On Jan 17, 2005, at 7:29, Joseph wrote:
How do I solve the problem with between patterns:
_1800
_1NXX
I would like all numbers 1800, 1877 etc to go through iaxtel
but all other numbers 1xxx via voipjet
When you combine these contexts, e.g. when you include them in your
default context, you need
Hi,
Is there a way of adding SIP clients using AGI ? I see that, only
extensions can be added using the AGI.
If not AGI, is there any other way of adding SIP clients other than
editing siop.conf manually ?
Thanks,
~Vamsi
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Hello Phil im from Guatemala, im living in Madrid but im thinking in came
back in july, if its helps to you, im thinking in make an installation of
asterisk to make calls, if you found something now to make calls please
inform me!
TIA
Edgar
I'm looking for a company that offers Guatemala
On Mon, 17 Jan 2005, GRD wrote:
But when trying to give a call, i'm always receiving not able to open
Zap channel from my asterisk box ...
Just a thought - are the permissions on the device nodes under /dev/zap/
correct? This is only an issue if running non-root of course.
Peter
I have. I use own developed AGI radius script for auth and acct.
Also I rewrote minitelecom radius module for CDR radius generating.
On Fri, 14 Jan 2005 15:31:16 -0300, Tenorio, Leandro
[EMAIL PROTECTED] wrote:
I'm currently trying to use a Radius server for acct and auth, cause
much of our
On Fri, 2005-01-14 at 14:38 -0800, Ben Greear wrote:
Hello!
I am trying to set up multi-link PPP using two T100P cards in one
machine, and 1 T405P card (the 4-port one) in another machine. I have
previously been able to get PPP working between the two T100P cards
in separate machines
Vamsi Pottangi wrote:
Hi,
Is there a way of adding SIP clients using AGI ? I see that, only
extensions can be added using the AGI.
If not AGI, is there any other way of adding SIP clients other than
editing siop.conf manually ?
Thanks,
~Vamsi
Hi
You can do this using Asterisk RealTime, which uses
Am Montag, 17. Januar 2005 09:14 schrieb Vamsi Pottangi:
Hi,
Is there a way of adding SIP clients using AGI ? I see that, only
extensions can be added using the AGI.
If not AGI, is there any other way of adding SIP clients other than
editing siop.conf manually ?
Thanks,
~Vamsi
You can of
Am Samstag, 15. Januar 2005 22:39 schrieb Scheda:
Hey, whenever I try to load, I get these errors
Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to
bind to 0.0.0.0 port 4569: Address already in use
Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource:
Do anyone have experiences with Euro ISDN in Sweden?
Does CallerID work properly? Both in and out.
Do anyone know of a reseller for Digium cards and/or CarrierAccess Adit 600 in
Sweden or Europe (EU)?
Thanks!
BR
Daniel Nyström
___
Asterisk-Users
Hi,
what happens if the dialplan contains something like
exten = s,1,AGI(agi://10.0.0.1)
exten = s,2,Dial(SIP/phone1|20|tr)
etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my
test cases, I always got a hangup and no further processing of the dialplan.
Any hints? (
Hi All,
Im new to *
I wonder if anyone have an idea how to make the following
with ASTCC.
I would like to have all SIP phones to work on prepaid basis
and without need to dial any access number, instead I would like to use the
phone as normal dialing only the destination number,
Hi,
I'd like to setup automatic recording of channels and send wav files via
email to extension user (to same email address as in voicemail.conf). Can I
access those addresses from dialplan or AGI ?
Regards,
Rob.
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Hello.
I have just arrived to Asterisk. I would like to know if Asterisk can
perform some functionalities I am looking for.
I want to allow voip over sip to some users. All of them must have
their own user name and password to login to Asterisk so only allowed
users can login. All calls started
Am Montag, 17. Januar 2005 11:26 schrieb Alberto Martínez:
Hello.
I have just arrived to Asterisk. I would like to know if Asterisk can
perform some functionalities I am looking for.
