On Fri, 2004-01-09 at 21:36, Steve Underwood wrote:
> WipeOut wrote:
>
> > Granted five 9's is never easy but in a cluster of 10+ servers the
> > system should survive just about anything short of an act of God..
>
> You do realise that is a real dumb statement, don't you? :-)
>
> A cluster of
No, but the Caller ID Information for a SIP extension is stored in
sip.conf, so yes, I did think about that.
As far as making sense, many meridian systems do this, and it is quite
helpful. This could help with the implementation of gastman, and also
end user phones. On the Cisco's and Polycom's,
Hi Jan,
>the last 2 days i was working on getting the 7920 Phones to work with
>Skinny & Asterisk; however no luck (yet).
Same here, no joy so far.
>Does anybody has a SEPDefault.CNF.xml and a SEP.CNF.xml handy for
>me ? it should be documented at the cisco page, but it isn't :-(
The default fi
On Fri, 2004-01-09 at 22:40, Brent Franks wrote:
> Does * support Called Party Identification? Say for example, I dial
> extension 2000, SIP sends back John Doe from the sip.conf file where
> extension 2000 is defined? Would this violate the SIP RFC?
Maybe you didn't think about the fact that ex
As I've said several times on this list[insert usual apology here], I still
haven't received the last 20 of 100 phones I ordered over 2 months ago. If
you get a hold of them please let me know
MATT---
-Original Message-
From: mikeu [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 10, 2
Anyone else having problems getting product from Chagres? They took my
payment almost two months ago and I still have not seen hardware. They have
been horribly unresponsive to my e-mails.
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On Friday 09 January 2004 11:49 pm, Paul Liew wrote:
>
> - Original Message -
> From: "Martin" <[EMAIL PROTECTED]>
> > > mkdir -p /sbin
> > > install -m 755 ztcfg /sbin
> > > make: install: Command not found
> > > make: *** [install] Error 127
> > > [EMAIL PROTECTED] zaptel]#
> >
> > Why
Ooops, It was a type that is how I tried it. But Asterisk do not see to
have a way to force codec negotiation if possible. * always want to
tran$code. (Which cost $$ and quality)
My original config should be corrected to
Case 1
--
[sip-a]
disallow=all
allow=g729
allow=alaw
SamW
At 01
- Original Message -
From: "Martin" <[EMAIL PROTECTED]>
> > mkdir -p /sbin
> > install -m 755 ztcfg /sbin
> > make: install: Command not found
> > make: *** [install] Error 127
> > [EMAIL PROTECTED] zaptel]#
>
> Why won't zaptel make install ?
>
> Martin
Martin,
Looks like somewhere alo
Does * support Called Party Identification? Say for example, I dial
extension 2000, SIP sends back John Doe from the sip.conf file where
extension 2000 is defined? Would this violate the SIP RFC?
- Brent
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Dear All,I have had
a problem that I have posted before, the asterisk kept crashingon me. I have
thought that may be before the problem is resolved, I couldtry to implement
a cronjob to run /usr/sbin/safe_asterisk, and if Asteriskis not running at
that time, it will start it automatically.
Dear All,
I recently came across
DynEXTENdb, a way to be able to include thousands of Extensions (routes). In my
application which is VOIP, we need to include more than 50,000 area codes due to
the USA LATA routing, and there is simply no way to do that with
extensions.conf. The way DynEX
On Friday 09 January 2004 07:02 pm, Martin wrote:
> I just rebuilt it and watched this time. What are the ? about ?
>
> [EMAIL PROTECTED] src]# cvs checkout zaptel libpri asterisk
> ? libpri/libpri.so.1.0
> ? libpri/pri.lo
> ? libpri/prisched.lo
> ? libpri/q921.lo
> ? libpri/q931.lo
> ? asterisk/
On Fri, 2004-01-09 at 20:55, Ken Alker wrote:
> Does * have the capability to screen calls? IOW, if someone calls in from
> outside (ie. not a local extension), can * ask the calling party to state
> their name, record it, ring the recipient, play the caller's name for the
> recipient, then giv
Have a look at http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
If you sip client is behind firewall you will not be able to connect
to FWD. However you can get around by using IAXTEL. check out this
page:
www.iaxtel.com/setup.html
David Kwok
Thanks for that.
