when i use the pound key . there is no pause .so that means that the pap2 box is waiting for aditional key is it ? how do we fix this ?thanksGiridhar BandiOn 3/10/06,
indsat [EMAIL PROTECTED] wrote:
Here's a site that will help you with PAP2 Dial
Hi,
Maybe this isn't the right way...but this is the first thing that popped
into my head;
Use two contextes. For example, context_A and context_B. For all group A
extensions, make context_A their default context and group B extensions to
context_B. Then, in each context, define only
Does anyone here use either Gradewell or inWeb for service? They are both UK
based. I'm trying to get a couple of
inbound IAX2 based numbers from both of them to work with no luck at all. The
one thing that sets these guys apart from
the rest of companies offering inbound numbers is they tie
Tzafrir Cohen wrote:
Still: no jump to line,
Ctrl + _
--
Best regards,
Bartosz Piec
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Robert P. McKenzie wrote:
Does anyone here use either Gradewell or inWeb for service? They are both UK
based. I'm trying to get a couple of
inbound IAX2 based numbers from both of them to work with no luck at all. The
one thing that sets these guys apart from
the rest of companies offering
Has anybody a firmware for updating a mediatrix 1102 (sip)?
--
Alejandro Vargas
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Thanks for the answer!
Kevin P. Fleming wrote:
Not at this time, no. I am not aware of any Asterisk-compatible cards
that operate in PCI Express slots yet.
I see. What confuses me is the following offer:
http://voipcomponents.co.uk/product_info.php/products_id/176
If I understand
Hi all (is a bit of a Cisco question, but perhaps it's
already been asked here).
I'm trying to setup a TFTP server for dynamic
configuration of my 7960 Cisco phones.
Cisco docs suggest I download the following files from
their web site:
* OS79XX.TXT
* SIPDefaultGeneric.cnf
*
how i can made the connection from my database name student to asterisk system because , when i use database name cdr so conection can be made and the system run but when change back tu student database
system asteris cannot find my database
___
2006/3/7, Giridhar Bandi [EMAIL PROTECTED]:
and when i place a call to local sip extension there is a long pause ( 15
sec )
before the call gets dialled
Use the # key as enter.
--
Alejandro Vargas
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Hi All
I am working on asterisk + digium developer card
, it has on FXO and one FXS
I want to work asterisk in the following way
1FXO connected to PSTN line
2the calls coming to PSTN line should be received
3SPI clients should be able to call outside through
PSTN
4There is no phone
my server will be in one country . and one group will be on another
country. . so pppoe will not work in here i think
thanks
Salaque
On 3/10/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Hi,
Maybe this isn't the right way...but this is the first thing that popped
into my head;
Use two
and one more information . user will dial to pstn number as well as
local extensions
thanks
Salaque
On 3/10/06, Mohammad Salaque [EMAIL PROTECTED] wrote:
my server will be in one country . and one group will be on another
country. . so pppoe will not work in here i think
thanks
Salaque
Try here... http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960
But you need an account on cisco to download this!
-Oorspronkelijk bericht-
Van: Mark Tinka [mailto:[EMAIL PROTECTED]
Verzonden: vrijdag 10 maart 2006 9:49
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Johnathan Corgan wrote:
When the user sets the forwarding number, store the user's context in
the DB along with the forwarding number. Make sure you have an invalid
extension 'i' in the user's context as well.
It helped me a lot but the 'i' extension is called when the call is
forwarded. Is
On Friday 10 March 2006 11:16, P.A. Oudakker wrote:
Try here...
http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone
7960 But you need an account on cisco to download this!
Go figure :), same place I downloaded the 7.5 firmware,
but didn't notice the other files way below.
Cheers,
Mark.
hi
is there a web tool for monitor and get statistic about the cdr
records stored in a postgres database?
thanks
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On 10:59, Fri 10 Mar 06, nik600 wrote:
hi
is there a web tool for monitor and get statistic about the cdr
records stored in a postgres database?
http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
Have fun
--
Michiel van Baak
[EMAIL PROTECTED]
Dear folks,
I have a problem with console/dsp using ALSA. I dont know why the output
sound is choppy sometimes and also the input one has an awful delay. Is
there anyone here with experince about ALSA channels or not?