I want to allow voip over sip to some users. All of them must have
their own user name and password to login
Also the following has worked great for me:
http://www.wifive.net/introduction.asp
Michael
Radovan Mihalik wrote:
http://www.sjlabs.com/sjp.html
SJphoneR is a VOIP softphone that allows you to speak with any PC, PDA,
stand-alone IP-phone and with any legacy wired or mobile phone (using
your VOIP
after # vi Makefile
and changes in coment the ligne with gcc
i have the same error before
In file included from /usr/include/linux/kernelcapi.h:13,
from /usr/include/linux/capi.h:18,
from chan_capi.c:35:
/usr/include/linux/list.h:604:2: warning: #warning
Am Montag 17 Januar 2005 11:26 schrieb Alberto Martínez:
Hello.
I have just arrived to Asterisk. I would like to know if Asterisk can
perform some functionalities I am looking for.
I want to allow voip over sip to some users. All of them must have
their own user name and password to login
hi vincent,
Vincent Guidoux schrieb:
I have a problem for install chan_capi
My pc: Suse 9.1, with asterisk current stable en cvs
And patch the chan_capi
chan_capi.c:1076: error: structure has no member named cid
as you are writing and apparent to the error message you are posting,
you are using
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make a dialplan without fixed extension
numbers to change the entire dialplan according to the customer
requests: what exten number do you want
On Sat, 2005-01-15 at 09:09 -0500, Doug Lytle wrote:
Mike Dent wrote:
Whilst on the subject of BT's, do the callers and called buttons function?
they dont seem to do anything on mine?
Yes, but the hand set needs to be off hook.
To add to Doug's reply...
---for people you have called---
1 -
On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make a dialplan without fixed extension
numbers to change the entire
I got this error while compiling:
configure: error: termcap support not found
I don't know how to solve this problem...
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hello Dave,
Monday, January 17, 2005, 12:50:13 PM, you wrote:
DC On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make a
Sorry... I have forgotten to say I am compiling from sources
downloaded from the asterisk web page.
AM I got this error while compiling:
AM configure: error: termcap support not found
AM I don't know how to solve this problem...
AM ___
AM
do a google search for
tdm400p hardware problems (fix)
This is a problem with the tdm card and driver
If you are using the older zaptel software the
file referenced in the doc is wcfxs.c if you
are using the cvs version the wcfxs file needs
to be replaced with wctdm.c also the line number
2127 is
Is there any way of logging all manager events to a file, similar to the
entries in logger.conf.
I was actually hoping that there was such an entry in the logger.conf
ManagerEvent = root,rootevents
This would allow someone to interrogate all events for a given user (in
this case root) from a
On Jan 16, 2005, at 8:45 PM, Joseph wrote:
What would be my best option to receive calls via VOIP.
I would like to use it as an alternative number when my main number is
busy.
The solution is not that easy as in order for customer to be a free
call
DID=Direct Inward Dialing provider would need to
Thank frank!
Now i have a un new prob
Went I make a call, the CAPI channel error
Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in
new stack Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request:
didn't find capi device with outgoing msn = 4202270. you should check
your
Hi,
I'd kindly ask for simple example if this is possible ?
Is any key press encountered during conversation and action taken in
dialplan ?
Thanks,
regards,
Rob.
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Asterisk-Users@lists.digium.com
Hi,
I'd kindly ask for simple example if this is possible ?
Is any key press encountered during conversation and action taken in
dialplan ?
Thanks,
regards,
Rob.
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Asterisk-Users@lists.digium.com
Here is the link
http://www.voip-info.org/wiki-ASTCC
SA
-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Enviado el: Martes, 14 de Enero de 2003 18:21
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] ASTCC
Dear Sebastian;
Thanks a
I am using the G729 stack from Intel with *.
But as far as I know the Grandstream can just connect with PCMU and * will
transcode the audio into G729, right?