Actually, my machine has
WipeOut wrote:
Granted five 9's is never easy but in a cluster of 10+ servers the
system should survive just about anything short of an act of God..
You do realise that is a real dumb statement, don't you? :-)
A cluster of 10 machines, each on a different site. Guarantees from the
power company
i worked with Mediatrix boxes before (in '01-'02). configuring them using
the ume (unit manager express) is highly suggested, as there are
"parameters" in there that you won't see in its telnet access... :)
generally it works fine, depending on how you use them, or how strict are
you in evaluatin
On Friday 09 January 2004 18:45, Mindworks Wireless wrote:
> Hello,
>
> I have searched the lists, and have seen this message posted numerous
> times, however there never seem to be any replies to it.
>
> I was curious if anyone had figured out the WARNING:
>
> File chan_iax2.c, Line 5466 (set_conf
Does * have the capability to screen calls? IOW, if someone calls in from
outside (ie. not a local extension), can * ask the calling party to state
their name, record it, ring the recipient, play the caller's name for the
recipient, then give the recipient the choice of answering or forcing the
Nicolas was good enough to add it to the bugtracker as a "do not add
this feature" feature. I think this is an often-requested item, so
if someone wants to take a swing at cleaning it up that would be
great.
http://bugs.digium.com/bug_view_page.php?bug_id=773
JT
Nicolas,
I'd appreciate
TeleSIP wrote:
Don't know if this has been addressed, but why isn't there a
file_include style directive for extensions.conf?
there is...search the archives or the wikiits something like #include
filename.conf
Oh yeah, it works, thanks ..
Not entirely obvious, I guess .. I thought it would ha
We are new in Asterisk - I was wondering if someone can recommend a good
phone sets to use with Asterisk in office environment. We need about 20
sets.
Also - What can we use for the receptionist phone?
Thanks,
Aram Ter-Martirosyan
As version 0.7.0 will arrive on monday and it would be nice if it had a
nice startup script in it, please post all your startup scripts on:
http://bugs.digium.com/bug_view_page.php?bug_id=312
If you are using alternate safe_asterisk scripts or asterisk monitoring
scripts/tools, please mai
Title: Message
Hello all. Has
anyone had any success using ChanIsAvail with only SIP channels? Is there
another, better way to check if an extension is busy without dialing
it?
Thanks,
B.
J.
> Don't know if this has been addressed, but why isn't there a
> file_include style directive for extensions.conf?
there is...search the archives or the wikiits something like #include
filename.conf
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There is...
#include filename
-Original Message-
From: Lion Templin [mailto:[EMAIL PROTECTED]
Sent: Fri 1/9/2004 7:14 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] file_inlcude .. why not?
Don't know
Hello,
I have searched the lists, and have seen this message posted numerous
times, however there never seem to be any replies to it.
I was curious if anyone had figured out the WARNING:
File chan_iax2.c, Line 5466 (set_config): Ignoring port for now
Everything seems to work just fine, however
Nicolas,
I'd appreciate a copy of this if possible... got a url where I can grab it?
Thanks
Andy
*** REPLY SEPARATOR ***
On 09/01/2004 at 10:43 Nicolas Gudino wrote:
>> Andy Powell wrote:
>>
>> >I'd be nice to be able to play a tone (or message) at AbsoluteTimeout -
>N
>where
Don't know if this has been addressed, but why isn't there a
file_include style directive for extensions.conf?
I find that my extensions.conf grows a lot, and it would be a lot nicer
to have a tree of files rather than one big file to try and navigate.
Also, I've got a couple different 'systems
http://edition.cnn.com/2004/TECH/internet/01/09/telecoms.powell.reut/index.h
tml
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We appreciate the efforts, but I think the majority of the folks here
would like to see the discussions kept centralized on the Digium
mailing lists unless it becomes completely intolerable (which we're
not near yet.)
JT
At 10:41 AM -0700 1/9/04, David Burr wrote:
What about something like htt
I just rebuilt it and watched this time. What are the ? about ?
[EMAIL PROTECTED] src]# cvs checkout zaptel libpri asterisk
? libpri/libpri.so.1.0
? libpri/pri.lo
? libpri/prisched.lo
? libpri/q921.lo
? libpri/q931.lo
? asterisk/doc/api
cvs server: Updating zaptel
cvs server: Updating libpri
cvs
Hello.
