has anyone successfully implemented SIPTAPI with asterisk? It would
i use it with Asterisk without problems.
appear to require a true proxy. I assume it will need a seperate user
you need either a separate user or use the user of the SIP phone.
account to register and
Never try upgrades half-asleep and 1/4-knowledgable!
Got a link from a friend about the FLITE TTS that was rewritten to work
really well with Asterisk. So I downloaded and installed it on my 1.0.9
server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install
process, got all
Try removing /usr/lib/asterisk/modules/* that would help. check if you
have extra modules in /usr/lib/asterisk/modules and backup them. after
that do a make install in asterisk-1.2.7.1 and that`s all.
Never try upgrades half-asleep and 1/4-knowledgable!
Got a link from a friend about
It could be an old module still left behind from the previous version. I
would delete everything in /usr/lib/asterisk/modules and then reinstall
(make install) and see if it will start.
On Thu, 2006-05-11 at 14:30, Shawn Porter wrote:
Never try upgrades half-asleep and 1/4-knowledgable!
possible need for onsite tech for Wilmington NC and Boston MA
if you reply pls make sure asterisk does not appear in subject line so it
does not get filtered
to list folder
thanks in advance
_
Dont just search. Find. Check out
Is there a way, when forwarding a voicemail to another extension, to
access to the directory
by name, or, is it better advised simply to have some extension-by-name
labels for my users
to forward their messages to?
Patrick W. Foster
___
Marco:
This is known as attended transfer and is easily found in
voip-info.org, try looking there before asking to the list. This will
avoid reading the same messages from different people every week. You
can find more about attended transfer in:
I seriously doubt that message has something to do with the stun
server. Have you read what the STUN server does? The message you are
getting is most likely to be because of wrong registration user name
or password in your voip phone.
Any way, if you are still interested in having the stun
I did have some extra modules (mysql_cdr, cepstral tts) but I can
start-over. Based on your suggestion, I went one step further. I have gone
through and deleted (rm -Rf just to make sure :) )
/etc/asterisk
/var/lib/asterisk
/usr/lib/asterisk
/usr/include/asterisk
/usr/sbin/asterisk
I am just
Información Capa Tres S.L. wrote:
Hello,
anyone has tested the TigerNetwork IPH202A or IPH202B Ip Phone ? I'm
very interested in known if the quality of this phones is OK, and if are
any problem with asterisk with this Phone.
I tested IP202 and ATA104, both are working well.
--
Daniel
Moises Silva,I've already tried to activate:features.confatxfer=*2as well as setDYNAMIC_FEATURE=atxfer in my [globals] of extensions.confBut I couldn't get it working, that's why I asked it in the mailing list.
Thanks for your help.Best regards,Marco MoutaOn 5/11/06, Moises Silva [EMAIL PROTECTED]
On Thursday 11 May 2006 16:51, Moises Silva wrote:
I seriously doubt that message has something to do with the stun
server. Have you read what the STUN server does? The message you are
I known what do a stun server.
getting is most likely to be because of wrong registration user name
or
You need the span= BEFORE any channel lines. You also need any options
before the channel lines.
Ex:
span=1,1,0,ccs,hdb3,crc4
loadzone=sg
defaultzone=sg
bchan=1-15
bchan=17-31
dchan=16
azyuky wrote:
I know and it's crazy isn't it because this is what i have in my zaptel.conf
--
Now
The problem is called talkoff. Search the mailing list archives.
Also post your sip.conf and zaptel.conf (sans passwords, of course)
Stefan Agethen wrote:
Stefan Agethen schrieb:
I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones.
I want to post the Solution for this, so
Sharon Lim wrote:
Are you looking for an web interface that write to asterisk config
files? if
yes, you can look at freepbx.org .
Hello,
just out of curiosity -- are you based in Malaysia?
Cheers,
Flynn
___
--Bandwidth and Colocation provided
On 05/11/06 19:46 Josué Conti said the following:
functions informs to transfer and the transference is ok. However, if
the agent tries to effect an attended transference the ATXFER, knocks
down the call. All the agents of this queue are with canreinvite=no in
i'm guessing that the feature
Tzafrir Cohen wrote:
On Sun, May 07, 2006 at 12:21:53AM -0400, Roger Gulbranson wrote:
On Sat, 2006-05-06 at 19:43 -0400, Steve Totaro wrote:
Roger Gulbranson wrote:
On Sat, 2006-05-06 at 07:42 -0400, Steve Totaro wrote:
I have a TDM4xxp card with no modules. My
On Thu, 2006-05-11 at 11:13 -0400, Steve Totaro wrote:
I was able to install the card and get wctdm to load with the timingonly
= 1 setting.