Because I know that iaxcomm and SJPhone for sure do not support G729 but I
can connect with those clients.
Maybe I can try to completely
-Original Message-
From: Joseph [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 1:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] pattern matching problem
How do I solve the problem with between patterns:
_1800
_1NXX
I
-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 7:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Can I start recording channel in
the middle ofconversation ?
Hi,
I'd kindly ask
hi,
i have a problem with distorted voicemail sound on our asterisk test
machine.
i'm using cvs-head (2004-01-17) and chan_capi 0.3.5 (with my patches to
make chan_capi compile with asterisk cvs-head) and a diva quad-bri isdn
card.
other things work well with my setup (dial in, dial out,
On Mon, 2005-01-17 at 12:56 +0100, Alberto Martínez wrote:
I got this error while compiling:
configure: error: termcap support not found
get termcap development package installed for your distribution.
--
Steven Critchfield [EMAIL PROTECTED]
___
Vincent Guidoux schrieb:
Now i have a un new prob
Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in new
stack
Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request: didn't find
capi device with outgoing msn = 4202270. you should check your config
well the error message says it
/hax0r n00b mode on
Which command and parameters do I need to use to get some legible (usable)
output to do the packet sniffing? I tried ethereal but it only gives me
loads of garbage?
/hax0r n00b mode off :)
Go to the Wengo forum, there is a thread in the technique section that
gives the
on 'make'
chan_sccp.c: In function `load_module':
chan_sccp.c:653: warning: passing arg 4 of `ast_channel_register_ex'
from incompatible pointer type
Now compiling sccp_actions.c 743 lines
Now compiling sccp_channel.c 279 lines
sccp_channel.c: In function
On Mon, 10 Jan 2005 19:38:23 +, John Middleton
[EMAIL PROTECTED] wrote:
Not an enterprise level system, but anyone used the www.intertex.se IX66?
Yes they work great no nat traversal issues,
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Asterisk-Users mailing list
According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is
this?
This should not be a problem for Asterisk?
Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN?
BR
Daniel Nyström
___
Asterisk-Users
I use iaxcomm (a little finicky to get working since it needs wxgtk files) and
linphone
Linphone is SIP and is a little trickier to get working when dealing with NAT
than iax (iaxcomm)
I chose these 2 because they seemed the easiest to get working.
They both work fine but are not as good sound
Curiosity got hold of me. I opened up my BT-10 (and it still works
afterwards..)
Under the keyboard (buttons) are four red LED's that appear to run in
parallel (they all flash at the same time when you put the power on).
These are used to light up the keyboard.
The Display LED (blue in my case)
Hi Dan, Steve, Michael, Bruno and others.
I will try to describe my VoIP environment below:
SERVER:
- FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
- iax.conf
[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
tos=lowdelay
jitterbuffer=no
[EMAIL PROTECTED] wrote:
According to CarrierAccess, the Adit 600 uses CAS for voice
signalling. What is this? This should not be a problem for
Asterisk? Does the Asterisk server need to reencode CAS into aLaw
when going to Euro ISDN?
Try this:
I'm just using default installation whatever Gentoo is providing; this
is their stable version.
Joseph,
While I also use Gentoo(as do many others), most will tell you NOT to
install * from portage. You can save yourself trouble by getting 1.0.3
or CVS and ditch the builds
On January 17, 2005 01:47 am, John Sellens wrote:
Just on the off chance that Canadian Asterisk users might be
interested in a place to discuss topics specific to the great
white north (sources, services, telcos, etc.), I created
the asterisk-canada mailing list:
I know as a Canadian I'm not
I'd be interested in a possible mailing list for a United States
Asterisk mailing list. We are in the very beginning stages of building
a pilot system using Asterisk, but based on the information I've found
on the internet so far, it looks very promising to scale the system to
our needs.
I'd be
Hi,
I'm having some problem with realtime:
let's say I have a dialplan like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
A call from ip phone 300 comes in, and it's been answered.
