I also thought it might be a coded mismatch. Maybe someone can explain
why outgoing calls work when incoming calls between the same phones
don't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, March 29, 2004 10:32
On this subject - has anybody managed to implement a method
of warning the caller that their call will expire? I've
written a similar PrePaid test app in PHP/mySQL, I can
calculate the timeout for the call and set the timeout etc -
but I'd like to warn the user with beeps 15seconds before
the call
Hello,
There is an agi script for this, but I use goto's in the extensions.conf. Its
not terribly efficient, but it gets the job done.
I tried the blacklist but it only payed attention to the callerid. The number
was completely ignored. I could only put in one WIRELESS CALLER, and there
are a
- Original Message -
From: Robert Wilson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 2:14 AM
Subject: Re: [Asterisk-Users] external SIP calls newbie question
Chris Stenton wrote:
I have configured a basic * box which allows external sip calls in. This
Rich Adamson wrote:
Wanta take a guess what would happen if Cisco decide to really enforce
the legal rules?
I'll bite:
Their market share would plummet in all their markets, and then smaller,
more innovative companies would become more able to compete with them,
and the overall marketplace
Now could Asterisk
- Be configured to allow the phone (ideally IP phone) to display which
real number is being called (ideally a name for that number)
( sales line ringing, support line, etc.)
YES
- Cause only phones part of a group to call if the number related to
that group rings
Well, it is what he asked for, perhaps it was because I didn't do all of it for him,
since I wanted him to learn rather than just copy...
Let me explain:
John : The scenario is that I want all calls originating from number x to be
routed to a particular extension
exten =
- Let the caller know its position in the queue (ie: you are number #
in the queue, please hold and an operator will hang on you)
This is not possible at the moment.. Anyone know better?
Actually it is possible have a look at the bug tracker - I would give you the url
but I can't get to
Hi,
On this subject - has anybody managed to implement a method
of warning the caller that their call will expire? I've
Two questions;
Has anybody successfully implemented this, either by way of
source changes or by using the T extension (possibly
something obvious I've missed?)
I made
Hi,
I have several communication interemption, with followings errors messages :
Mar 30 16:02:29 WARNING[131081]: chan_zap.c:5952 zt_pri_error: PRI:
received TEI check request for TEI = 127
Sending TEI check resp ri=21786 tei=87
Mar 30 16:02:30 WARNING[131081]: chan_zap.c:5952 zt_pri_error:
Hello,
We are trying to deploy a new asterisk server with a Wildcard T400P (quad
T1) card. It uses a custom voice recording app written in the perl AGI.
Now that the machine has been in production, it seems to lock up within 24
hours of reboot! When it locks, we can ping the machine, but we
thanks a lot for the tips...
I was planning on connect the x100p on a extension line, so that i could
from the A pbx, dial that extension and get a tone signal on the B
pbx... they are rather old pbx, and that solution would really give me
some phone bill savings...
From: Joe Dennick [EMAIL
On Tue, Mar 30, 2004 at 03:46:34AM -0500, Brian Capouch said:
Rich Adamson wrote:
Wanta take a guess what would happen if Cisco decide to really enforce
the legal rules?
I'll bite:
Their market share would plummet in all their markets, and then smaller,
more innovative companies
Brian Cuthie wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roderick Montgomery
Sent: Monday, March 29, 2004 4:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
...
###
### Hardware != Software
###
Cisco IOS
You can insert a PlayTone(busy) at extension.conf to emulate this behavior.
Daniel
NetOne Administrator wrote:
As you see, * generates no busy tone, it hangs up the channel. It's
your client which generates the tone. This is not something to be done
from *.
Regards,
Doichin Dokov
Ryan
On one installation of asterisk, I have a display on the console
when I have a incoming call on my zaptel card. every digit was
displayed, this was great. Does anyone know how I can get this back?
Thanks
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I'm curious as to what a Washington State free phone number is? I live in
Washington State(Spokane) and we get our PSTN service from Qwest which is
certainly not free!
Does BroadVoice.com provide VOIP type DID? Do they use IAX?
Thanks,
Ed
-Original Message-
From: [EMAIL PROTECTED]
On one installation of asterisk, I have a display on the console
when I have a incoming call on my zaptel card. every digit was
displayed, this was great. Does anyone know how I can get this back?
Thanks
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Do normal modems work with asterisk?
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Title: DTMF not being detected on PhoneJack-lite
I'm trying to get a PhoneJack-Lite to work on my Asterisk box. I've actually gdb'd the code and it looks like I'm never getting any DTMF events.
