I usually use Qmail www.qmail.org, in my humble opinion it is more straight forward to
configure than sendmail.
On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote:
I understand asterisk invokes sendmail in order to send email
notifications of messages left. Is there another application
The problem is some calls from the PSTN have hidden caller id so if you want to change
it to something else then modify chan_sip.c
#define CALLERID_UNKNOWNAsterisk
I've changed mine to:
#define CALLERID_UNKNOWNUnknown
-Original Message-
From: Shaun Ewing
Marcello,
This is something I am hoping for as well but I cannot find any features within the
code to allow Asterisk to modify/create this field. Ideally I'd like to see the
CallingPres function support Remote-Party-ID to disable/enable privacy. I actually
placed a feature request some months
Ironic, Im just working on something similar myself, you can either use the
appropriately named ex-girlfriend feature or I use GotoIf statements to match the
caller id and maybe a timer or something to route to another context.
; note page search in girlfriend
According to IANA's list of RTP payload types
(http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls within the
following range:
72--76 reserved for RTCP conflict avoidance [RFC3550]
I can't find much else in RFC3550 that defines it further but this
Typo in your OS79XX.TXT P00 ? instead of P0S !?
-Original Message-
From: Michael Løjtnant [mailto:[EMAIL PROTECTED]
Sent: 17 August 2004 13:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960
released
Hi Shaun,
Saw you post, and rushed to their
Ok please ignore me, I just tried 7.2 myself and worked fine with the same mods you
made !?
-Original Message-
From: Michael Løjtnant [mailto:[EMAIL PROTECTED]
Sent: 17 August 2004 13:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960
released
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
* -- SIP -- CISCO -- PRI -- PSTN
The PSTN sees no callerid.
*--- PRI[zaptel]-- PSTN
Callerid is there... which makes me think it's the cisco, not the
PRI/PSTN/telco.
CISCO PRI-- * PRI [zaptel]
Callerid IS
Hi,
We've been working a lot with Asterisk in SIP for over 6 months but I've finally
succumb to the pressure of H.323. I need to find a way to do what we do with SIP but
with H.323. That is to have calls from H.323 peers placed into their own unique
context (unique to the endpoint placing the
I had a similar issue when installing my G.729 licences. I contacted Digium support
and an engineer logged into my system and performed some hocus pocus and got it
working for me ...
-Original Message-
From: Derek Samford [mailto:[EMAIL PROTECTED]
Sent: 25 March 2004 18:29
To: [EMAIL
channels
You could simply redirect both of them to a meetme room with the 'q' flag
set for no messages. I'm using that method for an application right now.
MATT---
-Original Message-
From: Low, Adam [mailto:[EMAIL PROTECTED]
Sent: Monday, March 22, 2004 12:15 PM
To: '[EMAIL PROTECTED
Hey All,
I'm developing a reception style console (like many others) to answer incoming calls
to a main line number, request who they want to speak to and then have the
receptionist call the desired party and announce the calling party before putting them
through.
This should be fairly
I tried to get that working as well and also found it was not available in the SIP
image. You can't do pushes either to the phone like you can with SCCP.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: 14 March 2004 13:27
To: [EMAIL PROTECTED]
Subject: Re:
Has anyone reported a bug for this ? if so what's the id ?
-Original Message-
From: Andrew Thompson [mailto:[EMAIL PROTECTED]
Sent: 11 March 2004 23:02
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 and short delay before voice
startsafter ring.
Steve Dolloff wrote:
We
I posted the results of my real world analysis of codec bandwidth usage on this list a
couple of weeks back. Here's the table I put together and an example of calculating
bandwidth over ADSL.
G.711 over Ethernet = 95 Kbps per channel
G.711 over IP/PPP = 86 Kbps per channel
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice
star ts after ring.
Thanks for the information. You have saved me a few hours on the phone
with TAC. smile
Low, Adam wrote:
We have a TAC case open on this issue (reference DDTS CSCed48311) as well
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently
it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's
what Cisco stated) but now we are hearing that it will not be fixed in that release
but would most likely be further down the
Reece,
I have a similar setup by the sounds of things (running 0.7.2 with AS5300) and on
private number calls what you actually get is 'Anonymous 010101010101' and as far as
I remember it was always like that for me. How are you pulling the callerid into your
script ?