Then I switch the sip context to realtime, putting the exten in the
db and using the
Hi,
I want to create this system :
Desk1 SIP Phone adsladsl Desk2 SIP Phone
|
|
adsl
Desk3 asterisk Server
My question is : when Desk1 call Desk2 ,
Good Day List,
I have my asterisk box setup to be an ntp server, and my zultys
4X4 phone is able to get the time, however
I must first select the TimeZone Offset and then it will use the
time setting from my server.
This is a hassle because every time the phone reboots
For what it's worth, I'm working with Zultys trying to solve this exact
same problem. So far, they've told me to take an ethernet trace, because
they claim the DHCP option 42 isn't being sent, but I know this is not the
case, because the phone knows the time, just not the time zone. There is
a
I would like to have all SIP phones to work on prepaid basis
and without need to dial any access number, instead I would
like to use the phone as normal dialing only the destination
number, for example 00464090510.
I use the AccountCode for authentication. This is how, for example:
exten =
I believe you can specifiy header information. If you know what portion of
packet deals with that information, you (in theory) would be able to do it.
On Monday 17 January 2005 02:52 pm, Nabeel Jafferali wrote:
I would like to have all SIP phones to work on prepaid basis
and without need to
I know as a Canadian I'm not interested in a list Just for Canadians -- It's
just fragmenting the help available for very little benefit. I do, however,
appreciate the thought.
-A.
I'm a Canadian also, and I second that
___
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Hi all
I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.
I need to now the name os de file oraspecific category link where i can download it.
If you can send me the file is beter ;-)
Thanks inadvance
Regards
Wert
Do you Yahoo!?
Yahoo! Mail - Find
In the next couple of weeks we will be starting the beta phase of our
Guatemala POP. If you could wait, welcome.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Astin
Sent: Sunday, January 16, 2005 6:23 PM
To: [EMAIL PROTECTED];
I can now place calls from Grandstream (via Asterisk) to mutualphone.
I did this by disabling the G729 (and G723) codecs in the Grandstream, so
that * takes care of any recoding.
What it looks like is that the G729 stack of the BT101 is not compatible
with the one that mutualphone is using.
Same here, interested in the details of a SS7/Asterisk solution.
Regards
MIKE
Steve,
I also would be very interested in getting those details. We would very
much like to move forward with SS7, please feel free to contact me off
list.
Cheers,
Ben Merrills
Griffin Internet
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.
What is IOS? Am I the only one who uses Cisco phones and doesn't know that
acronym?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN:
I know as a Canadian I'm not interested in a list Just for
Canadians -- It's just fragmenting the help available for very
little benefit. I do, however, appreciate the thought.
I disagree. I have joined the new list and feel that as long as it is
focused on discussions like:
- DIDs in Canada
On Mon, 17 Jan 2005, Wilson Pickett wrote:
/hax0r n00b mode on
Which command and parameters do I need to use to get some legible (usable)
output to do the packet sniffing? I tried ethereal but it only gives me
loads of garbage?
/hax0r n00b mode off :)
Go to the Wengo forum, there is a thread in
So it seems me that with realtime we cant'use variables as extensions
for an easyer manteniance of the dialplan.
Doesn't RealTime itself make for easier maintenance of extensions since
its database driven?
-Matthew
___
Asterisk-Users mailing
Two more information:
1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or
better, it is very less(some miliseconds) than the begining(10 seconds) of
a call.
Any ideas!?
Denis.
Em Seg 17 Jan
Hi Denis,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
...
- Same problem with DIAX oldest DLL;
It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0
Please try an older version of DIAX, like 0.9.8c.