Just tried it.
# cd zaptel
# make clean ; make install
mkdir -p /sbin
install -m 755 ztcfg /sbin
make: install: Command not found
make: *** [install] Error 127
[EMAIL PROTECTED] zaptel]#
--
intoxicated, adj.:
When you feel sophisticated without being able to pronounce it.
Hi,
is it possible to use Asteriks for translating SIP to H323 and vice versa?
I am looking to implement the following Setup
SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC
Basicly i want SIP fones to talk to H323 fones and and SIP Fones to
access PSTN Gateway(s) in a H323
Check this out. I recently closed a bug I had written, #495
"ExtraChannel in transfer causes crash" Now I've been able to reproduce
it, and somewhat narrowed down the culprit. But before I write another
bug report, I wanted to see if anyone else had experienced the following
(or would like t
On Fri, 2004-01-09 at 13:31, john lawler wrote:
> I've got a couple of different situations where I'd like to do something
> like zapbarge into a specific channel but I'd like to be able to
> actually talk to the party or parties on the channels, not just listen
> like w/ zapbarge.
>
> There
Here are just a few examles for clarification.
Lets assume my local area code is (212)
--- local calls - usually free --
555-= local call 7-digit dialing area
212 555-= local call 10-digit dialing area
toll calls - metered ---
Yeh maybe I sold my sole to the devil a few times... but Im not telemarketing. Im calling back coustomers Ive hade in the past and doing polls in 1 business, and the other is student lone consolidation.
>From: [EMAIL PROTECTED]
>Reply-To: [EMAIL PROTECTED]
>To: [EMAIL PROTECTED]
>Subject: Aster
The FTC ratio is what the FTC allows for droped calls, say the dialer is calling 10 phone #'s per agent connected. As for the ip phones are they needed? or can I just use regular phones with headsets? and as to the 2 week setup I know it will take alot longer but I have a temp dialer until I can g
Since my earlier inquiry about gateways went unanswered perhaps
rephrasing will help. Does anyone here have experience with standalone
SIP FXO gateways like those from Mediatrix? Care to share their
experiences with them? Off list if necessary.
Michael
--
Michael Graves
I can smile now. I made my * work with my
Cisco. Finally. First problem was Ethernet on my Linux. After
installing * on a different machine, I got rid of that "icmp udp unreachable"
error. My next problem was the call stays on on Cisco gateway, but the ATA
drops it. I figured out it was
Testing between ATA and Asterisk is working fine. I am getting voicemail
etc. But when I'm trying to call to the "carrier side" i find it not
working. I see on my Cisco gateway that it negotiated the g711ulaw codec,
but when the "state" goes into active, I just automatically get busy tone
from my
Paul Liew wrote:
- Original Message -
From: "Michael" <[EMAIL PROTECTED]>
Sent: Thursday, January 08, 2004 10:40 PM
if you are on a call on the Budgetone 101 and a 2nd call is received,
instead
of a call waiting beep being played, it rings on the handset speaker!
which
makes it almost
Arnold,
Sorry!
I was expecting my email address to be on the header of my message!!
(I'm on the digest)
If anybody is interested then please contact me off-list:
[EMAIL PROTECTED]
Sorry again,
Ad.
On 9 Jan 2004, at 8:44 pm, [EMAIL PROTECTED]
wrote:
Message: 4
From: "Arnold Ligtvoet" <[E
>Yes, in most places in the USA local
calls are totally free, no per min>charge.
This is not true in the US for business
lines. Residential lines have a "free" local calling area.
However, business lines from an incumbent local exchange carrier like SBC
nearly always charge rates for 7-dig
>
> Yes - the Wiki link above about "call queues" has the info and links that
> you need to look at.
Also, could be great is you install a new patch, to add some great
functionalities to your call center. This path is located:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
On Thursday 08 January 2004 03:22 am, Dave Cotton wrote:
> On Thu, 2004-01-08 at 07:42, Steve wrote:
> > Hi,
> >
> > I used to have a Grandstream phone connected to Asterisk a few months
> > ago. Worked just great!