-Original
I've done a fare amount of analysis on codec bandwidth requirements and you should
remember that you typically will require more bandwidth over ADSL than you would over
any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711
channel rather than 86kb over straight
So for us Dummies out here :) who just know it works.
Yep, it sure does, I thought it was something people might find interesting. Its
certainly been a challenging subject for me to try and provide reliable and high
quality voice service over ADSL. In my experience it seems to depend a hell of
I have
the same issue ...
-Original Message-From: Glenn Dalgliesh
[mailto:[EMAIL PROTECTED]Sent: 01 March 2004
16:03To: [EMAIL PROTECTED]Subject:
[Asterisk-Users] CVS login
I seem to be having trouble with cvs login.
anyone having similar problems
It just hangs
Or perhaps I should say 'adams.psknet.com' is down, box appears to be down ...
-Original Message-
From: Low, Adam
Sent: 01 March 2004 16:14
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] CVS login
It seems to me that 63.171.251.202 (adams.psknet.com) is problematic
-ID,
specificallyCLID priva cy
Low, Adam wrote:
Hey All,
I have a Cisco AS5300 running SIP against an Asterisk server with
multiple C7940 phones.
My issue is that from what I see in chan_sip.c there is no support
for
the
Remote-Party-ID field in relation to withholding the calling
Impressed. Does some countries have laws on SIP implementations? Wow. ;-)
We operate a large traditional telephone network in several countries and as I am sure
you are aware lawful intercept is a requirement on traditional networks. We've
extended our network to provide VoIP gateways
Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: 27 February 2004 12:30
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID,
sp ecifically CLID priva cy
Low, Adam wrote:
Could you please point me in direction of standard documents, drafts
I've been testing a nice little box that has precisely what you requested. Its made by
Aethra (Spain) I believe and know as the VIP3001 or VIP3002 and it runs both SIP/H323
and allows you to select if you want to send calls of the VoIP or over the PSTN. It
works great with Asterisk running SIP.
Hey All,
I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940
phones.
My issue is that from what I see in chan_sip.c there is no support for the
Remote-Party-ID field in relation to withholding the calling partys number. This is a
legal requirement for many
As Tilghman indicated X is definitely not required to build Asterisk, we run RH9
without any X related packages installed and it compiles and runs perfectly.
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: 17 February 2004 19:17
To: [EMAIL PROTECTED]
Subject: Re:
I don't think there is really an issue with 'which' analog phone, the only issue (I am
aware of) with interoperability is in relation to CallerID. In the US it seems FSCK (I
understand from my Aussie colleague that FSCK is also used in Australia) is always
used and across Europe it seems to be
Hmmm did you read any of the docs on cisco.com ?
You need to set the 'message_uri' option to the extension that you run VoiceMailMain
on into the configuration file (SIP000XXX.cnf) for the phone.
-Original Message-
From: John Fraizer
To: [EMAIL PROTECTED]
Sent: 11-2-04 6:22
Subject:
I have
a Vega 50 BRI working without any of the issues you mentioned, the dual SIP
registrations is normal for most multi-line boxes enabled split
users.
Rgds,
Adam
-Original Message-From: Glenn Dalgliesh
[mailto:[EMAIL PROTECTED]Sent: 05 February 2004
20:11To: [EMAIL
Does anyone have any documentation on Asterisk + oh323, I am trying to allow a H323
peer to send me calls that I want to push out to SIP phones but am having trouble
passing the digits dialed from the oh323 peer and dialing those digits onto a SIP
client.
Any docs much appreciated or even
res_monitor.so: Resource for
recording channels.
-Original Message-From: Rattana BIV
[mailto:[EMAIL PROTECTED]Sent: 05 February 2004 16:20To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Record
conversation
Hi,
Does anybody know if it is possible to record a
Several people have requested more information on my cluster setup, I'll try to put
something together today but things are very busy here at the moment ... but keep an
eye for a mail today ...
-Original Message-
From: David Luyens [mailto:[EMAIL PROTECTED]
Sent: 03 February 2004 07:39
Apologies for the belated reply but I've spent the weekend fighting DDoS attacks
against Superbowl sites ... )c;
Ok, well I am not sure what went wrong with previous testing but I have tried this
again with Cisco 7940's and Cisco AS5300's and indeed the RTP stream flows directly
between
I'm confronted with an issue that I am sure many others are too with Asterisk and
scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a
large volume of simultaneous calls but have the feeling that the hardware requirements
to handle large volumes of RTP streams
http://www.digium.com/index.php?menu=resellers#Europe
-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: 26 January 2004 11:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Need Europian vendor for Digium hardware.
