Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls. It is
hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that
these packets are not spaced out at 20ms. In general you see something like:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
I am working on a project vide over IP
I am asking you to inform me whether asterisk software PBX supports video
over IP
hubert
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Hi!
Does * supports video? Especially, SIP or IAX?
Is there any cool client for Linux and Windows that is NOT H.323?
--
WBR,
Max Tulyev (MT6561-RIPE, 2:463/[EMAIL PROTECTED])
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Hello
When I tried loading TDM400P module using insmod command, I get following error
messages. Is there some problem with my asterisk installation. Please advise. Thanks
Tony
$insmod wcfxs
Using /lib/modules/2.4.20-8/misc/wcfxs.o
/lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol
Hello,
Anyone aware of any CRM products projects that intagrete with *? Or that
integrate with any telephony products? Is there some open API for such
integration, or are they all proprietory?
Thanks
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
You could try
$ modprobe zaptel
$ modprobe wcfxs
You need the zaptel bits first.
At 09:52 23/12/03, you wrote:
$insmod wcfxs
Using /lib/modules/2.4.20-8/misc/wcfxs.o
/lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk
___
I seem to have the same problem now,
were you able to resolve this ?
joachim.
At 22:41 6/11/2003 -0500, you wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot
On Tue, 2003-12-23 at 10:52, tony banks wrote:
Hello
When I tried loading TDM400P module using insmod command, I get following error
messages. Is there some problem with my asterisk installation. Please advise. Thanks
Tony
$insmod wcfxs
Using /lib/modules/2.4.20-8/misc/wcfxs.o
Hi,
I'm a newbie to the list, but have been screwing around with Asterisk
for the last 6 months or so (on a purely experimental basis so far).
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm
unsure where the line is drawn in terms of Linux issues or Asterisk
issues.
At
Hi,
Has anybody been successful in running the 7914 expansion unit for the
Cisco 7960G IP phone? For anybody unaware of what the expansion unit
does, it provides 14 additional buttons, with an LCD display. The idea,
is that with an expansion unit (a 7960 can take upto 2 of these units),
a
Hi,
In rc.local I added the line /etc/rc.d/run-asterisk
I then created a small script of 2 lines called run-asterisk
#!/bin/sh
/usr/sbin/asterisk
do a chmod 755 on the file and reboot.
The Asterisk server then starts at every reboot.
Regards
Dave
-Original Message-
From: [EMAIL
I use http://cr.yp.to/daemontools.html. Besides starting asterisk on boot
up it keeps an eye on the process and restarts asterisk if it crashes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Tuesday, December 23, 2003 6:38 AM
To:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 - Delay 5ms
Packet 53 - Delay 50ms
Packet 54 - Delay 5ms
Packet 55 - Delay 5ms
Is there anyway to space them out evenly at 20ms??
The 20 ms is not the inter-packet timing, its the relative content of what's
within the
An even better way to get asterisk started is to use the init scripts provided with
asterisk and the zaptel kernel modules.
cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk
cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel
Then do the proper linking, etc to get asterisk to start in your
I'm using grandstream phones, when on a call and a second call comes in
the
call waiting indication is to play ringing which means you cant actually
hear
your original call. I want to stop this but cant, heres my options
1. Change the callwaiting indication, I assume this is produced
Hi Guys..
Just wondering if someone could help me with a few questions please. were currently
using the ulaw codec with our grandstream/iconnect/asterisk setup and its working
pretty good except for the fact it downloads heaps. Does anyone know a good site to
get referances to how much each
make config does both the copy and the neccecary linking...
Thorsten
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott
(7805)
Sent: Tuesday, December 23, 2003 8:50
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Starting Asterisk
Eduardo Goncalves wrote:
On Mon, 22 Dec 2003 15:48:37 -0600
Steven Critchfield [EMAIL PROTECTED] wrote:
asterix:~# modprobe tor2
Zapata Telephony Interface Registered on major 196
Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000
irq 7 Did not get DONE signal. Short file
You should be able to just order Trunk Lines.