I would be highly appreciated if anyone could help me.
Regards.
M.
This is basically what I had. I've changed the configs to match your setup and
I still
can't get incoming calls. I have the exactly same problem with inWeb as well.
The configs are:
iax.conf:
[gradwell]
type=user
username=xxx
secret=YYY
context=gradwell-in
host=dynamic
you can use the attached patch, to avoid the use of sipsak.
try the following lines in your extensions.conf:
exten = 99,1,Answer()
exten = 99,2,Set(_CONTENT-DISPOSITION=desktop)
exten = 99,3,SendText(Testmessage)
the patch is for asterisk 1.2.1 but should work on newer versions.
let me know if
jed is ?
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Mar 06, 2006 at 04:42:12AM -0800, Dovid
Bender wrote:
I use PICO (nano for CentOS). Works great.
The original pico is a buggy and horrible editor.
Nano has fixed some of
the bugs. Still: no jump to line, no decent
multi-level
BTW regarding AOC you may want to have a look into this:
http://www.snom.com/wiki/index.php/Advice_of_charge_%28AOC%29_in_SIP
On Friday 10 March 2006 04:37, Dofear wrote:
Can this feature be used to display total balance left (for a phone) on the
display of the phone?
_
From: [EMAIL
what kernerl are you using ? when did you did modprobe
zaptel and modprobe ztdummy it loaded without a
problem ?
--- Zach A [EMAIL PROTECTED] wrote:
Hi,
As I am having problems with MoH and have tried
everything to solve it
and nothing worked, I was thinking maybe the timing
source, i.e.
How can i configure the following scenario,
- User 'A' dials into Asterisk,
- Asterisk puts user 'A' on hold
- Dials Out to User 'B'
- Consults user B' if he wants to take the call (Press 1)or divert to voicemail (press 2)
- Depending on the option chosen,either user A' call is bridged with the
On 03:02, Fri 10 Mar 06, Dovid Bender wrote:
jed is ?
An editor
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
I guess this is the right way. That's the context's job. Don't forget to
setup your users'es context in iax.conf and/or sip.conf according to
contexts you just created in extensions.conf
For instance:
# iax.conf/sip.conf
[user_grp_A]
context=grp_A]
...
all_the_other_stuff
[user_grp_B]
No, you don't.
John Joseph wrote:
Hi All
I am working on asterisk + digium developer card
, it has on FXO and one FXS
I want to work asterisk in the following way
1FXO connected to PSTN line
2the calls coming to PSTN line should be received
3SPI clients should be able to call outside
Hi,
zapata.conf and zaptel.conf has different sintaxes. In zaptel.conf, you
should use only:
fxsks=3-4
The zapata.conf of yours seems to be ok, but I don't see the group=1 on
it. And that -- Executing Dial stuff has nothing to do with receiving
a call from PSTN.
[]'s
MM
[EMAIL
Hi,
I have some problems transfering
call from phone to phone with my Asterisk. When I dial Flash I can hear for half
a second the dial tone, but it stops suddenly. The other phone hear the on hold
music and pressing flash key another time I get back to the previous
channel.
On the
Or just write the rules needed, like:
exten = _1XX0[1-4].,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1XX0[5-9].,1,Dial(SIP/[EMAIL PROTECTED])
and so on.
[]'s
MM
C F wrote:
you will have to do that in the gotoif statement, something like this:
exten = _X.,1,GotoIf($[${EXTEN:2:1}=1]?20)
exten =
Hi Sharath,
Not only possible but already been done before. Normally involves
user A speaking their name and that audio file being played down the line to
user B who can then decide if he wants the call.