Must accepts wire transfers and ships to Sofia.
You need a little more to make this script reboot the phone. It basically instructs
the phone to check a file called 'syncinfo.xml' at its TFTP URL. This file needs to
contain the following line:
IMAGE VERSION=* SYNC=2/
The number 2 above is the sync value which must be different (I
canreinvite=yes within sip.conf entities ...
-Original Message-
From: Al [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 2:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re-Invite between SIP phones
Anybody knows what do I need to tell Asterisk
to issue a re-INVITE
/gateways.
Please provide more information on your setup ...
-Original Message-
From: Al [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 2:52 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Re-Invite between SIP phones
Already did that, but it's not working.
Al
--- Low, Adam
All,
I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and
outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work
fine but outbound faxs receive congestion from *.
I've got packet dumps from both sides and everything appears normal but
Florian,
Sorry you haven't heard anything but we've recently decided not to offer this product
out side of Holland. If your still interested we have another product called ISDN-Flex
that provides SIP/H.323 PSTN access inbound/outbound but you need to be connected on
on one of our IP or
Hey All,
I've started to try and distribute the functionality of my single * server amongst a
few varying servers. The issue I have is that when splitting out the voicemail portion
onto a dedicated server I am no longer able to inform the voicemail application (when
call originated from a
You could add an initial digit based on whether it was a busy or no
answer forward, use the extra digit to determine the message played on
the VM server and just strip it back off to get the mailbox number.
Email me direct if that isn't clear enough.
This is actually what I have at the
Second that !
-Original Message-
From: Cees de Groot [mailto:[EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in
Paris
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backwards (C) symbol on
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
Actually we just had dinner and had left our things in his car which
(according to the police inspector) was entered through the trunk using a
half a tennis ball.
Mark
Yep
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
I have to object to that, as a rule of thumb the Dutch only rob tourists who
are dressed like tourists and act like tourists, that's what we all agreed
to here and live by --
We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway.
If you'd like to speak to an account representative please contact me personally by
email.
Rgds,
Adam
-Original Message-
From: reseaux [mailto:[EMAIL PROTECTED]
Sent: 13 November 2003 13:52
To:
Low, Adam wrote:
We can offer SIP based VoIP call termination in The Netherlands,
Austria and Norway. If you'd like to speak to an account
representative
please contact me personally by email.
Hmmm, this information should be on a website somewhere...
Your probably right
Hi *ers,
If anyone with the capability and more appropriately the time, fancies developing a
patch to provide sip debug ip_address capability with Asterisk I am sure they will
be eternally praised (c;
Rgds, Adam
* DISCLAIMER *
This message and any attachment are
Hi Bryan,
I am aware that the IETF have an Internet Draft in the pipelines for SRTP which can
provide encryption and there is a lib out there available at:
http://srtp.sourceforge.net/srtp.html
I guess the real question would be if there is any intension to include this (or an
equivelant) in
I'm not sure if it would be really practical to have a built in switch (although
useful) within the phones. You really don't want your phone worrying too much about
switching other ethernet frames whilst a call is in progress, you will probably then
run in to queueing problems as you need to
I don't have a single client that runs 10Mbps ethernet in their offices anymore and
to
tell them that the phone will downgrade their network speed to 10Mbps
puts them off the phone straight away..
Hey WipeOut,
Maybe I am missing something here but why would it downgrade their network
Well I disagree, there are numerous companies providing E1 channel banks, my personal
favourite is J-tech of which I can find the damn link to their page for now ...
Digging ...
A quick google with e1 channel banks also found:
http://www.valiantcom.com/vcl_cb/vcl_cb.html
-Original
I am guessing you are running without reinvite's, I'm running with reinvite's with
latest CVS release and 79x0 phones without any issues with conferencing...
-Original Message-
From: Adam Rothschild [mailto:[EMAIL PROTECTED]
Sent: 08 October 2003 15:49
To: [EMAIL PROTECTED]
Title: Message
All, seems I too am suffering from posts to the list and being accused of
SPAMing
-Original Message-From: AntiSpam UOL
[mailto:[EMAIL PROTECTED] Sent: 26 September 2003
20:48To: [EMAIL PROTECTED]Subject: RE:RE:
[Asterisk-Users] RTP routing..
to your mail client to delete all mail from
AntiSpam
UOL [EMAIL PROTECTED]..