I cannot run ./zttest though, I get Unable to open zap interface: No
such device or address Does this mean that I am not getting timing?
Is this a
Is there a public download site for Linksys/Sipura firmwares? I found
nothing on Linksys site. I'm currently running 4.1.10(e) on my SPA-942.On 5/11/06, Adrian Carter [EMAIL PROTECTED]
wrote:
I had a very similar issue just today with some Linksys SPA-941's...
Of a collection 15, 5 of them
I'm using DUNDi.
My lookup returns 'IAX2' for the tech, and 'dundi:[EMAIL PROTECTED]/3254101'
for the destination.
How do I dial this?
I've tried dialling it with:
Dial IAX2/dundi:[EMAIL PROTECTED]/3254101
passed from my AGI script, but the other endpoint (xxx.187.142.204) is
returning:
I was thinking in sending you an attachment, but I have decided to put
it on the web, you can get it in
http://phpmexic.u33.0web-hosting.com/wordpress/ebuilds/vovida-stun-0.96-ebuild.tar.bz2
Let me know if you it have bugs
Regards
On 5/11/06, Serghei Amelian [EMAIL PROTECTED] wrote:
On
Well Marco, thats different ;)
You could start your last post saying that and may be you would have
had more responses.
Please let us know the asterisk version you have, the complete
features.conf file and what does the console says in verbose mode?
On 5/11/06, Marco Mouta [EMAIL PROTECTED]
Did you set up a dundi iax user in iax.conf?
On Thu, 11 May 2006, Douglas Garstang wrote:
I'm using DUNDi.
My lookup returns 'IAX2' for the tech, and 'dundi:[EMAIL PROTECTED]/3254101'
for the destination.
How do I dial this?
I've tried dialling it with:
Dial IAX2/dundi:[EMAIL
does any one know
if Grandstream BT 100 has a 'do not disturb' function ?
one of the BT's is not picking up calls all of a sudden
out going is fine
but when you dial in
you get forwarded directly to voice mail.
thanks
___
--Bandwidth and Colocation
No... do you have an example of what that looks like? I get more matches on
google for 'the early history of hungarian cabinet making' than I do for DUNDi
examples.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 11, 2006 9:43 AM
To: Asterisk
Roger Gulbranson wrote:
On Thu, 2006-05-11 at 11:13 -0400, Steve Totaro wrote:
I was able to install the card and get wctdm to load with the timingonly
= 1 setting.
I cannot run ./zttest though, I get Unable to open zap interface: No
such device or address Does this mean that I am not
Douglas Garstang wrote:
No... do you have an example of what that looks like? I get more
matches on google for 'the early history of hungarian cabinet making'
than I do for DUNDi examples.
[dundi]
type=user
dbsecret=dundi/secret
context=dundi-e164-local
Best regards,
Florian
Take a look here:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+IAX
Just add a user named dundi with a dbsecret of dundi/secret and it should
work.
Aaron
On Thu, 11 May 2006, Douglas Garstang wrote:
No... do you have an example of what that looks like? I get more
Thanks. I got it working.
Now I've hit a HUGE snag.
We're doing all of our call routing from a database accessed from AGI. When we
trunk calls from one asterisk system over to another via IAX to terminate the
call, the dialling parameters are defined by what's in the dial command on the
second
Hey all, am running into a problem with * 1.2.1 recently. When we leave a
voicemail for someone, occasionally when they check the vm, * doesn't play
back the message that was recorded. I can see the vm audio file on the
server, but when it's checked from a phone, it just skips to the end of the
I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created
from regexten in sip.conf, and I do an 'extensions reload', I lose all the
priority 1 NoOps! This can't be right... this means that in a production
environment, if you make a change to your dialplan and do an
Douglas Garstang wrote:
We're doing all of our call routing from a database accessed from
AGI. When we trunk calls from one asterisk system over to another via
IAX to terminate the call, the dialling parameters are defined by
what's in the dial command on the second system, not the first. This
I am trying to use digium TDM04B type cards combined with call
files like:
Channel: Zap/1/5551212
Context: smvoice-dialout
Extension: 9
Application: Playback
Data: demo-congrats
To call out and play messages. My problem is as soon as the phone is dialed
the message demo-congrats starts
I'm also seeing that re-registrations from the phones are not recreating the
priority 1 NoOP's I have to completely restart Asterisk, and they come
back. I assume they're being repopulated from astdb. Good grief.