You can still download it from:
http://www.laser.com/dante/diax/diax098c.zip
or
I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on
this and I think the issue may be related to the setting of the Time
Offset
3.4. Time Offset
The time offset field specifies the offset of the client's subnet in
seconds from Coordinated Universal Time (UTC). The offset
On January 17, 2005 10:26 am, Nabeel Jafferali wrote:
I disagree. I have joined the new list and feel that as long as it is
focused on discussions like:
- DIDs in Canada
That's a -biz question
- VoIP taxes/regulation in Canada
While not specifically -biz, all that can be said on that at
the image file to get it working with asterisk
sorry for the acronym
wertNabeel Jafferali [EMAIL PROTECTED] wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.What is IOS? Am I the only one who uses Cisco phones and doesn't know that acronym?--
Denis Galvão - iSolve wrote:
Two more information:
1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or
better, it is very less(some miliseconds) than the begining(10 seconds) of
a call.
Any ideas!?
Oh, thanks!
How do I know which codec is used on Adit 600?
Does the server need to reencode it at all, or is the codec the same on Euro
ISDN?
If it has to reencode everything, it really seems to be CPU critical when using
30 FSX lines into 30 Euro ISDN lines.
Btw, when using Adit for connecting
I too had the same problem - it fixed itself the other day.
Of course, it was five days after reporting the problem with no response
from NuFone...additionally, if I attempted to call and # in the 707 area
code the call would not go through.
The other problem that I find with NuFone is the CLEC
On Mon, 17 Jan 2005 07:11:20 -0800 (PST), R A [EMAIL PROTECTED] wrote:
Hi all
I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in
the cisco web page.
I need to now the name os de file or a specific category link where i can
download it.
If you can send me the
On Mon, 17 Jan 2005 10:23:57 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.
What is IOS? Am I the only one who uses Cisco phones and doesn't know that
acronym?
Internetworking Operating
Nabeel Jafferali wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.
What is IOS? Am I the only one who uses Cisco phones and doesn't know that
acronym?
Internetwork Operating System (I. O. S. or IOS). If I remember
correctly, the phones don't
I have had the same problem when calling across Asterisk from Diax to a SIP
phone. If Asterisk Answers the call before the Dial to the SIP phone
there is no delay. Otherwise there is a 10-20 second delay in the Voice
path!
Peter
-Original Message-
From: Dan [mailto:[EMAIL PROTECTED]
Just my CAD$0.02 though.
C'mon, at least throw in a loonie :P
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
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the image file to get it working with asterisk
You mean the SIP image? Look at the Wiki, there's info on what service
contract you have to buy from Cisco to get access to it. I believe
technically, although the service contract gives you access to the
image, you need a specific SIP image license
you can call it firmware upgrade for SIP
i just want to register this phone with asterisk,
I apreciate any other idea to do it.
thanks in advance
wert
Nabeel Jafferali [EMAIL PROTECTED] wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.What
Em Seg 17 Jan 2005 13:51, Steve Kann escreveu:
Yes, it sounds like there's a discontinuity in the timestamps when you
set up your call, but it seems Dan can't reproduce this.
The fix is probably:
a) The jitterbuffer needs to be reset after the transfer, or
b) The timestamps sent need to be
Em Seg 17 Jan 2005 13:43, Dan escreveu:
Hi Denis,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
...
- Same problem with DIAX oldest DLL;
It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0
Please try an older version of DIAX, like 0.9.8c.
You can still download it from:
GSM Codec is 13k plus overhead. That may work?
Peter
-Original Message-
From: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Sent: 15 January 2005 07:07
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DIAX
Dear Dan;
Thanks alot for your kindly reply.
Well, what u advise us to
Hello Matthew,
Monday, January 17, 2005, 4:34:16 PM, you wrote:
So it seems me that with realtime we cant'use variables as extensions
for an easyer manteniance of the dialplan.
MB Doesn't RealTime itself make for easier maintenance of extensions since
MB its database driven?
So this is
No. You should work on configuring xlite to register with asterisk.