Have fun coding but as its been done before my
suggestion is you post a RFP and
Hi
On fresh install ( both OS and Zaptel ) after
modifying /etc/zaptel.conf
as
fxsks=2
loadzone=us
defaultzone=us
when I do ztcfg -vv , I get the following
error
Channel map:
Channel 02: FXS Kewlstart (Default) (Slaves:
i am trying to install sangoma a200d to my centOS server but i am receivig
this error message:
ZT_CHANCONFIG falied on channel 1: invalid argument (22)
Please help me on hopw to solve this issue! tnx!
Have you called Sangoma's tech support number?
I just implemented the same card about
Hi all,
I currently have an Asterisk test server behind a TZ170 Sonicwall
firewall / NAT box, with several DIDs.
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by
thanks yep . it should ok . let me try
thank again
Salaque
On 3/10/06, Melcon Moraes [EMAIL PROTECTED] wrote:
I guess this is the right way. That's the context's job. Don't forget to
setup your users'es context in iax.conf and/or sip.conf according to
contexts you just created in
Hi all,
I currently have an Asterisk test server behind a TZ170 Sonicwall
firewall / NAT box, with several DIDs.
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by Sonicwall, so I thought I'd pass it along.
I've used the iaxcomm softphone and a snom 200 behind serveral different
On 3/10/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
Hi all,Has anyone else run into this, or figured out the rationale for it?
I've noticed the same thing on my TZ150. I'll try that setting this weekend and see if it makes a difference.
-Brian
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
thanks
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I ordered a new Sangoma card for testing with Asterisk last year. It
worked great with our SBC PRI but I ended up using a Lucent TNT
gateway instead for the PRI connection. It allowed us to share the
ports with our dialup modem customers.
The A101 card is available on Ebay in case someone
Hi;
I've been asked to look at a large asterisk system implementation, which
would be a candidate for either a large cluster of PCs or a smaller
cluster based on Signate's SGI box(es).
I've waded through the requirements document, and I think I have more or
less all of the requirements covered
On 12/21/05, richard Coco [EMAIL PROTECTED] wrote:
Hi,we have interconnected Asterisk with a HiPath4000 V1.0using a H.323 Trunk. You have to install the oh323channel from [1]. On your HiPath4000 V1.0 or V2.0 youneed a HG3550 board for IP-Trunking.
If you have the version 3.0 then the HiPath
What about ResetCDR() just before Dial()?
nik600 wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
thanks
Hi all,
I want to link three incoming Bell Canada centrex pstn lines (which
currently go to an old norstar pbx) into asterisk.
Can anyone suggest the most painless (i.e., just works) way to do
this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter,
or the twice-as-costly
Thank you for the
support.
Filipe Mordhorst
Joinville - SC - Brasil
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Wai Wu
Enviada em: sexta-feira, 10 de
março de 2006 00:15
Para: Asterisk
Users Mailing List - Non-Commercial Discussion
Assunto: RE:
Hi All,
I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on
5060 and asterisk on 5065.
The setup is that people use serweb to create an account and register a
phone. Their calls are routed from ser to asterisk and then inbound on
IAX2.
The server has a public and an
Does anyone know if asterisk can detect and handle if a phone is
forwarded in the dialplan?
Aaron
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On Fri, March 10, 2006 14:49, Dr. Michael J. Chudobiak said:
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by Sonicwall, so I thought I'd pass it
along.
I've used the
Rich Adamson wrote:
i am trying to install sangoma a200d to my centOS server but i am receivig this
error message:
ZT_CHANCONFIG falied on channel 1: invalid argument (22)
Please help me on hopw to solve this issue! tnx!
Have you called Sangoma's tech support number?
