Hello all,
I know this isn't strictly Asterisk, but I'm sure that there are more people
here using the Cisco 7960 w/ SIP, so I thought I'd post here.
I've just bought a Cisco 7960 phone to use with Asterisk. It came with the
CallManager image on it.
I've got the 4.4 SIP images (P0S3-04-4-00).
Don't know if this will help, but have you seen:
http://www.junghanns.net/asterisk/page12.html
?
On Friday, September 12, 2003, at 03:06 AM, Jamie Carl wrote:
Hey all,
I was playing around with IAXTEL last nite and have outgoing calls
working a treat. I'm sure I woke a few people up in the
Hello again,
After doing some searching of the list archives, I came across a message by
John Todd posted back in July ()
To cut a long story short, to be able to use SIP on my phone, I need to
P0S30203.bin image.
Is there anyway of getting this image without getting a Cisco SMARTnet
agreement?
Archiving calls depends on the definition of archive. When do you actually
archive?
If the information is stored in a buffer (memory, disk, tape) for technical
reasons - do you then archive the call?
Of course in general one is not allowed to store the information for a long
period on his
John Todd wrote:
I'm using a Cisco 7960 with asterisk and any recording
on the machine, be it voicemail prompts, time of day,
echo test message, etc, is cut off for the first 1/4 to
1/2 second. I've tried setting the phone to gsm but
it still happens.
Before running any application that has
On Fri, 12 Sep 2003, Olle E. Johansson wrote:
Before running any application that has sound playback (Playback,
Background, VoiceMailMain2, etc.) it would be wise to execute an Answer
first, then a Wait(2) to allow for VoIP channels to fully establish and
settle.
John, in order to
Hi all,
Got it working. Time to have some fun :)
-Shaun
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 12 September 2003 04:33, Andrew Joakimsen wrote:
I know it is possible to record calls, it will record them to a
directory you define on the server. But are you required to provide
archives/recordings of the calls or permit real-time
Hi Shaun and anyone else looking for Cisco images,
I don't know what the support contract would cost on a 7960 for the
Cisco TAC, but for the ATA186 it's a great, big $8/year. This gives you
access to the Cisco TAC, images, and support team which do a fantastic
job of follow-through.
So I
I guess the subject line says it all. I want to purchase a card to use
in my small office/call center.
I have ten lines coming into this place, and I current have four of them
for public use... *call center* and the rest for business. But the
problem is, the lines are not all supplied by the same
Tais M. Hansen wrote:
I know it is possible to record calls, it will record them to a
directory you define on the server. But are you required to provide
archives/recordings of the calls or permit real-time tapping?
You should ask some kind of justice department.
We did here in Denmark and were
Lee Goodman wrote:
Lets say I have an * at my business, with 7960 SIP phones. All the sip
phones are registered using their extension number (like 305), but I would
also like to put my SIP URI on my business card and in a name format, not an
extension number (like lee.goodman), so that the SIP
Timothy Soos wrote:
Good. Since I do not have the VoIP system set up yet, and I control the IP
network (which is still small), please tell me what I need to do to monitor
and record SIP to SIP calls.
In SIP.conf:
canreinvite=no
Then use the Monitor application.
--
Alastair Maw [EMAIL
On Thu, Sep 11, 2003 at 09:30:35PM -0700, John Todd wrote:
Before running any application that has sound playback (Playback,
Background, VoiceMailMain2, etc.) it would be wise to execute an
Answer first, then a Wait(2) to allow for VoIP channels to fully
establish and settle.
Adding
On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote:
nope
when I click on something on the left I get a FQDN not just the pne you had
Hmmm.
Further info: it works with Microsoft Internet Explorer. It
does not work with Mozilla 1.4 under Linux. It also does
work with
On Fri, 2003-09-12 at 00:46, [EMAIL PROTECTED] wrote:
Hi all,
When using trying to dial a number in the US, all I get is a Sprint recorded
voice saying something like Number could not be recoginiised, Please hit 1
then enter the area code you want to dial in, then something or rather like
Hi,
I use * fine with my ata186 but i can't find on the web an answer to my
question, in the configuration menu you can choose the caller id method,
does somebody in europe know what must i but into the field to make it work
with my french phones ?
The caller id displayed fine if i'm in
On Thursday 11 September 2003 02:13 am, Timothy Soos wrote:
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not be possible.
Thank you to everyone that answered. I will do some further study and
experimentation
I know this isn't strictly Asterisk, but I'm sure that there are more people
here using the Cisco 7960 w/ SIP, so I thought I'd post here.