Does the PhoneJack-Lite work with Asterisk? Are there some limitations with using it that I
Gary Franczyk wrote:
Hello,
We are trying to deploy a new asterisk server with a Wildcard T400P
(quad T1) card. It uses a custom voice recording app written in the
perl AGI.
Now that the machine has been in production, it seems to lock up
within 24 hours of reboot! When it locks, we can
Thanks to all those who replied. The anti ex-girlfriend facility seems
to be doing just what I was after. Maybe I just didn't have enough
ex-girlfriends; or maybe just not enough that turned into stalkers.
Regards,
John
___
Asterisk-Users
James Moran wrote:
Do normal modems work with asterisk?
Taken from the FAQ:
Can I use my modem to connect to the PSTN?
The answer is short: No you cannot. You'll need special telephony
hardware.
Further info under: http://www.voip-info.org/wiki-Asterisk+FAQ
HTH,
Martin
having problems with snom phone installstion
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I have also seen this here in the UK with the latest version (RC14) of the Zaptel BRI
drivers from Junghanns.net...Please can you confirm what hardware you are using?
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of zouhair echchelh
Sent: 30
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi.
I'm getting the following error at random intervals on my TE410P with Asterisk
CVS-03/30/04-11:49:01-CEST.
I have two spans active, one connected to my Telco, the other to a Siemens
PABX. Both spans display this behavior at random intervals.
Hello,
Thank you very much for your answers, it seems Asterisk can do what I
want which is a _great_ news.
Now, Could I ask if anyone have a good pointer for the
difference/similarities strength/weakness of Asterisk when compared to
Gnu/Bayonne which seems another possible solution.
I am sure
Hey all..
I just got my SPA-2000 and I am having an issue.
Whenever a call is answered, the init hangs up on me.
I am going to guess that it is a config option on the advanced / sip page,
but I am still waiting for sipura to give me access to the support page.
Could someone send me a screen
Hello All,
I've finally become so frustrated with my current router that I'm
seeking alternatives. I presently use a Linksys BEFSR81 which was
chosen for its QoS capability. However, that device quite routinely
loses WANLAN connectivity requiring a reboot. Sometimes it goes day
or weeks just
Title: Queue_log field definitions
Can anyone tell me the field definitions for the queue_log file in the Asterisk log directory?
1080593958|1080593892.0|salesq|NONE|ABANDON|1|1|50
Can anyone help me get call transfers working?
I have grandstream handytone-286 sip ATAs. Attached to these, I have
Teledex B150D telephones. Are there magic lines I need in my sip peers
to enable these folks to transfer? A call rings in at, say, 7145551212,
goes to x100, and they want x101.
Hello,
I have asterisk installed and working nicely for internal calls using
SIP. However, when I establish an outside call, it rings and connects
properly but I get no audio on either end (the call stays connected).
Asterisk's logs say the following:
-- Executing Wait(Modem[i4l]/ttyI0, 1)
Andy Powell wrote:
- Let the caller know its position in the queue (ie: you are number
# in the queue, please hold and an operator will hang on you)
This is not possible at the moment.. Anyone know better?
Actually it is possible have a look at the bug tracker - I would
give you the
On Tuesday 30 March 2004 06:25, John F. Baird wrote:
Thanks to all those who replied. The anti ex-girlfriend facility
seems to be doing just what I was after. Maybe I just didn't have
enough ex-girlfriends; or maybe just not enough that turned into
stalkers.
The anti-ex-girlfriend
The quick question: Do the digium drivers for the Digium Wildcard TE410P
(4 port T1/E1/PRI 3.3v card) , the T100P (single port T1), and the
TDM400P support hot plug PCI? I am also noting that while the TDM400P
doesn't state the voltage requirements, it looks like a 5v card. I hope
that I am
Michael Graves wrote:
Hello All,
I've finally become so frustrated with my current router that I'm
seeking alternatives. I presently use a Linksys BEFSR81 which was
chosen for its QoS capability. However, that device quite routinely
loses WANLAN connectivity requiring a reboot. Sometimes it
Senad,
I can do better than that:
http://bugs.digium.com/bug_view_page.php?bug_id=214
which says that the patches have been merged into cvs :D
HTH
Andy
*** REPLY SEPARATOR ***
On 30/03/2004 at 17:00 Senad Jordanovic wrote:
Andy Powell wrote:
- Let the caller know its
After almost a month of battling with them I've
cancelled my account at galaxyvoice.com.
My advice to anyone considering them as a voip provider
using * as a SIP client would be don't.
A more unhelpfull,unknowledgable and rude group of
people you could never wish to do business with.
Mark
Wanta take a guess what would happen if Cisco decide to really enforce
the legal rules?