I just
Hi , where can i find examples of dial plans using ring/hunt group feature?
thanks
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John Joseph wrote:
Hi
On fresh install ( both OS and Zaptel ) after
modifying /etc/zaptel.conf
as
fxsks=2
loadzone=us
defaultzone=us
when I do “ztcfg -vv “ , I get the following
error
Channel map:
Channel 02: FXS
Hi All,
I'm wondering if I use DUNDi in an enterprise environment, I own and manage all
the servers, can I make 1 set of public and private keys and exchange them
between servers or do I need to make a key for every server? Is there
something with astgenkey/openssl that uses any server
Hello list,
Three questions on dialplan commands/functions:
- cmd BACKGROUND:
a) If i just want to allow user to send DTMF ONLY while the message is
played (with no additional time after ), does TIMEOUT RESPONSE = 0 make
sense or this action produce some border effect ?
b) if YES (border
Mike Clark wrote:
snip
Have you called Sangoma's tech support number?
I just implemented the same card about two weeks ago and really didn't
have any installation issues using fc3 and trunk, however their
documentation is a little on the rough side. Install info seems to be
a little in one
Hi,
I wrote this thread to find someone who uses this. Please write if u r
using this as i would like to clarify regarding disconnect
supervision.
Thanks
On 08/03/06, Wireless [EMAIL PROTECTED] wrote:
Does anyone have this working on 1800MHz eg TMobile or Orange in the UK
and does CLID work
Hi Waldo,
We have a friend who does the recording for us. She is very good, if you are interested contact me.
FelixOn 3/9/06, Tom [EMAIL PROTECTED] wrote:
I have one that we work with.Digium also does this with Allison.Contact me off list for more info.TomAt 05:19 PM 3/8/2006, you wrote:Can
I have had a tough time finding a good stable combination of
motherboard with either a TE410P or TE406P and would like to solicit input for
suggestions. Currently, we are utilizing the Supermicro P8SCT. I have disabled
the onboard NIC cards and I am currently using a separate 3COM PCI
OK, so apparently no one is using
GnuDialer, is anyone out there using any other predictive dialers on asterisk?
Thank you,
Adam Vocks
From: Adam Vocks
Sent: Thursday, March 09, 2006
12:41 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject:
Aaron,
of course it is possible. r u using AMP? if positive, look atcustom extension.
there you can define a forward application so using the proper process , you can have there the required forwarded destination
all the best,
Mickey
On 3/10/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Does anyone
At 03:39 PM 03/09/2006, you wrote:
you will have to do that in the gotoif statement, something like this:
exten = _X.,1,GotoIf($[${EXTEN:2:1}=1]?20)
exten = _X.,2,GotoIf($[${EXTEN:2:1}=2]?30)
Close, but it needs spaces around the equal signs, like:
exten = _X.,1,GotoIf($[${EXTEN:2:1} = 1]?20)
On Fri, 10 Mar 2006 15:04:18 +0100
nik600 [EMAIL PROTECTED] wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
Hello!
There's the g-option for the Dial-cmd that allows to execute the next
extensions in the current context when the callee hangs up.
I would need the same for a call where the caller hangs up, concretely
i have to inform a agi-application of the end of a call. Does someone
know a way to do
Adam Vocks wrote:
OK, so apparently no one is using GnuDialer, is anyone out there using
any other predictive dialers on asterisk?
Thank you,
Adam Vocks
*From:* Adam Vocks
*Sent:* Thursday, March 09, 2006
Hello,
I'm not detecting caller Id through DTMF with my TDM400 card and don't
know how to put this to work. Could someone please help me ? How should
I configure * to achieve this?
Thanks,
Fernando
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It sounds like they referring to a load balancer
that would provide one live external IP and handle sessions for multiple
internal servers.
_Mobilcomhttp://www.mobilcom.net
- Original Message -
From:
Gabriel
Afana
To: Asterisk Users Mailing List
I have been looking at the medium-rate codecs in Asterisk - ADPCM and
G.726. Both of these are adaptive PCM codecs - the G.726 one is a little
more expensive in processing power, however both are 32k bit-rate.
I am experiencing problems using G.726 where the audio level is high. It
produces loud
We're not using AMP, we customized WAY too much stuff to be able to do
that. What I'm looking for is a fix for the moved temporarily response
we receive from phones so we know if someone forwarded their phone to a
bad number or not.