They are also known as ground start lines. They are usually for
incoming only so you would have something like 4-5 Trunk lines for the
incoming DID's and the rest would be regular pots lines.
In your CAC, you would take the Trunk lines and they
hello,
Is there a way to transfer the call via gnophone, without calling other
user and pressing conf on both calls, it seems that all traffic is still
going through the gnophone, not that optimal i guess.
thanks
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
Hello All,
Does anyone here know how I might provide music into a conference room
when there is only one participant. Dead silence tends to confuse
non-techies who think that they've done something wrong, even after the
entry announcement.
Michael
--
Michael Graves
Hello,
I'm having some * and sendmail integration problems, probably because i don't know too
much about sendmail. My server crashes when I forward voicemail from one * voicemail
box to another, everything else works. E-mail notification works on all boxes when
new mail arives, the problem
We're starting to integrate * with our customer service software.
Basically we're pulling off events from the management interface. We're
also making some small patches to the code to deliver more events about
the channel variables, etc.
Anton Yurchenko wrote:
Hello,
Anyone aware of any
Interesting. For the record, the MultiTech MVP-130 comes with a default
setting
of 60ms packets on all of its supported codecs. I changed the packet
sizes to
20ms because I had never heard of anyone using such large sample sizes.
Andres wrote:
On Monday 22 December 2003 19:58, Rich Adamson
On Tue, 23 Dec 2003, Rich Adamson wrote:
If a collision or dropped packet occurs (in a voip udp environment) there
is no way to retransmit the missing/damaged packet. Missing one packet isn't
a big deal, but if you have collisions and/or dropped packets, there is a
very high probability that
Which events do you refer?
Regards,
Gus
- Original Message -
From: Jonathan Tew [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 12:25 PM
Subject: Re: [Asterisk-Users] Asterisk + CRM
We're starting to integrate * with our customer service software.
Hello,
I had a partner of mine present a Centrex 21 brochure and ask how many of
those features can I fulfill. There is nothing out of the ordinary, it's
stuff like call hold, call forward, 3-way calling, etc. Has anyone
assembled a how-to that shows how to configure PBX or Centrex type
Hi all,
I am trying to get Capi Dial to use a specific outgoing msn. I can't get
it to work. If I make a test call to 0703241494 (same isdn line, just
one of the other numbers) I don't get CLID at all. Any ideas?
; use 0703241432 as outgoing msn
exten =
The problem occurs when the software is expecting the packet in a certain
timeframe so that it can reassemble it in a timely manner. It's not a big
deal with a web page or something along that lines. But when a voice
application cannot get reassembled in a timely manner, you'll surely notice
it!
On Tuesday 23 December 2003 02:27, Hubert Kiyimba wrote:
I am working on a project vide over IP
I am asking you to inform me whether asterisk software PBX supports
video over IP
IAX explicitly supports images, video, and URLs. See the gnophone
client.
-Tilghman
Hi,
try it without prefix (else dtag uses first msn) -
so if your city code is 07032 and phone no (msn) 41432
- exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
Thomas
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Patrick
Gesendet: Dienstag,
Does anyone have any example perl agi script that does a database get. I am
being thick and can't seem to get the return value:
print DATABASE PUT big bigger biggest \n; This bit works fine
print DATABASE GET big bigger \n;
Now what do I do to get the my value from the database get??
On Tue, 2003-12-23 at 09:48, Christopher J. Wolff wrote:
Hello,
I had a partner of mine present a Centrex 21 brochure and ask how many of
those features can I fulfill. There is nothing out of the ordinary, it's
stuff like call hold, call forward, 3-way calling, etc. Has anyone
assembled a
I'm not sure under what circumstances (from an overall performance
perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
suggest choosing the size is roughly equivalent to MTU size. The 60ms
setting should result in larger packets which might be okay for high
speed uncongested
Dear members,
I am writing to inquire whether Asterisk can serve as video switching
software for the purposes of video conferencing over IP on a campus network.