I'll bite:
Their market share would plummet in all their markets, and then smaller,
more innovative companies would become more able to compete with them,
and the overall marketplace would be
having problems with snom phone installstion
Please tell us what's up. I recently installed several SNOM phones and
worked through many minor issues. Let me know and I'll tell you what I can
:)
--Ernest
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[EMAIL
In my
setup, When asterisk receives a SIP INVITE request, the request URI in my case
is [EMAIL PROTECTED]
. The SIP INVITE PDU's message header also contains a To: field. In
my case the To: field is [EMAIL PROTECTED]
. It seems that asterisk "accepts" the request-URI number as
the called
MIS wrote:
Can anyone tell me the field definitions for the queue_log file in the
Asterisk log directory?
1080593958|1080593892.0|salesq|NONE|ABANDON|1|1|50
fprintf(qlog, %ld|%s|%s|%s|%s|, (long)time(NULL), callid,
queuename, agent, event);
___
Before I go off and post a feature request on the bug tracker, I want to
make sure I've not misgoogled or miswikkid and not found an existing
capability.
What I'm looking for is the ability to determine whether or not a queue has
any queue handlers (active agents), and if it does not, bypass
Interalab Sales wrote:
Could you have asterisk running and not allowing you to overwrite while
trying to install? Do you have root rights to create files in the
asterisk folders?
Well, one of the first things I found was that nothing at all worked
unless I was root, so I've done the entire
How did the launch meeting go?
rt
On Mar 29, 2004, at 1:36 PM, Steven M. Sokol wrote:
The VON show has started off with a number of interesting
announcements.
First among these is a big announcement from Pingtel that they have
created
a not-for-profit corporation called SIPFoundry. This new
On Tuesday 30 March 2004 12:34, Terence Parker wrote:
Wanta take a guess what would happen if Cisco decide to really enforce
the legal rules?
I'll bite:
Their market share would plummet in all their markets, and then smaller,
more innovative companies would become more able to
Making use of Asterisk´s resources I can see that when 2 connections between
users is active, this activity
generates a huge ammount of packages on server interface where Asterisk is
running. So, I can see that Asterisk controls the calls system usage.
Is there a way to set up Asterisk to avoid
Hi all,
I compiled/installed chan_capi.so without problems. When I launch
Asterisk, I get the following error:
---
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
A SIP call comes into the * server on a number that I
want to immediately sack to vovoicemail. How would
this be achieved
Kurt
__
Do you Yahoo!?
Yahoo! Finance Tax Center - File online. File on time.
http://taxes.yahoo.com/filing.html
Actually, ignore that... forgot to take the check the calendar pill
this AM. Doh!
rt
On Mar 30, 2004, at 11:46 AM, Ryan Thrash wrote:
How did the launch meeting go?
rt
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When my snom200 receives an inbound SIP external sip call,
it somehow rejects the call and with a busy tone. The debug shows the
following error:
channel.c:1142 ast_read: Exception flag set on
'SIP/sipphone-7796', but no exception handler
what does this mean and how can I debug it
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jon Lawrence
Sent: Tuesday, March 30, 2004 12:50 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
...
I have no problem with the idea of paying cisco for software
On Tue, 30 Mar 2004, John Chambers wrote:
Another worrying thing that I've noticed: The stuff at the start
of the make (that scrolls off the top too fast to read ;-) first does
a mkdep, and then these messages appear:
cli.c:31:19: build.h: No such file or directory
dlfcn.c:40:25:
Yep, that would be my guess
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Tuesday, 30 March 2004 6:47 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
Rich Adamson wrote:
Wanta take a guess what
Hi,
Is Wildcard TDM400P capable of sending DTMF callerid?
Does asterisk support it?
I know X100P does not, but I have found no info as to TDM400P...
/Stig
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On Mon, 29 Mar 2004, Eric Wieling wrote:
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the
Today has been the day for satellite questions, apparently, so I'll
proxy one out to the rest of the community... I asked this
tangentially a month or two ago, but I'll put it in a more blunt way:
If you have IAX2 trunking mode experience over satellite, please let
us know your experiences
Sorry I forgot the subject in the last
post.
When my snom200 receives an inbound SIP
external sip call, it somehow rejects the call and with a busy tone. The
debug shows the following error:
channel.c:1142 ast_read: Exception flag
set on 'SIP/sipphone-7796', but no exception handler
This is not all bad. It is good research into the strategies the
big players are using. It looks like the big money players (ATT, Nortel,
Siemens, Cisco, etc.) are really trying to push into VoIP in a big way. The
other big positive is the fact that people are actually, well,
Hi,
there are something that is using mysql instead postresql?