-Original Message-
From: Douglas Garstang
Sent: Thursday, May 11,
Douglas Garstang wrote:
We're doing all of our call routing from a database accessed from
AGI. When we trunk calls from one asterisk system over to
another via
IAX to terminate the call, the dialling parameters are defined by
what's in the dial command on the second system, not the
Jerry Geis wrote:
I am trying to use digium TDM04B type cards combined with call
files like:
Channel: Zap/1/5551212
Context: smvoice-dialout
Extension: 9
Application: Playback
Data: demo-congrats
To call out and play messages. My problem is as soon as the phone is
dialed
the message
As a note, if you don't create the dundi_local context in extensions.conf,
SIP will create the context and doing an extensions reload won't get rid
of the registration information. Also, if you notice a phone isn't
showing up in the list, try doing a sip prune realtime exten and then
a sip
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 11, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 'extensions reload' clears Regextens
As a note, if you don't create the dundi_local
Dan Elder wrote:
Hey all, am running into a problem with * 1.2.1 recently. When we leave a
voicemail for someone, occasionally when they check the vm, * doesn't play
back the message that was recorded. I can see the vm audio file on the
server, but when it's checked from a phone, it just skips
Does Asterisk support a Brooktrout TR1000 ?
http://www.cantata.com/products/tr1000/
It seems that they have linux drivers.
I have one around and was wondering if it works with *.
Thanks,
--
---
Erick Perez
Linux User 376588
http://counter.li.org/ (Get
Aaron,
I tried that. If I have no dundi_local in extensions.conf, I get the automatic
priority 1 NoOps (and yes, an 'extensions reload' doesn't clear them), but of
course, because I have no dundi_local in extensions.conf, I also have no
priority 2 Dial commands, so how can the system reach
On Thursday 11 May 2006 18:32, Moises Silva wrote:
I was thinking in sending you an attachment, but I have decided to put
it on the web, you can get it in
http://phpmexic.u33.0web-hosting.com/wordpress/ebuilds/vovida-stun-0.96-ebu
ild.tar.bz2
Let me know if you it have bugs
I'm not lucky,
The answer is to create a separate context for your regextens (or, more
appropriately, name it in sip.conf and let chan_sip create it) and then
include that context in your dundi_local context where you have the
dialing information.
Regards,
- Brad
-Original Message-
From: [EMAIL
I'm trying to connect an Asterisk T1 port to a
Dialogic card. The Dialogic side is an external VMS.
I setup for ISDN-PRI between systems and have
green lights on both card/ports. Zttool shows connection is good also.
However, when I tryattempt terminate or
originate a call to either
Ok, thanks guys. I think I got this simple case working.
Now I have the major issue... when the dial on the second Asterisk system is
called, it's in extensions.conf, and the ring options are hard coded. However,
the agi script on the first box grabbed the ring options from a database... I
Thanks for the fix :)
About DNS, I think only the primary stun address should be registered.
Try using as stun server grievous.ivsol.net, is our stun server and is
installed with the ebuild I sent you. That will give us more hints
about where the problem might be.
Try using the stunclient
http://www.asterisk.org/hardware
Erick Perez wrote:
Does Asterisk support a Brooktrout TR1000 ?
http://www.cantata.com/products/tr1000/
It seems that they have linux drivers.
I have one around and was wondering if it works with *.
Thanks,
--
Now accepting new clients in Birmingham,
On Thursday 11 May 2006 21:10, Moises Silva wrote:
Thanks for the fix :)
no problem
About DNS, I think only the primary stun address should be registered.
Try using as stun server grievous.ivsol.net, is our stun server and is
installed with the ebuild I sent you. That will give us more hints
Hi!
I've went through the READMEs and could not answer this question:
During installation, the Setup program asks:
Would you like update/upgrade wanpipe drivers? (y/n)
For a pure Asterisk TDM installation - is it required to patch the
kernel or is this only when using the sangoma cards as
Klaus,As far as I've been told, yes, you do need to compile the kernel modules (even if it's just TDM functionality).AlexOn 5/11/06, Klaus Darilion
[EMAIL PROTECTED] wrote:
Hi!I've went through the READMEs and could not answer this question:During installation, the Setup program asks: Would you
On mine, I had that happen, until I turned off subscribe to MWI or
something like that in the config (sorry - I can't remember the exact
verbiage.) I also upgraded the firmware around the same time, but I
think from advice I got on this list, the MWI setting was the reason...