Hubert
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There's no reassembly with udp, and there is no sense of packets arriving
in the same order as what was sent. Udp is a best-effort low-overhead way
of transmitting data (with UDP often times referred to as the Unreliable
Data Protocol). Changing to TCP would allow reassembly, however the
On Tuesday 23 December 2003 10:59, Rich Adamson wrote:
I'm not sure under what circumstances (from an overall performance
perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
In our network we set UAs to use 60ms (using G729). Actual data measurements
indicate a call
I have, in iax.conf the register statement:
register = username:[EMAIL PROTECTED]
This causes registration attempts to iaxtel.com for both IAX and IAX2.
Every once in a while there is a packet for port 4569 keeping the IAX2
registration alive. This is fine.
But, I have a barrage of
You have not covered very much of the
configuration that can be done here. So with that I have come up with a
very generic config for you that I have not tested and is to the best of my
memory but I will give it to you as a starting point. I am posting the
extensions.conf, zapata.conf and
You say The server crashes I assume you mean that Asterisk
core dumps and sendmail continues to run just fine. If you
can send mail out of the box sendmail is confgured well
enough and I doubt the problem is there.
If you can get Asterisk to dump then what you need to do is
use a debugger to
On Tue, 23 Dec 2003, Hubert Kiyimba waxed:
Dear members,
I am writing to inquire whether Asterisk can serve as video switching
software for the purposes of video conferencing over IP on a campus network.
Hubert
http://www.gnophone.com/
--
Chris Maj cmaj_hat_freedomcorpse_hot_info
One thing Centrex is that Asterisk is not is a turn key
system. With Asterisk you have to either build the PBX your
self or pay someone to build yu one. With Centrex you
simply write a check.
THat said, you can build anything you want so of cource the
feature list can match.
The best way to
On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
There's no reassembly with udp, and there is no sense of packets arriving
in the same order as what was sent. Udp is a best-effort low-overhead way
Right, UDP itself does not care about order, but at the application layer you
can keep track
Hello,
has anyone come across a module for WEBMIN to configure * ?
webmin info http://www.webmin.com/
Thanks
Doug
--
FREE Unlimited Worldwide Voip calling
set-up an account and start saving today!
http://www.voippages.com ext. 1003
http://www.pulver.com/fwd/ ext. 83740
Look in the directory /etc/init.d (/etc/rc.d/init.d on
some systems)
You put a script in there called asterisk. There is a
sample called asterisk.init in the source. copy it to
/etc/init.d/asterisk
You may want to study the other files in /etc/init.d to see
how they work.
Next read the
I assume there are several people on this list that
have Cisco Call Manager implementations under their
belt
We are beginning a call manager implementation and
the first question I asked Cisco was, should we use
SIP or Skinny. Cisco is pushing me towards Skinny,
saying that I will lose some
On Tue, 2003-12-23 at 17:13, [EMAIL PROTECTED] wrote:
Hi,
try it without prefix (else dtag uses first msn) -
so if your city code is 07032 and phone no (msn) 41432
- exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
Thomas
Thanks for the pointer Thomas. I removed the areacode from msn=
The reason you use UDP over TCP for realtime meadia is that
TCP's ability to reliably deliver every packet in order actually
sounds worse. Reason being is that with a UDP system a dropped
packet sounds like just a dropout but if you used TCP the audio
stream would be held up and delayed in a
On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote:
Does anyone have any example perl agi script that does a database
get. I am being thick and can't seem to get the return value:
print DATABASE PUT big bigger biggest \n; This bit works
fine print DATABASE GET big bigger \n;
Now
Hi!
Does anyone here know how I might provide music into a conference room
when there is only one participant. Dead silence tends to confuse
non-techies who think that they've done something wrong, even after the
entry announcement.