If I modify the modules.conf, and write load = cdr_mysql.so get this error:
loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/cdr_mysql.so:
cannot open shared object file: No such file or directory
loader.c:359 load_modules:
On Tuesday 30 March 2004 19:01, Brian Cuthie wrote:
My beef with Cisco is that the software license doesn't travel with the
device. Without the license you can't buy an upgrade even if you want to.
Indeed that bit is a complete joke. I can't think of anything that could be
done about it
Title: is Asterisk capable for SIP-H323 translation?
Hi
is Asterisk capable for SIP-H323 translation? Any manual how to do this?
Thank you
Konstantin
What I'm looking for is the ability to determine whether or not a queue
has
any queue handlers (active agents), and if it does not, bypass sending
the
caller to the queue and pass them on to a message or IVR system.
-Chris
http://bugs.digium.com/bug_view_page.php?bug_id=214
This is
it is not included with the asterisk distribution. you must download it
separately. asterisk_addons.
-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 2:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] mysql or postgresql?
Hi Martin
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
Hi,
When i run the asterisk with my FXO x100p and configure:
vi /etc/zaptel.conf
fxoks=1
loadzone=us
defaultzone=us
# vi /etc/asterisk/zapata.conf
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
I haven't seen anything about this is the archives, so here we go.
Sorry, its a long one.
My setup:
Dual Xeon 2.4 GHz.
One TE410P card.
Span 1 populated with a PRI.
Span's 2-4 empty.
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
I'm currently having a
vozip wrote:
group=1
signalling=fxo_ks
mailbox=2468
callerid=Phone 1 2468
channel=1
ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
ANY IDEAS.!
CHEERS.!
VOZIP
The error
James Golovich wrote:
The mkdep simply builds .depend files in each directory of the source
tree. make uses this to determine what needs to be rebuilt if one of the
header files has changed. There is nothing to worry about at all with
that part.
OK; I'll ignore it. It can be confusing when a
The answer is in the error use FXS signalling. replace fxo_ks with
fxs_ks.
Sean
-Original Message-
From: vozip [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 2:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problem with configuration.
Importance: High
Hi,
On Mon, 2004-03-29 at 20:25, Steven M. Sokol wrote:
Not currently. There is a bounty for the development of working Wildcard
drivers for Free/Net/Open BSD. Care to write them?
On Mon, 2004-03-29 at 20:33, James Moran wrote:
Dam wish I was that good to do that.
You can pitch into the
The firebox has the UDP timeout set pretty low by default, this is a good thing
to help prevent DOS attacks, but isn't a really good thing for a SIP device.
There is no option in the GUI to set this.
However you can go into the config file itself and modify the following:
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 Forced.
When I 'make clean and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
I'm curious as to what a Washington State free phone number is? I live
in Washington State(Spokane) and we get our PSTN service from Qwest which
is certainly not free!
The poster was probably referring to IPKall (http://www.ipkall.com/).
IPKall will assign you a Washington State (USA)
On Tue, 30 Mar 2004, Kevin Walsh wrote:
I'm curious as to what a Washington State free phone number is? I live
in Washington State(Spokane) and we get our PSTN service from Qwest which
is certainly not free!
The poster was probably referring to IPKall (http://www.ipkall.com/).
IPKall
What are reciprocal comp minutes? Please explain.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
Sent: Tuesday, March 30, 2004 1:12 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] asterisk @ home ?
Must be a CLEC trying to build up
On Tue, 30 Mar 2004, calvis wrote:
What are reciprocal comp minutes? Please explain.
In some states, the competitive and incumbent phone carriers bill each
other for calls that they terminate from the other. If a Qwest customer
calls a Dave's Phone Company customer, I will get a small amount
Hello, and thank you for your time answering my question.
I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to upload
SIP configurations via *.cnf file from my tftp server, do I need to
include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root
directory to be uploaded
Hello,
I am working on making chan_mgcp work properly with IP10S from
SwissVoice. I have patched chan_mgcp from asterisk 0.7.2 and it seems to
work pretty well but not in all cases. I give the patch with this
message and wait for all feedback.
Best regards,
Daniel ANDRE
--
Daniel ANDRE
Hi,
Thanks for the help. You were correct. There was some data missing in the
extension.conf file
I was able to call one SIP phone from the other. I was even able to call an
H323 IP phone registered to the gnugk GK (It has Asterisk registered to him
as a GW).
But, I have another problem rigth
What should my allow= line look like in h323.conf for G.729?
I've tried
allow=G729A
but this doesn't seem to be right. These codec indentifiers sure are
mysterious. Take g711alaw. To allow it you seem to have to use allow=ALAW.