Worked for me..
Low, Adam wrote:
All, seems I too am suffering from posts to the list and
being accused
of SPAMing
-Original Message-
*From:* AntiSpam UOL [mailto:[EMAIL PROTECTED
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your
IOS version and IOS config file ...
I have not specified any allow's or disallow's in my * config for the codecs with my
5300, I also use Cisco 79xx phones and I use the option within the phones config file
to
-server
dtmf-relay rtp-nte
codec g711alaw bytes 80
!
gateway
timer receive-rtcp 1000
!
sip-ua
no oli
sip-server ipv4:62.39.85.19:5060
!
!
line con 0
line aux 0
line vty 0 4
password 7 094D4210160B
login
!
end
On Mon, 2003-09-29 at 14:17, Low, Adam wrote:
I wouldn't
WipeOut,
Well will you really run out of bandwidth ?
Would that be due to other (normal Internet traffic) traffic or would it all be RTP
traffic, I ask because maybe some kind of priority queuing might be more effective ...
It's a good question, the source and destination address/port of RTP
??
Later..
Low, Adam wrote:
WipeOut,
Well will you really run out of bandwidth ?
Would that be due to other (normal Internet traffic) traffic
or would it all be RTP traffic, I ask because maybe some kind
of priority queuing might be more effective ...
It's a good question
WipeOut,
I just started to whiteboard this and had some realisations/questions:
1. I guess/hope your ADSL connection is not NAT'd ?
2. You will need two NIC's as I assume you will have two separate next hop gateways
with each ADSL connection!
3. How would you load balance the inbound calls over
port 5060 open , the SIP
registration
works nice but I didn´t receive sound...
Andre Lomonaco
-Mensagem original-
De: Low, Adam [mailto:[EMAIL PROTECTED]
Enviada em: Friday, September 26, 2003 9:06 AM
Para: '[EMAIL PROTECTED]'
Assunto: RE: [Asterisk-Users] RTP routing
Excellent news, congratulations !!
-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]
Sent: 26 September 2003 15:38
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
I just got back from Boston where we completed testing of the
Slow machine? H I think its time I invested in hardware but my PII works great !
-Original Message-
From: Peter Pauly [mailto:[EMAIL PROTECTED]
Sent: 12 September 2003 12:29
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Start of all recordings cut off
On Thu, Sep 11,
My 5 cents ...
Since the ideal situation would be real-time monitoring then maybe a more effective
solution would be to sample/duplicate the packets in the IP layer rather than
expecting Asterisk to perform yet another auxiliary function.
Cisco like most vendors are in a position were they
This looks good to me, much better than the ilogical Cisco Call Manager voicemail menu
structure ...
-Original Message-
From: Don Pobanz [mailto:[EMAIL PROTECTED]
Sent: 12 September 2003 15:21
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Voicemail menu structure
There has
I've been building a number of applications (SMS gateway, 411 directory interfaces,
blah blah) recently along the same lines, I am mostly using Perl/MySQL and of course
using the Cisco XML interface. I noticed people requesting more information on the XML
interface and so I thought I'd drop a
Sure is, you can set the accountcode=13213 within each entity of sip.conf (or iax.conf
I believe).
-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
Sent: 04 September 2003 17:10
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Question about cdr_sql fields
Maybe its just me but I find this question a little confusing, the tone duration
should have no impact on tone recognition and typically in my experience the duration
of the tone is defined by how long the user holds down the button !?
-Original Message-
From: James Sizemore
Hi All,
I've been developing all sorts of applications for use on our 79xx handsets but am
having great difficulty with formatting, I just can't seem to be able to produce a
line feed between lines on the stuff actually displayed on the phone. Has anyone else
has experience or success with
I'm not running the latest CVS release but found a couple of days ago that CDR's were
not being inserted into my MySQL tables, I restarted Asterisk and it worked fine again
...
-Original Message-
From: Tais M. Hansen [mailto:[EMAIL PROTECTED]
Sent: 18 August 2003 18:09
To: [EMAIL
I didn't get any feedback on this, I guess its nobody else has come across the
requirement maybe ?