-Steve
That list is obviously not complete ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric ManxPower Wieling
Sent: May 11, 2006 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and
Even if I answer n, the Setup script still compiles wanpipe modules.
Thus I guess answering yes is only needed if I want to have kernel
sources synchronized with the installed modules (binaries). If I do not
care about a kernel source tree which includes latest wanpipe drivers I
think
That's why am asking here, i already checked that list.
On 5/11/06, Nabeel Jafferali [EMAIL PROTECTED] wrote:
That list is obviously not complete ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric ManxPower Wieling
Sent: May 11, 2006 2:43 PM
I am looking to setup paging using the auto answer feature
on the Grandstream GXP2000. I am thinking I will follow the method as
described here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I will setup the 4th account on the phone to auto
answer.
Does anyone
The wav file seems fine when I can catch it, I'm able to play it through
winamp without a problem. Something is making * skip to the end of the
message not actually play any audio, then it moves the message to the
'old' folder, and agents don't normally check that folder. Any ideas where I
can
i've checked for that
it is 'subscribe to MWI'
but it is set to 'no' already
i looked in the full log
for error and disabled came up with these
May 11 16:17:39 VERBOSE[26307] logger.c:
-- dialparties.agi: Extension 107 cf is disabled
May 11 16:17:39 VERBOSE[26307] logger.c:
--
Following up with some more tests... I just left a message in a voicemail
box, and got the same behaviour (i.e. it says its starting to play ('First
Message') then it goes to the VM ending menu (i.e. press 7 to delete)
without playing the message. On this particular test.. I went ahead left a
2nd
Dan Elder wrote:
The wav file seems fine when I can catch it, I'm able to play it through
winamp without a problem. Something is making * skip to the end of the
message not actually play any audio, then it moves the message to the
'old' folder, and agents don't normally check that folder. Any
Use a macro that uses the ParkAndAnnounce application and set the
return context there.
On 5/11/06, Steven [EMAIL PROTECTED] wrote:
I have an issue with call parking and hope there is some undocumented feature
for this. ;-)
We are replacing our legacy PBX with asterisk, but to save money over
Steve Totaro wrote:
Roger Gulbranson wrote:
On Thu, 2006-05-11 at 11:13 -0400, Steve Totaro wrote:
I was able to install the card and get wctdm to load with the
timingonly = 1 setting.
I cannot run ./zttest though, I get Unable to open zap interface:
No such device or address Does this
Douglas Garstang wrote:
What am I trying to achieve? Uhm... a carrier grade, highly redundant
(ie multiple servers), VOIP solution with advanced business(not
residential) features such as findme/followme, incoming and outgoing
blacklisting/whitelisting(user/org/company level), user/prefix
On Thu, 2006-05-11 at 10:33 -0600, Douglas Garstang wrote:
[snip]
When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the
ring time and ring options of the original SIP call between servers.
Iirc you can pass variables on the IAX link to the other side. Maybe you
can use
Yes, I have the exact same problem.
:(
-Original Message-
From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 11, 2006 5:48 AM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook
Thanks for the information, I will surely look into it!
Nitin
On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Have you looked at CBeyond? I like their T1 SIPConnect product.
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PM
I am having some
troubles with DTMF detection on zap channels when the remote caller is calling
from a noisy cell phone. It is actually detecting multiple DTMF tones (usually
2 or 3) when only one is sent (i.e. I press 3 and Asterisk is
detecting that as 333.) I dont know the exact
What version of Asterisk?
On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am looking to setup paging using the auto answer feature on the
Grandstream GXP2000. I am thinking I will follow the method as described
here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I've set voicemail almost successfully, only a minor detail remains :-)
I can't get the dates in my local language (spanish). In sip.conf,
zapata.conf and voicemail.conf, I've set:
language=es
and my locale is es also. However, the days and months names still
appear in english in the
if you ar using SIP clients, try changing DTMF transfer mode.
For test use
sip debug
on your * console, then place a call and watch the output. In INFO or
rfc2833 mode you must see the codes like SIP messages. If you are using
inband transfer mode (DTMF codes are transferred like sounds) you
1 - 100 of 129 matches
Mail list logo