> >
> > Then today I do a new install, rather than an upgrade, and all of a
> > sudd
On Thursday 08 January 2004 12:03 pm, Dave Cotton wrote:
> On Thu, 2004-01-08 at 17:28, Steve wrote:
> > On Thursday 08 January 2004 03:22 am, Dave Cotton wrote:
> > > I just downloaded my mail to start the day, SIPphone had emailed me
> > > with a firmware update for GS, having had exactly the pro
On Friday 09 January 2004 02:40 pm, Iain Stevenson wrote:
> Prices? Are we talking a 7960 for the price of a SNOM?
>
> Iain
Oops, just realized I replied to the wrong person
> --On Friday, January 9, 2004 6:00 pm + Adthrawn
>
> <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I know it's not
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of empire
underground
Sent: Friday, January 09, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Screen Pop & Remote Agents
> can I put a .csv file in the sql DB and have it dial from there? and
will
the * and # are hard coded.
unless "b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
is what your refering to.. which doesnt say how to use it.
Jeremy McNamara wrote:
David Burr wrote:
Does the MeetMe monitor for DTMF tones to trigger an AGI?
If not is this a planned feature?
show app
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Philipp von Klitzing
> Sent: Friday, January 09, 2004 6:52 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Mailing list growth
>
[...]
> "higher-level implementation" list that deals spec
Thilo Salmon wrote:
Hi everyone,
any chance * could be used as a b2bua without forcing the media stream
through the same box? I would love to do some computing on incoming
calls, do things like setting another callerid and the forward the call
to another sip UA - all without any audio traversing
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
>
>
> I have not been to USA (yet) :)
>
> Ta
> SJ
For comprehensive info by area code (and as pointed out it does
On Fri, 2004-01-09 at 14:20, Arnold Ligtvoet wrote:
> message posted on behalf Of Adthrawn
> [SNIP cisco stuff]
> > I'll now feel ashamed, and sink into my seat :-)
> >
> > Best,
> > Ad.
>
> Perhaps it would have been better to provide an email address or phonenumber
> where people can contact you
> case 1
> --
> [sip-a]
> allow=g729
> disallow=all
> allow=alaw
Try:
[sip-a]
disallow=all
allow=g729
allow=alaw
The "disallow=all" clears your previous setting of "allow=g729"
--
END OF LINE
-MCP
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Information on the way things are structured here can be gleaned by
Googling for 'North American Numbering Plan'. Way too much information can
be found at http://www.nanpa.com/
k.
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL PROTECTED]
Sent: January 9, 2004 10:50 AM
To: [EMAI
On Friday 09 January 2004 13:55, [EMAIL PROTECTED] wrote:
> > Hi,
> >
> > Do the callers in USA dialling from USA Telco lines always have
> > to prefix the CITY/AREA code with "1" in order
> > To successfully make a call to other USA destinations?
> >
> >
> > I have not been to USA (yet) :)
>
--__--__--
Message: 1
From: Terence Parker <[EMAIL PROTECTED]>
Date: Fri, 9 Jan 2004 11:25:23 +0800
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem registering FWD
Reply-To: [EMAIL PROTECTED]
--Apple-Mail-1-822243116
Content-Transfer-Encoding: 7bit
Content-Type: text/plain;
charset
Stephen J. Wilcox wrote:
You are trying to have both ends act as users, cisco can support emulating a
network interface (isdn protocol-emulate in serial interface config) but in my
experience i could get the circuit up but it would bounce and i couldnt get
signalling to work.. to be fai
Hi everyone,
any chance * could be used as a b2bua without forcing the media stream
through the same box? I would love to do some computing on incoming
calls, do things like setting another callerid and the forward the call
to another sip UA - all without any audio traversing the * box. Any
ideas?
Some areas in the US also use 10 or 11 digital dialing for all calls,
whether they are local, long, toll or non-toll.
> -Original Message-
> From: Eric Wieling [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 09, 2004 1:53 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] USA dia
message posted on behalf Of Adthrawn
[SNIP cisco stuff]
> I'll now feel ashamed, and sink into my seat :-)
>
> Best,
> Ad.
Perhaps it would have been better to provide an email address or phonenumber
where people can contact you directly. Now everybody who is interested has
to reply to the list.
I always like to use * to negotiate a codec which both sip clients
support. But * do not try to go in that direction, * try to be in the
middle and try to translate(code convert) the rtp stream, which will
deteriorate the call quality and it will also cost for license if we are
translating using G
Just a little clarification on USA local calling:
Local calls are generally free for residential customers, unless they are on
a increasingly rare "measured local service". However, business customers
almost always pay for local calls on a measured basis.