Even though ALAW does not show anywhere as an identifier when you
Has Asterisk ever been audited for common security holes, such as buffer
overruns?
A quick grep through the source for routines that should never be used,
like strcpy, strcat, etc., reveals a lot of it. I fear I fear.
Has anyone flung pathology at IAX2 to see if it stands up to malformed
On Tue, 2004-03-30 at 07:34, Gary Franczyk wrote:
Hello,
We are trying to deploy a new asterisk server with a Wildcard T400P (quad
T1) card. It uses a custom voice recording app written in the perl AGI.
Now that the machine has been in production, it seems to lock up within 24
hours of
have you installed the mysql-devel package?
-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 6:11 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: mysql or postgresql?
thanks for awnser, I've already download from CVS the
NOTICE[1125329600]: chan_sip.c:5609 handle_request: Registration from
'sip:[EMAIL PROTECTED]' failed for '192.168.1.100'
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Hello all.
Has anyone managed to get SIPCALL.co.uk's service working with the [*] box?
I've managed to register with other SIP providers but not SIPcall.
The debug just show's [*] attempting to register.
But receiving a 401 error everytime.
Cheers
Matt
Sip read:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:12250
From: sip:[EMAIL PROTECTED];tag=1e263406-3e84-45fb-a971-6f08bf684275
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Contact: sip:192.168.1.100:12250;methods=INVITE, MESSAGE, INFO,
SUBSCRIBE,
Depends if what you have in OS79XX.TXT is different then
what is running on your phone. If it isn't it won't bother to touch
the image files, if it is then it will try to load whatever image
you have specified in OS79XX.TXT.
So far I have been unable to tell the phones to boot out of anything
but
What version of asterisk are you using, and what version of
the SNOM firmware?
--Ernest
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
jcSent: Tuesday, March 30, 2004 10:20 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Exception
flag set - snom200
FYI,
Follow the Quick Start Guide from Carrier Access to setup
the CMG (Customer Media Gateway) Router card. Follow the Asterisk
mgcp.conf wiki page setup. The only issue I had was with the CAC CMG
card, it defaults to strict policy message exchange and dial-tone will not come
across
Title: RE: [Asterisk-Users] SoftFAX/spandsp - txfax
Hi Steve and all,
1. Faxing from asterisk back to the same asterisk (from one Zap channel to another)
doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the
receiving side needs CNG in order to switch to fax
I have recently noticed that the Action: Originate options in asterisk
1.0 CVS has changed sometime between 2/23 and 3/18.
I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk
) that allows me to make calls like this using the Manager Interface on
port 5038.
Has
anyone had any luck using a 7910 with SIP image.
Some
information I found says 7910 is skinny only, other info suggests the 7910 may
take the 7960 sip image.
Can
anyone offer their experience ?
Cheers
Peter
Search the archives.
On Tue, 2004-03-30 at 19:00, Peter Mitchell wrote:
Has anyone had any luck using a 7910 with SIP image.
Some information I found says 7910 is skinny only, other info suggests
the 7910 may take the 7960 sip image.
Can anyone offer their experience ?
Oliver Kaven wrote:
Hello, and thank you for your time answering my question.
I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to
upload
SIP configurations via *.cnf file from my tftp server, do I need to
include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root
I figured it out This works -
exten = s,1,Wait,1 ;
Wait a second before answering.
exten = s,2,Answer
exten =
s,3,SetVar,loopCnt=0
exten = s,4,Background(welcome)
exten =
s,5,SetVar,loopCnt=$[${loopCnt} + 1]
exten = s,6,gotoif,$[${loopCnt}
= 3]?s|7:s|9
exten =
Title: Message
You
could use the t extension to accomplish this. But if you're happy with
your way... :-)
Sean
-Original Message-From: Gene Kochanowsky
[mailto:[EMAIL PROTECTED] Sent: Tuesday, March 30, 2004
8:53 PMTo: [EMAIL PROTECTED]Subject: RE:
[Asterisk-Users]
Title: Message
How would you use the t extension to
accomplish this?
Gene
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
Sent: Tuesday, March 30, 2004 9:03
PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Caller
entered digits ignored
That's what the 'silence' files were invented for.
See loligo.com (forgot the exact reference, but do a wiki
for J Todd's sound files).
Yes, it's a hack, but it works.
Cheers,
Willy
- Original Message Follows -
Greetings,
Below is part of the contents of my extensions.conf file.
I have posted before but didn't get any replies so i'll ask again in a
more simple way :
Does H323 work on asterisk out of the box? I notice there is already a
channels/chan_h323.c file, but creating an h323.conf file I can't seem
to get H323 working.
Do I have to compile an additional
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