MeetMe() now has an option M that does exactly that. Be
Which events did you add ?
Van: Jonathan Tew [mailto:[EMAIL PROTECTED]
Verzonden: di 23/12/2003 16:25
Aan: [EMAIL PROTECTED]
Onderwerp: Re: [Asterisk-Users] Asterisk + CRM
We're starting to integrate * with our
Justin,
Comments inline:
At 01:06 24/12/03 +1100, you wrote:
Hi Guys..
Just wondering if someone could help me with a few questions please. were
currently using the ulaw codec with our grandstream/iconnect/asterisk
setup and its working pretty good except for the fact it downloads heaps.
I've used both the syntax you have given and the perl module. AGI-getvar()
returns nothing for arguments that work from the CLI
(Also when I run agi-test.agi, the only thing that works is the SAY NUMBER.
SEND TEXT doesn't work and nothing at all is printed to teh console)
I am using redhat 8.
Hi!
try it without prefix (else dtag uses first msn) -
so if your city code is 07032 and phone no (msn) 41432
- exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
Thomas
Thanks for the pointer Thomas. I removed the areacode from msn= in
capi.conf and from the dial statement. Tried
i mean AGI-database_get()
- Original Message -
From: Muhammad Nasim [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 6:41 PM
Subject: Re: [Asterisk-Users] perl database get
I've used both the syntax you have given and the perl module.
AGI-getvar()
returns
On Tuesday 23 December 2003 11:13, Robert Hajime Lanning wrote:
I have, in iax.conf the register statement:
register = username:[EMAIL PROTECTED]
This causes registration attempts to iaxtel.com for both IAX and
IAX2.
Every once in a while there is a packet for port 4569 keeping the
IAX2
I have found a phone that I wish I had not! This is by far the worst
phone to setup. I have finally upgraded it to Sip but once this got
done it I am not able to get it unlocked so I can enter the rest of the
settings. So if anyone out there can tell me how to setup my DNS server
to tell it
I have read a lot about ztdummy, but I miss something.
I don't have any digium hardware, but want to do Meetme.
I read that I need ztdummy installed in order to do a conference room.
I followed all steps to get ztdummy compiled and installed (including uncoment on
makefile)
When I install the
On Tue, 2003-12-23 at 19:39, Philipp von Klitzing wrote:
[snip]
If you read the CAPI documentation you'll find that @ will help you to
_hide_ your ID (this is called CLIR) - however from your message I
understand that you want to do the opposite? So just drop the @.
If your problem
What firmware did you upgrade to?
If its version 5.0 and above the default password is cisco and to unlock
it you press settings then 9.
NO cisco's docs are simple.. You are just trying too hard.
bkw
On Tue, 23 Dec 2003, Ariel Batista wrote:
I have found a phone that I wish I had not! This
I am looking to speak to anyone else that has connected to Voiceglo using
Asterisk. I'm using SIP and have most of the issues worked out. But remote
outbound ringing doesn't work. So it would be nice to discuss configs.
Maybe someone out there is using IAX instead.
cameron.
Brian West wrote:
What firmware did you upgrade to?
If its version 5.0 and above the default password is cisco and to
unlock it you press settings then 9.
NO cisco's docs are simple.. You are just trying too hard.
I want to thank you for the password of cisco. It worked. I have
finally
I've got to agree. Once you figure out the first phone, all the others
take about 30 seconds to configure.
The Cisco SIP documentation is located at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/index.htm
cameron.
On Tue, 23 Dec 2003, Brian West
On Tuesday 23 December 2003 12:41, Muhammad Nasim wrote:
I've used both the syntax you have given and the perl module.
AGI-getvar() returns nothing for arguments that work from the CLI
Try AGI-get_variable()
(Also when I run agi-test.agi, the only thing that works is the SAY
NUMBER. SEND
On Tuesday 23 December 2003 12:44, Muhammad Nasim wrote:
i mean AGI-database_get()
Then that probably means that the database key does not exist.