I've just bought a Cisco 7960 phone to use with Asterisk. It came with the
CallManager image on it.
I've got the 4.4 SIP images (P0S3-04-4-00).
If
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,
On Fri, 5 Sep 2003, Zak wrote:
I have changed the pci slot of the fxo so that it won't share IRQ with
another device but the
Hi all!
Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and
I faced with a low call volume heard just for people who is not under the
ATA. I mean, if I call a person whose extension is connected at the ATA, he
can hear me perfectly, but I get a low call volume.
Is it
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
Try the CLI command:
SIP debug
...and you'll propably see that the FWD SIP server fwd.pulver.com answers something
like
ough... That's ugly...
It's referring to the fact that you're trying to communicate with a private IP
address (192.168.x.x).
With SIP clients like Xten, you configure
There has been discussions about the voicemail menus and some of us
would like to see an overall plan for the voicemail menus.
There are 3 primary ways of arranging the menus. First is a tree
structure, second is a random access structure and the third would be a
hybrid of the two. (Comedian
I have been reading archive post in regards to h323 support, and I am not
clear on this:
1.
Is h323 enabled and ready to use if one compiles asterisk, zaptel and libpri
(as shown on asterisk.org) ?
2.
If it is, is it h323 or oh323?
3.
If it is not, does one just need to follow instructions in
Semi-works with Opera. Some of the javascripting doesn't work. But aren't
we all happy IE users these days? ;)
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Pauly
Sent: Friday, September 12, 2003 6:35 AM
To: [EMAIL PROTECTED]
Subject: Re:
Is there a way to limit the duration of any single voicemail recording?
I'd like to put a cap on that limit, say 2 minutes or whatever, for those
long winded individuals and can't seem to find a reference for it.
___
Asterisk-Users mailing list
Did you see /etc/asterisk/voicemail.conf ?
maxmessage=120 is 2 minutes
Martin
On Fri, 12 Sep 2003, Rich Adamson wrote:
Is there a way to limit the duration of any single voicemail recording?
I'd like to put a cap on that limit, say 2 minutes or whatever, for those
long winded
I searched for say number in the * google archives and have not found reference to
options for say number. I would like to have * say digits instead of the hundreds
and thousands. EG, 1234 would say one two three four.
___
Asterisk-Users mailing list
Thanks Martin, my config did not have that in there. Don't know why, but
its there now. Thanks.
Did you see /etc/asterisk/voicemail.conf ?
maxmessage=120 is 2 minutes
Martin
On Fri, 12 Sep 2003, Rich Adamson wrote:
Is there a way to limit the duration
I have found this behavior with the 7960's, but not with the Pingtel
Xpressa or ATA-186. I think I've read it's a bug in the 7960 firmware; I
am running 5.1 and see this problem. If anyone knows if it's been
corrected in 5.3, I'd like to know. Meantime the Answer/Wait sounds like
a good
Oops should have looked a little harder to find the say digits.
Sorry.
On Fri, Sep 12, 2003 at 10:21:32AM -0500, PJ Welsh wrote:
I searched for say number in the * google archives and have not found reference to
options for say number. I would like to have * say digits instead of the hundreds
I think what you're looking for is ast_say_digits -- see
http://pbx.usedontmiss.com/say_8c.html#a2
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast?
Use oh323.
Download the openh323 and pwlib tarballs from openh323.org
Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY!
good luck
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]
-Original Message-
On 12/9/03 14:19, Dave Cotton [EMAIL PROTECTED] wrote:
On Fri, 2003-09-12 at 14:34, Zara Trousk wrote:
Please note that 192.168.0.10 is my internal IP, xxx.xxx.xxx.xxx is my
external IP and NNN is my FWD number (ID)
;
register =N:[EMAIL PROTECTED]/EXTEN
I can assure you that FWD
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
Try say digits digits
Bye bye
- Original Message -
From: PJ Welsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 12, 2003 5:21 PM
Subject: [Asterisk-Users] say number question
I searched for say number in the * google archives and have not found
reference to options
We've changed E1 providers and I'm trying to reconfigure an E400P to
make it work with the new lines. They're supposedly standard EuroISDN
lines (in the UK). I'm initially just trying to get a single line up.
I have the following in /etc/zaptel.conf:
span=1,0,0,ccs,hdb3
bchan=1-15
Thanks John and all,
Unfortunatelly this will not work for me, because the SIP phones are
agents and I'm managing incomming calls through a queue.