Aaron
Tele Cost Price Reducer wrote:
Aaron,
of course it
you are using gnu dialer? i dont find too much info
about that, i work in a call center i'm interesed in install this
soft
- Original Message -
From:
Adam Vocks
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, March 10, 2006 11:07
AM
wath is the link of the vcidialer?
Vladimir Montealegre Estailes
Bogota-Colombia
Este Mensaje Esta Hecho 100% con Electrones Reciclados
- Original Message -
From: Saul Diaz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
i've got 3 queues:
QUEUE A
- (mon,tue,wed,thu,fri) 8-12/14-18 - Queue the call
else play a mex reporting the service period
QUEUE B
- (mon,tue,wed,thu,fri) 8-12- Queue the call
else play a mex reporting the service period
QUEUE C
- (mon,tue,wed,thu,fri,sat,sun) 8--20 - Queue the call
else
Is there an easy way to disable flash transfers? I'd prefer the users hit # to
transfer, since some users are hanging up a call, then dialing another one
without giving the handset enough time to actually hangup the call, so it
appears that they are transfering the 'ended' call to the new
Sean,
I can concur as far as the
comments hereregarding the sipsak syntax. We use sipsak to update
the display on our phones so our agents know if they are logged in or logged
out.
The sipsak syntax we use with good results
is:
sipsak -M -O deasktop -B "(your message)" -r 5060
-s
Rich Adamson wrote:
Hi all,
I currently have an Asterisk test server behind a TZ170 Sonicwall
firewall / NAT box, with several DIDs.
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly
Hi
iam also looking postpaid and prepaid solution
ram
On 3/2/06, Dovid Bender [EMAIL PROTECTED] wrote:
what do u mean by solution ? please define what youwant.--- ram
[EMAIL PROTECTED] wrote: Hi did you got any solution iam also looking the same solution if you find kindly tell me which
It's in zapata.conf
On 3/10/06, Dan Elder [EMAIL PROTECTED] wrote:
Is there an easy way to disable flash transfers? I'd prefer the users hit #
to transfer, since some users are hanging up a call, then dialing another one
without giving the handset enough time to actually hangup the call, so
rtp debug will tell you if Asterisk is handling the rtp for what IP.
On 3/10/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 9, 2006, at 9:30 PM, Marc Archer wrote:
Hi All,
This is probably a stupid question, but I'm trying to figure out if I
Asterisk is in the middle of the media
Whisker, Peter wrote:
I have been looking at the medium-rate codecs in Asterisk - ADPCM and
G.726. Both of these are adaptive PCM codecs - the G.726 one is a little
more expensive in processing power, however both are 32k bit-rate.
I am experiencing problems using G.726 where the audio level
Anyone have the 7970 xml config for sip yet?
Aaron
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On Fri, 2006-03-10 at 11:52 -0600, Aaron Daniel wrote:
Anyone have the 7970 xml config for sip yet?
Aaron
[EMAIL PROTECTED] ~ $ cat SEP0014A89EF5E3.cnf.xml
device xsi:type=axl:XIPPhone ctiid=203849429
uuid={96f8508b-10ef-f98c-d20d-0471777ec725}
fullConfigtrue/fullConfig
I have seen this in the archives any number of times -- but I have yet to
find a solution that works for me. Any suggestions you can offer would be
most appreciated.
Here are the answers to the questions I have seen asked in earlier
posts. The log contains:
Mar 10 12:26:29 WARNING[3015]
Hi all,
I currently have an Asterisk test server behind a TZ170 Sonicwall
firewall / NAT box, with several DIDs.
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by
Hi Miguel, probably not what you are after but if you install
[EMAIL PROTECTED] it's a great way to see how AMP have implemented their
dial plans.
Other than that there are hunt codes samples on the www.voip-info.org
Cheers
Dean
-Original Message-
From: [EMAIL PROTECTED]
Dean Collins wrote:
Hi Miguel, probably not what you are after but if you install
[EMAIL PROTECTED] it's a great way to see how AMP have implemented their
dial plans.