-Tilghman
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Thanks for the reply.
1. My VAD is turned off (00140014), and it didn't help for that cut-off.
I
am not sure if OutboundProxy has to be configured to have it working fine.
Or this just happened to me? What is your ATA's software?
2. I tried dtmfmode=inband on sip.conf, and
How do you reset the unit without pulling out the plug. The easiest way
to get the info you are looking for, is to get an 8 buck CCO account.
On
Tue, 23 Dec 2003, Adthrawn wrote:
Hi,
Has anybody been successful in running the 7914 expansion unit for the
Cisco 7960G IP phone? For
Hi!
1. My VAD is turned off (00140014), and it didn't help for that cut-off.
Then check if you have a firewall in between * and your ATA that closes
the port due to inactivity of your ATA. Also use SIP DEBUG in the CLI
to try to see a bit more of what is going on. You could also use
Hi Chris,
In this situation, how do I modprobe ztdumy before * get started ?
SW
Message: 6
Date: Tue, 23 Dec 2003 09:33:07 -0800 (PST)
From: Chris Albertson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Starting Asterisk
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Look in the
7914's don't work with SIP. SCCP only. And why do people keep talking
about this 8 dollar CCO account ... Its a service contract on the Cisco
ATA-186. The one for the 79XX's are over 80.00/yr
bkw
On Tue, 23 Dec 2003, Lists wrote:
How do you reset the unit without pulling out the plug. The
Is there anyone who could show me code (or point me in
the right direction) to allow users or PABX Admin to
generate their own * conf files.
If there isn't anything I will just have to start it
myself. Any suggestions for basics to start with. I
believe the issues are going to be about
Hi Philip, I found the problem. My sip.conf config was changed by somebody
else. :( The external IP was uncommented and that's what is causing my
problem.
- Original Message -
From: Jess Magnaye [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, December 23,
I have gotten the Cisco 7960 working with my Asterisk system under SIP.
The version is 5.03 that I am using. Cisco Support said I should not
upgrade to version 6 yet. My next question is the sound is patchy when
people here me. But I can hear them just fine not patchy. I have the
188 page Admin
I know someone mentioned doing this once before however I
cant find it.
Anyone remember if or how it was successful?
Thanks!
Ariel Batista wrote:
P.S. the Contract for the 7960 cost us $ 83.40 for each phone. This I
feel is high.
This smells like a Cisco re-certification fee to me.
Jeremy McNamara
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resending.
Can anyone help me in trying to understand what
would be the problem. appreciate ur time. i need to get this
working.
thanks a lot,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 8:15
PM
Subject:
It's the day before Christmas here in Sweden, actually the night before at this time...
We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
merry-christmas-mode with the yet undocumented CLI command frosty-mode on, a mode
where the PBX will connect all incoming
Is there anway to do MOH on a FXS extension like what is done using SIP.
There has to be a way within manager or something, to send this call to
MOH and then retreive the call.
I need to set this up, so users are just hitting one button to put
callers on hold and one or another button to retrieve
Is there any place where I can download sample files for the
cisco 7960 (SIP) ?
The call quality is really pretty good. I think better than Vonage over
an FXO bridge. If you are looking for a home provider with direct SIP
support and local phone numbers this is a good choice. If anyone has
questions or comments about my configuration please pass them along. I
have noticed
Merry Christmas Ollie from all of us Asterisk people in the US/East Coast
region.
AJ
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If you purchase a new telephone, the warranty is more like $15. It's more
for used phones.
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, December 23,
Speaking of MSN/Windows Messenger, how does one call someone?
Using the configuration specified, I've registered it with Asterisk, but
it requires that I add a Passport contact.
Does anyone have experience calling a sip endpoint without it being a
Passport account?
-g
On Mon, 2003-12-22
Balaji,
I also have the
same issue. Works fine on any phone except GS for me.
After a bit of
research I found a post saying set the phone to offer only one codec set.