Anyone knows a SIP softphone that supports disabling call waiting?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I've tried to find documentation on if Asterisk supports DNS SRV records for sip servers.
Reading the source of channel_sip.c it seems not:
hp = gethostbyname(hostname);
if (!hp) {
ast_log(LOG_WARNING, Host '%s' not found at line %d\n, hostname, lineno);
return -1;
}
why not h323?
Andrea
Langley, Sean wrote:
Use oh323.
Download the openh323 and pwlib tarballs from openh323.org
Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY!
good luck
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
The Voicemail2 is better one, has more bug fixes, more functionality and
Voicemail (1) should stop existing soon.
regards
Martin
On Fri, 12 Sep 2003, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and voicmail2() ?
The show
On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and voicmail2() ?
The show application commands contains exactly the same text, giving
no hints.
From the application stand point there is little difference,
this behavior looks alot like cisco bug CSCdz77783:
---
When a 7960 IP phone using SIP, makes a call to a device like a media server
which starts playing the immediately after answering the call, a clipping of
first couple of words may be noticed.
There is no known workaround for the
Hi all
I've only been working with Asterisk for a matter of days but have
already grown into a big fan =) Much as I've managed to get internal
calling working fine, I have a configuration running on an old PII-233
on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond
W6692 PCI Card
If your proider is using a System-X switch don't bother
the e400p does not work correctly with those, you need a te410p
and your problem sounds like that. the e400p has a framer that is not
compatible with some switches.
sorry to tell yoyu that
Michael
On Friday 12 September 2003 5:43 pm,
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_book09186a00800c31f5.html
On Fri, 2003-09-12 at 06:11, Julien wrote:
Hi,
I use * fine with my ata186 but i can't find on the web an answer to my
question, in the configuration menu you can choose the caller
Try adding ,crc4 to the end of the span definition.
Mark
On Fri, 12 Sep 2003, Alastair Maw wrote:
We've changed E1 providers and I'm trying to reconfigure an E400P to
make it work with the new lines. They're supposedly standard EuroISDN
lines (in the UK). I'm initially just trying to get a
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323 gateway (a
h323 - PSTN gw)?
- Asterisk ip: 192.168.1.10
- h323-PSTN gw: 192.168.1.20
I've tried:
exten = _9,1,Dial(OH323/192.1.1.20)
or
exten =
Title: New Implementation Questions
G'Day,
I am looking at possibly implementing this system and was wondering if anyone knew recommended system requirements?
I've only been looking at this product for a few days and I'm not sure of all the questions to ask for researching a new
OK, so I've done this:
*CLI pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
Sending Set Asynchronous Balanced Mode Extended
[00 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 0 EA: 0
TEI: 000EA: 1
M3: 3 P/F: 1 M2: 3 11: 3 [ SABME ]
0 bytes of data
Um, you have to take your Rep out to breakfast? Sounds like someone's got a
good scam going on :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David C. Troy
Sent: Friday, September 12, 2003 11:26 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Steven Critchfield wrote:
On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and voicmail2() ?
From the application stand point there is little difference, but from
the configuration stand point there is a fair
All,
I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway.
Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know
which cards, if any, exist for a 7206VXR to act in a similar capacity,
either as a T1/PRI, DS3, or POTS FXO/FXS?
What other Cisco routers can act as
I beleive the newer 3700 series3725, 3745. I just got a quote from
Verizon for a 3725..
Mike
- Original Message -
From: David C. Troy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 12, 2003 1:24 PM
Subject: [Asterisk-Users] 7206 as SIP-PSTN Gateway?
All,
I
Also, don't limit yourself to Cisco. There are many vendors out there that
make SIP trunking gateways...
- Original Message -
From: David C. Troy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 12, 2003 1:24 PM
Subject: [Asterisk-Users] 7206 as SIP-PSTN Gateway?
All,
Hi all!
Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and
I faced with a low call volume heard just for people who is not under the
ATA. I mean, if I call a person whose extension is connected at the ATA, he
can hear me perfectly, but I get a low call volume.
Is it
Sean,
Thanks very much...
I was really not clear what I supposed to do.
Now I should be able to enable h323 in my * box.
Am I going to do configuration changes in oh323.conf or h323.conf files?
PSS... h323 gateways do not perform authentication! Is that true? (my
question no 4 !!!)
Thanks
SER, the SIP Express Router, version 0.8.11 have a new solution for supporting NAT,
the nat helper module.