Regards
Scott
Scott M. Stingel
Emergin
On Fri, 09 Jan 2004 13:40:45 -0600, Steven Critchfield wrote:
>On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote:
>> Hi,
>>
>> Do the callers in USA dialling from USA Telco lines always have to
>> prefix the CITY/AREA code with "1" in order
>> To successfully make a call to other USA destinati
--- Senad Jordanovic <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
Not "always". My local phone company (Verizon in So. California)
s
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
>
>
> I have not been to USA (yet) :)
>
> Ta
> SJ
In all cases of long distance, 1 plus the area code is used
Hi!
> Im new to Asterisk and I would like to get some imput from all of you.
> First I would like to start by telling all of you I am starting a call
> center Within the next 2 weeks with 12 agents to start.
Asterisk is probably a very good choice for you, however you'll need more
time than th
Generally speaking, Yes. The usual dial plan in the USA is as follows:
NXX- (Free Local Call to number in same Area Code)
NXX-NXX- (Free Local Call to number in different Area Code)
1-NXX- (Toll Call to number in same Area Code)
1-NXX-NXX- (Toll Call to number in different Area Cod
Has anyone implemented a soft-fax within *? If not, is there a SIP or IAX
or IAX2 client for * that would function as a receive-only fax device?
Thanks!
--Mike
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On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote:
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
1 usually signifies a long distance call. It also is prepended
Prices? Are we talking a 7960 for the price of a SNOM?
Iain
--On Friday, January 9, 2004 6:00 pm + Adthrawn
<[EMAIL PROTECTED]> wrote:
Hi,
I know it's not really the place, but if anybody in the UK (or US) is
interested, I'm clearing out lots of new Cisco stock...
7970G's (colour LCD),
I've got a couple of different situations where I'd like to do something
like zapbarge into a specific channel but I'd like to be able to
actually talk to the party or parties on the channels, not just listen
like w/ zapbarge.
There are two scenarios I can think of right now where it'd be very
Sorry for the reply to my own post.
Aparently Gary has submited the fix to the digium bug list. If you haven't
already, please check it out. =)
Thanks,
--
Sean Swallow
On Fri, 9 Jan 2004, Sean Swallow wrote:
> Thanks Mark... We've hacked it here to work for now, but look forward to
> your C
On Fri, 2004-01-09 at 11:42, john lawler wrote:
> Hi guys,
>
> I've had a sporadic problem recently with one of my users on our POTS
> line. About 1/3 of the time he dials a number (usually from a speeddial
> on his phone, I think), he'll get some phone company message (from the
> outside) abo
At 19:31:16, Tilghman Lesher wrote:
TL> On Friday 09 January 2004 11:23, Robin Calmegård Siurua wrote:
>> I have some problems when trying to install Asterisk on Mandrake.
>>
>> gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o
>> `gtk-config --libs gthread`
>> /usr/lib/gcc-lib/i486-sla
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with "1" in order
To successfully make a call to other USA destinations?
I have not been to USA (yet) :)
Ta
SJ
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[EMA
Hello,
I've subscribbed to "IConnect". I use it eclusively for outbound
calling. I like the rates they charge but people I call complain about
the audio quality. They say it sounds like I'm using a "cheap mic." or
they
complain about echo. The sound is very clean at my end. I'm using
a Bund
>
> Late night. I've been to http://www.grandstream.com/TEMP/FIRMWARE/ I just
would like to find 1.0.4.17 so I know I'm not introducing any new bugs if I
have to go back. I meant to say if you know somewhere else to get 1.0.4.38.
I also tried just downloading it from my grandstream but it didn't
Mark Spencer wrote:
I still think we need something more fine grained. I think we can add the
asterisk-biz list, and eventually something akin to a newbie list, but
need a more appropriate name, IMHO.
like an asterisk-virgin
* for the very first time
Now lets see how long it will take you to get
Hi Everyone,
Im new to Asterisk and I would like to get some imput from all of you.First I would like to start by telling all of you I am starting a call center Within the next 2 weeks with 12 agents to start. I have also been looking at alot of dialers! The prices are outragous! So a friend of
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 09, 2004 11:10 AM
Subject: RE: [Asterisk-Users] SIP and error talking to voicemail
>
> > Original Message
> > Subject: RE: [Asterisk-Users] SIP and error talking to voicemail
>
On Friday 09 January 2004 11:23, Robin Calmegård Siurua wrote:
> I have some problems when trying to install Asterisk on Mandrake.