In the xlite Sip Proxy menu, you will need a User Name, Password,
Sip Proxy, and Domain/Realm defined to match entries in your
sip.conf definitions.
to which entry have to corespond Domain/Realm parameter in X-lite
Hello I have a 800 DID setup to dial into my Asterisk server and I'm
wondering if it's possible to ID when it's a payphone or not? I
suspect it's not since I'm getting calls from someone else's SIP or
IAX box.
If I had a digium card installed and connected to a couple lines would
I be able to
Hello All,
Please forgive the lack of understanding as of yet but I have been trying
to follow the mailing list messages over the last few days and would like
to know if someone could wither point me into the right direction or
possibly give me a brief overview of the complete process.
That was the hint I needed. Try adding this to your dhcp.conf:
option time-offset -480
(-480 is for PST, -420 is mountain, etc.)
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 17 Jan 2005, Ronald Hartmann wrote:
I have been reading the RFC
[EMAIL PROTECTED] wrote:
On January 17, 2005 01:47 am, John Sellens wrote:
Just on the off chance that Canadian Asterisk users might be
interested in a place to discuss topics specific to the great white
north (sources, services, telcos, etc.), I created the
asterisk-canada mailing list:
I
Digium is the company behind the Hardware to Asterisk.
Try its website:
http://www.digium.com
They have a developers kit that could reach your needs.
Denis.
Em Seg 17 Jan 2005 14:13, [EMAIL PROTECTED] escreveu:
Hello All,
Please forgive the lack of understanding as of yet but I have been
[EMAIL PROTECTED] is believed to have said:
In order to get the message button to work - programme it with the
extension number for your voice-mail. On your BT-100's phone web page -
it looks something like..
Voice Mail UserID:[300] (User ID/extension for 3rd party voice
mail system)
Are there affordable DIDs (preferably IAX) available anywhere outside
the US? I want to use it to meet ICANN requirements for providing a
valid phone number, yet pre-empting some of the telemarketing calls my
domain registrations generate. (Yes, I asked a similar question about
900# availability
Hello everyone,
Does anyone know how to change the TDB400 clocking to receive fax
properly (with spandsp) ?
I currently have some frames slips so I always receive the first line
of the fax. From what I saw from the Opencall bug tracking, we are
supposed to be able to change the TDM clocking.
Wouldn't be best to consolidate all these list. I've been on the list
less than a week and already have too much to read.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial
[EMAIL PROTECTED] wrote:
Hello All,
Please forgive the lack of understanding as of yet but I have
been trying to follow the mailing list messages over the last
few days and would like to know if someone could wither point
me into the right direction or possibly give me a brief
overview of
On January 17, 2005 10:55 am, Mark Halverson wrote:
Of course, it was five days after reporting the problem with no response
from NuFone...additionally, if I attempted to call and # in the 707 area
code the call would not go through.
Do you have the ticket # from your support@ email?
The
Hi!
Is there any way to receive in * server a call from
a Terminal adapter in G.723/G.729 and then convert it to G.711?
I'm wondering this because I can only place all
thru Broadvoice in G.711 but most of customers have ADSL connection with 128k
upstream, so the result is that they can
Cisco uses firmware for IP phones.
And the phone models that can do SIP are 7905, 7912, 7940 and 7960.
7902 can only use SCCP, but you can use SCCP with * with basic
functionality.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher L.
Wade
Since I want the PDAs to talk to Cisco CallManager, I think I should better
look for Skinny pocket pc clients. Isn't that correct!
Walid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Monday, January 17, 2005 12:52 PM
To:
Scratch this idea, I just rather have one list, and maybe a website to
see all the list together where I can type in my question to find an
answer quickly (sort of like Dell's support center)
-Original Message-
From: Gyrion, Larry M.
Sent: Monday, January 17, 2005 9:14 AM
To: Asterisk
Linux does not have it's own sip/h323 modules (ip_conntrack_sip and
ip_conntrack_h323), however I have found these modules available in the
Linksys WRT54GS open source firmware. Would it be legal to use these
modules with another Linux distribution (eg, RedHat, Gentoo, Debian..)?
--
Chris
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