See http://www.voip-info.org/tiki-index.php?page=SER+nat+support
It mangles the SIP header, the SDP and also actively keeps the NAT assosciation for
clients behind NAT open in order to be
exten = _9X.,1,Dial(H323/[EMAIL PROTECTED])
If it's not working it's worth looking at the reson:
h.323 debug
h.323 trace 3
regards
Martin
On Fri, 12 Sep 2003, Cerrajetto wrote:
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323
So you don't receive any answer from the other side ?
Is the circuit in alarm ? Can they do remote loopup test ?
It might be that they don't have their D-channel turned on ...
Martin
On Fri, 12 Sep 2003, Alastair Maw wrote:
OK, so I've done this:
*CLI pri intense debug span 1
you can copy voicemail.conf.sample to be your voicemail.conf ...
Martin
On Fri, 12 Sep 2003, Olle E. Johansson wrote:
Steven Critchfield wrote:
On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and
Hi,
Anyone configured X-Lite with asterisk? Any help on config, both on X-Lite and
asterisk?
I managed to have SJPhone working OK with * but not X-Lite.
Any ideas?
-Z
--
__
Sign-up for your own personalized E-mail at Mail.com
[snip]
Regarding the real-time monitoring, it'd be great if we could
develop an extension to zapbarge/scan that let you tap in the
callerid of the person you're wanting to monitor. It's a little hard
to find things in a 120 channel bank sometimes...
--
Alastair Maw [EMAIL PROTECTED]
MX Telecom
All,
I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway.
Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know
which cards, if any, exist for a 7206VXR to act in a similar capacity,
either as a T1/PRI, DS3, or POTS FXO/FXS?
What other Cisco routers can act as
[EMAIL PROTECTED] tftpboot]# cat OS79XX.TXT
P0S30100
Get this image as well.
Shaun Ewing wrote:
Hello all,
I know this isn't strictly Asterisk, but I'm sure that there are more people
here using the Cisco 7960 w/ SIP, so I thought I'd post here.
I've just bought a Cisco 7960 phone to use with
At 11:24 AM 9/12/2003, you wrote:
Anyone configured X-Lite with asterisk? Any help on config, both on X-Lite
and asterisk?
Yep. Here's an acceptable SIP.CONF entry...
[551212]
type=friend
username=551212
secret=12345678
host=dynamic
qualify=1000
nat=1
mailbox=551212
context=default
Does anyone know how you specify MD5 auth on a register = line?
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello
I have my * behind a NAT (Netgear) but on the Netgear DMZ
I have my * register to a Cisco proxy on the outside of the NAT.
The registration works, but the Via and Contact fields still have the inside
address.
Yes, I have nat=yes
Yes, I have fromdomain=ipaddress
Any ideas
sample config
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| Mark Spencer wrote:
| | Remember the S100U has to generate high voltage (-48V when idle, plus
| | about 70v a/c on top of that when ringing), and it does it all with 5V.
| | It does get a little warm. That's normal.
|
| Wow.
To clarify, you have to upgrade to 3.1 or 3.2 first, before attempting a
version four load.
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore
Sent: Friday, September 12, 2003 1:41 PM
To: [EMAIL PROTECTED]
Subject:
Langley, Sean wrote:
Use oh323.
um no, its H323.
Download the openh323 and pwlib tarballs from openh323.org
Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY!
OH323 is that third party driver that is out there. You cannot follow
directions for H323 and
[top-posting madness continued]
Instead of making Asterisk do this work, wouldn't it make more sense
to just have a smart ethernet sniffer that handled the whole
transaction? I have no details on it, but I would guess that the
previously-named Carnivore project here in the USA and it's
I have a parking.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 45 ; Number
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
You may find more
Actually, I think that making the individual keystrokes
user-configurable would not be all that difficult...
I like the idea of being able to totally customise how it works. I know
Octel does this - when I worked at Jersey Telecoms we replaced our carrier
voicemail platform with an Octel one and
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
You may find more
On Friday 12 September 2003 02:37 pm, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters,
Is it this maybe?
Communication controller: Tiger Jet Network Inc. Model
300 128k
__
Do you Yahoo!?
Yahoo! SiteBuilder - Free, easy-to-use web site design software
http://sitebuilder.yahoo.com
___
Asterisk-Users mailing
What does 'dmesg' says ?