It looks like we
have to set the phone to use one codec GSM
I am concerned
that you cant use passwords when
100% agree. I think this thread is getting strung out much further
then Olle's original question relative to commenting on half vs full
duplex.
Lots of great discussion though thanks to all that participated!
Rich
The reason you use UDP over TCP for realtime
What firmware did you upgrade to?
If its version 5.0 and above the default password is cisco and to
unlock it you press settings then 9.
NO cisco's docs are simple.. You are just trying too hard.
I want to thank you for the password of cisco. It worked. I have
finally gotten the
I'm just double checking.. I was told it wasn't possible but i'm going to
ask just in case.
Can you set outbound callerid on a channelized T1?
bkw
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It's the day before Christmas here in Sweden, actually the night before
at this time...
snip
It's been fun spending the fall with the Asterisk project. I look forward
to next year, with the new handbook coming in place, with many new
applications and features and - hopefully - many new
On Tue, 2003-12-23 at 19:22, Brian West wrote:
I'm just double checking.. I was told it wasn't possible but i'm going to
ask just in case.
Can you set outbound callerid on a channelized T1?
I think there is a way to do something like DID with the 4 digits of
DTMF passed before the call. It
7914's don't work with SIP. SCCP only. And why do people keep talking
about this 8 dollar CCO account ... Its a service contract on the Cisco
ATA-186. The one for the 79XX's are over 80.00/yr
Careful Brian... things aren't always what they seem. There is some
flexibility built into their
HAHA you apparenlty aren't where we are... PRI is over priced... 3600/mth
SBC Victim... they have to backhaul it 110 miles.. where CT1's can be
served by the local CO.
bkw
On Tue, 23 Dec 2003, Steven Critchfield wrote:
On Tue, 2003-12-23 at 19:22, Brian West wrote:
I'm just double checking..
I have gotten the Cisco 7960 working with my Asterisk system under SIP.
The version is 5.03 that I am using. Cisco Support said I should not
upgrade to version 6 yet. My next question is the sound is patchy when
people here me. But I can hear them just fine not patchy. I have the
188 page
The fun part is getting a clueful reseller on the phone to sell you the
correct thing.
bkw
On Tue, 23 Dec 2003, Rich Adamson wrote:
7914's don't work with SIP. SCCP only. And why do people keep talking
about this 8 dollar CCO account ... Its a service contract on the Cisco
ATA-186. The
Hi all,
Just thought I'd pass along some pointers when outdialing with Voicetronix's
OpenLine4 card.
I was having a tough time dialing out from *, it probably has something to
do with chan_vpb.c not waiting to hear the dialtone before telling the card
to dial. A quick fix was to insert a , in
Merry Christmas from the Colorado Organization for Victims' Assistance.
Our (Comdial) PBX fried after a power failure. Thanks to Mark Spencer,
Digium, VCCH, and the friends who support this group, we are now back
on the air.
We wish everyone good health for the coming year.
merry xmas olle and you all in the list happy holidays!!!
Miguel
On Tue, 2003-12-23 at 23:04, Olle E. Johansson wrote:
It's the day before Christmas here in Sweden, actually the night before at this
time...
We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
u can ignore the passport request.
u need to change the registry settings to make a phone
call. Do a search and
u ll find the details.
-B
- Original Message -
From: Glen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 3:57 PM
Subject: Re: [Asterisk-Users] MSN
i tried with only GSM too. With only GSM it doesnt
even connect to GS. Then someone recommended to use ulaw and alaw and that
helped. But the call drops after 10 secs. i did a 'sip debug' and what i found
is that MSN doesnt even recognize that call is in progress
and then drops the call.
this is a time to reflect, and i have much to reflect for come the end
of a year.
for all those that i've pissed-off through out the year with nasty
comments and such...
merry christmas to all.
sorry (but come the 1st it's a new year and therefore can create a new
list to atone for) G
Ok anyone ever detect and generate DTMF ABC and D?
bkw
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