Other than that there are hunt codes samples on the www.voip-info.org
Cheers
Dean
thanks Dean, i read the wiki and
Or maybe, you should try the C flag in your Dial().
[]'s
MM
nik600 wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
Hi
I would like to say that I'm a happy user of it. But
currently I'm still waiting for support from CT.
My gateway is here, but it's doing nasty things,
so currently I'm not a good source for information.
Sorry
Adibar
On Fri, Mar 10, 2006 at 07:03:25PM +0300, [EMAIL PROTECTED] wrote:
Miguel wrote:
thanks Dean, i read the wiki and found that its easier than i thought,
i have this ring group but i want that if the extension doesnt answer,
the call ends there , i mean, if theres no one available only the
first extension will ring, if nobody answer, automatically hangup,
ok, thanks for your reply, tomorrow i'll test and let you know
bye
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Hi adiar,
Just have a few questions. Hope u can give me the answers.
Well im currently having analog lines to my asterisk which is used to
bridge calls from PSTN to Voip (DISA). Im facing issues with hang up
supervision etc. Im thinking of bridging calls using the GSM gateway
in future. Just
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am looking to trade for a new or used Sangoma Analog A200 card with
echo cancellation. I have finished my testing with the OpenSwitch
card and want to test with the sangoma. Anyone out there looking to
do the same?
Sean
-BEGIN PGP
Hello,
I'm having an apparent issue where caller id name isn't coming through
my IAX2 channels. The name shows up in the asterisk cdr log, but my IAX2
application doesn't receive it. I'm running asterisk 1.2.4.
Is this a known problem or config issue?
Thanks!
Hello All,
Ive been doing more and more research on trying to setup a cluster/load
balancer for Asterisk. All the Asterisk boxes would be using a config
that is the same between them all (via a DB), but we want one location
to point the phones to, and from there that machine/device will send it
Hi all,
Can somebody tell me, how to set no crc4 in my zaptel.conf?
I connected my A104D Sangoma card to E1/isdn and each I tried to make call I
get the errors below.
My protocol analyzer can see only setup info and release complete.
WIRELESS2*CLI set debug 9
Core debug was 0 and is now 9
That's because the duration is counted from the time of dialling. billsec is
what you want if it's to calculate the duration the call was active.
To change what shows you need to change call-log.php in
/var/www/html/admin/cdr/
Instead of duration extract billsec - you can still label it
If you
implement multiple Asterisk systems, your challenge is going to be in ensuring
that phones registered to one Asterisk know how to reach phones registered to
another Asterisk system. Good luck with that!
Doug.
-Original Message-From: Ron McCarthy
[mailto:[EMAIL
On 3/10/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
OK apart of my beleive that sonicwall is a piece of crap (personal), try to do a port forwarding for the IAX port (4569)
Some people always subscribe to the Get what you pay for theory. Since they are usually priced on the low-side
Yes
On 3/10/06, Dan Elder [EMAIL PROTECTED] wrote:
It's in zapata.conf
is that
transfer=yes
if I set this to no, does this keep the # transfer functionality that is
setup w/AAH?
Thanks
___
--Bandwidth and Colocation provided by
Would an internal DUNDi configuration help the asterisk servers share
their extension info? Or, use e.164 with an internal DNS zone to
lookup the routing information. SIP phone logs into Asterisk 'A' and
a script runs to update the e.164 DNS info pointing the DID to
Asterisk 'A'
Hello,
I have used GnuDialer in a test environment and it does work. There
isn't much documentation out there on it but it is in production at
several sites. You should go to the GnuDialer website and post on
their forums for more information.
http://www.gnudialer.org/
The other GPL predictive
Awesome, that works, 'cept now the dialplan doesn't work lol. I've
programmed the voicemail button in, but anything I try to dial doesn't
make it past the first digit.
Aaron
Greg Oliver wrote:
On Fri, 2006-03-10 at 11:52 -0600, Aaron Daniel wrote:
Anyone have the 7970 xml config for sip
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