Martin
On Fri, 12 Sep 2003, James Sharp wrote:
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
To followup this is what debug shows:
DEBUG[1158913328]: File chan_zap.c, Line 2602 (zt_handle_event): Got event
Wink/Flash(3) on channel 5 (index 0)
DEBUG[1158913328]: File chan_zap.c, Line 2893 (zt_handle_event): Winkflash, index: 0,
normal: 15, callwait: -1, thirdcall: -1
DEBUG[1158913328]:
More followup:
If I call echotest it doesn't work, but if I call a real phone (Zap or
SIP via a remote Asterisk Server via IAX it works.
On Fri, 2003-09-12 at 14:32, Eric Wieling wrote:
I have a parking.conf:
[general]
parkext = 700 ; What ext. to dial to park
Hi,
I have some questions regarding IAX, IAX2 and encrypted authentication.
How can I know if IAX or IAX2 is used between two * servers?
There is any guide about how to configure encrypted authentication (not in
clear text)between two * servers?
I hear on this list a couple of days ago that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
In your dialplan, do you perhaps have some sort of overlap that it's
going to instead of the call parking?
Something to consider.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Wieling
Sent:
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 12, 2003 7:09 PM
Subject: Re: [Asterisk-Users] ATA caller ID
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_book09186a00800c31f5.html
On Fri,
IAX2 uses 4569 UDP port.
You can see iax2 calls with iax2 show channels. Also you can send the
calls in IAX2 simply by Dial(IAX2/blahblah)
Also IAX2 is more recent, has more fixes and has the trunking mode to save
bandwidth if you're sending more than 10 calls to another destination.
regards
what does 'dmesg' says ?
Martin
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect
Hi,
I recently purchased some G729 licenses for asterisk. I'm concerned with the
registration process. My build tools are not physically located on the same machine
from which I build asterisk. I build RPMs on another machine and then install them on
my production server. Am I going to
Hi
I have a problem with a new DECT phone I have bought
The key pad works like a Mobile phone where you dial first then pick up
the line, but it seems to dail too fast or spuriously, ie 012826736464
show on thew Asterisk console as 0012282677, could any one offer advice
how to fix?
Also when
On Fri, 12 Sep 2003, Martin Pycko wrote:
Also IAX2 is more recent, has more fixes and has the trunking mode to save
bandwidth if you're sending more than 10 calls to another destination.
martin, why 10 calls? is this codec dependent? thanks in advance for the
info...
- wasim
Yes!
The registration hardware locks to system parameters, and does some
software locking to certain inodes in your linux fs.
You must run registration on the production server. Their license
scheme sucks!
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
Because IAX2 in trunking mode adds the 10 bytes header ... So It might not
be a good idea if you're going to have only two calls.
Martin
On Fri, 12 Sep 2003 [EMAIL PROTECTED] wrote:
On Fri, 12 Sep 2003, Martin Pycko wrote:
Also IAX2 is more recent, has more fixes and has the trunking mode
Hrm, another issue. My phone does G729A... This codec only does G729B but is
advertised as doing G729.
Guess I'm SOL.
-z
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Zac Sprackett
Sent: Friday, September 12, 2003 4:19 PM
To: [EMAIL PROTECTED]
I have a * setup with Grandstream SIP phones dialing out through
Nikotel via SIP. When i dial out and the other side picks up, the
Grandstream keeps ringing for another seconds and two and the sound
coming from the other side is lost. After these two seconds the call
is connected find and works
I use both Ciscos and Asterisk as Sip gateways to pstn.
I can say a lot of good things about both, and a few bad
things as well.
The Ciscos are a very solid product with very good very fast tech
support. It also has some really nifty fax detection with
redirection via email options (AS5300 and
Is there a way to determine which channels belong to fxo vs. fxs
devices? I need to write an auto-configuration program that can match
up channel numbers to types.
I have to assume there's an unknown ordering of fxo and fxs cards.
Suggestions? TIA
Trying to figure out why I'm having all of my test (and demo) perl script in a defunct
status. Each run creates a problem:
ps output
root 26253 1356 0 16:39 pts/100:00:00 asterisk -vvvc
root 26270 26253 0 16:40 pts/100:00:00 [pj.pl defunct]
root 26271 26253 0 16:40 pts/1
For anyone who's interested,
I called Cisco at 800-213-1542 and talked to a Service Contract Sales
Representative for a Smartnet SNT 8x5xNBD on my 7960. It was $8/yr,
charged to my credit card, and I had my CCO access four hours from the
time I'd called. You will need to have the serial number
1 - 100 of 114 matches
Mail list logo