>
> gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o
> `gtk-config --libs gthread`
> /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackw
>a
David Burr wrote:
Does the MeetMe monitor for DTMF tones to trigger an AGI?
If not is this a planned feature?
show application MeetMe
Jeremy McNamara
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Here's my sip debug output. anybody knows
why Cisco sent * is CANCEL msg? Can someone tell me what ATA version
are they using? Maybe this is also another issue.. I am using
v2.16.
This is using G711ulaw.
Sip read: > SIP/2.0 100 TryingVia:
SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fb
TC wrote:
>> Yes, it does work behind NAT quite well. (at least it does for me
>> with Draytek Vigor 2600 router)
> Can you clarify one point, do you do any redirection on that router I
> know most ppl redirect the SIP TCP control port and a say 10 UDP rtp
> ports on the NAT device then on the SIPu
Hi,
I know it's not really the place, but if anybody in the UK (or US) is
interested, I'm clearing out lots of new Cisco stock...
7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone),
7935's (conference phone) and 3550-24-PWR switches.
I also have boxes of 7914's, the single-791
Hi,
the last 2 days i was working on getting the 7920 Phones to work with
Skinny & Asterisk; however no luck (yet).
Does anybody has a SEPDefault.CNF.xml and a SEP.CNF.xml handy for
me ? it should be documented at the cisco page, but it isn't :-(
I still have the issue that the 7920 spits out "N
On Friday 09 January 2004 09:56, WipeOut wrote:
> massimo wrote:
> > Hi, I would like to log incoming and outgoing call in the file
> > master.csv. I have to do something in the extension or in the
> > conf file because at the moment * does not log nothing
>
> It should automatically log all calls.
Hi guys,
I've had a sporadic problem recently with one of my users on our POTS
line. About 1/3 of the time he dials a number (usually from a speeddial
on his phone, I think), he'll get some phone company message (from the
outside) about how the call could not be completed as dialed or
somethi
What about something like http://www.asterisk.bz
Philipp von Klitzing wrote:
Hi!
I still think we need something more fine grained. I think we can add
the asterisk-biz list, and eventually something akin to a newbie list,
but need a more appropriate name, IMHO.
As discussed before that
On Fri, 2004-01-09 at 11:12, WipeOut wrote:
> I understand.. Thanks for explaining..
>
> I am going to give kernel building a go but I don't think it will be on
> a production box for a while, not until I am happy that I have at least
> some understanding as to what I am doing..
Your welcome an
Does the MeetMe monitor for DTMF tones to trigger an AGI?
If not is this a planned feature?
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I have some problems when trying to install Asterisk on Mandrake.
gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs
gthread`
/usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld:
cannot find -lXext
collect2: ld returned 1 exit status
I understand.. Thanks for explaining..
I am going to give kernel building a go but I don't think it will be on
a production box for a while, not until I am happy that I have at least
some understanding as to what I am doing..
Thanks again, as always you are a great source of informed help..
La
> Original Message
> Subject: RE: [Asterisk-Users] SIP and error talking to voicemail
> From: "Dave Cotton" <[EMAIL PROTECTED]>
> Date: Fri, January 09, 2004 1:03 am
> To: "Asterisk List" <[EMAIL PROTECTED]>
>
> On Fri, 2004-01-09 at 06:37, [EMAIL PROTECTED] wrote:
>
> > How co
> On Fri, 2004-01-09 at 16:05, TC wrote:
> > > Yes, it does work behind NAT quite well. (at least it does for me with
> > > Draytek Vigor 2600 router)
> > Can you clarify one point, do you do any redirection on that router
> > I know most ppl redirect the SIP TCP control port and a say 10 UDP rtp
Thanks Mark... We've hacked it here to work for now, but look forward to
your CVS changes.
--
Sean Swallow
On Thu, 8 Jan 2004, Mark Spencer wrote:
> No, it's some side effect of the 2.6 changes... strtok goes away in 2.6,
> i'll try to fix it.
>
> Mark
>
> On Thu, 8 Jan 2004, TC wrote:
>
>
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