-Original Message-
From: Low, Adam [mailto:[EMAIL PROTECTED]
Sent: 12 August 2003 12:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Malicious Call Trace
All,
Has anyone
All,
Has anyone had any thoughts/discussion on providing a malicious call trace feature
within Asterisk. Most legacy PBX's support this feature which allows a handset user to
indicate using DTMF during a call that it's a malicious call which instructs the PBX
to send a specific Q931 message
get
blocked
by asterisk, and never reach the phone.
The setup is the same : 7960 -- asterisk -- C2651-
PSTN
Yves
|-+-
| | Low, Adam |
| | [EMAIL PROTECTED]|
| | Sent
it to a 0.
}
It isn't an error, so it should just return. Change that and the
function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
Brenton, Yves, ...
I've located the cause of the problem in chan_sip.c but am
= No when they are in a NAT
environment.
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
I've been reading up on the SIP and related (SDP/RTP
1. what's the sequence to press on a SIP phone to transfer a
call to another
extension.
Which SIP phone? Soft/hard ? Phone specific ...
2. what's the same thing if you want to hold an incoming
call, speak to the
other extension, then pass the call?
Which SIP phone? Soft/hard ? Phone
: [Asterisk-Users] stupid questions ..
Sip phones on the system are Grandstream Budgettone 100's.
Was assuming it wouldn't be phone specific :)
they have flash key which is meant to send a DTMF.
thanks for the help with the dial string.
Dave
- Original Message -
From: Low
Personally, I've compiled Asterisk on Redhat and Debian without any problems on
either, I think generally Asterisk compiles very easily no matter what the distro but
I would recommend that you use the one you are most comfortable/experienced with.
-Original Message-
From: Sean Rodger
:16:16 PM
On your sip.conf for each sip endopoint set canreinvite = yes.
That way the rtp stream won t go through *. The only problem
though is for
ATA 186. They need canreinvite = No when they are in a NAT
environment.
- Original Message -
From: Low, Adam [EMAIL
I've not got a sound card in my RH9 * box and music on hold works great as long as you
have mpg123 in /usr/bin
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 27 July 2003 20:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] can't get musiconhold to
I asked the same question a couple of weeks ago and was told by Digium that its not
commercially available yet but the source code is available under NDA with Digium.
I'll dig out my contact and send off-list ...
Adam
-Original Message-
From: Marcel Prisi [mailto:[EMAIL PROTECTED]
I've previously been using G711alaw on both the AS5300 and the phones but feel the
need for a less bandwidth hungry codec for those users that are connected behind ADSL
and so was investigating G.729 but ..
Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940
Hielke John,
I too have the 7940 phones working perfectly in my setup with the exception of 7940's
that are NAT'd when SIP'ing towards the AS5300 and then I find one way voice path but
strongly suspect the mini firewall's we are using but am yet to debug this.
John has some example configs on
Hi All,
I seem to be having a problem with calls from Asterisk into the AS5300, I am sniffing
the session between the AS5300 and the Asterisk server and I see the Asterisk server
send a SIP INVITE and the AS5300 responds with a SIP 100 TRYING but then I do not see
any more SIP signalling
Greetings,
I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from
Asterisk. I have been unable to identify through the docs how specifically this should
be configured in Asterisk and have not been able to get things working through trial
and error.
I am sure I
Concepcion [mailto:[EMAIL PROTECTED]
Sent: 17 July 2003 13:40
To: [EMAIL PROTECTED]; Low, Adam
Subject: Re: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability
* DISCLAIMER *
This message and any attachment are confidential and may be privileged or otherwise
protected from
Title: Message
William,
I am running 7960/7940's with 5.1 (Asterisk SIP) without problems
although I did have some issues (too numerous to mention)with new phones
that had never been operated on a CallManager network first. It seems the
firmware must be upgraded to support SIP and this can
Not me I'm afraid, I'm running Asterisk -SIP- Cisco AS5300 -E1- PSTN .. no Quicknet
hardware for me ...
-Original Message-
From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED]
Sent: 17 July 2003 15:49
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help Needed
Thanks
James, thanks I appreciate it.
-Original Message-
From: James Golovich
To: '[EMAIL PROTECTED]'
Sent: 17/06/03 18:41
Subject: Re: [Asterisk-Users] Firewall Silly - anyone can help with a CVS tar ball ?
* DISCLAIMER *
This message and any attachment are confidential and
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