Hi,
> -Original Message-
> OK. Well, here are a couple of newbie-type thoughts on the whole
> Vertical Service Code (CLASS) hard-codings.
*snip*
Yes, having a way to redefine class codes would be excellent. Especially
since 'industry standard' only means 'industry standard in greater U
> Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They
> advertise "no new software features", but it does include bugfixes for a
> number of things. I know there was a discussion about the 0.4sec delay,
> which is said to be resolved in this firmware (CSCed48311: Media takes 0.4
I've tried with Vigor 2200 and 2500 without success. I could get a
single client to work with external Services like fwd.
With * the router freezes or reboots itself. I've tried all possible
settings, Open Ports, DMZ Host, Portforwarding, allow fragmented packets
but no success.
Seems to be the
On Tue, 25 May 2004, jo wrote:
> Sorry, no solution but same problem. Downgrading brings this message on
> Suse9.0, 2.4.21:
>
> [app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240
> ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so:
> undefined symbol: ast_get_txt
>
Try:
exten => s,1,Playback(thanksforcalling)
exten => s,2,Dial(SIP/&SIP/1112|30|m)
exten => s,3,Voicemail(uEXTEN)
exten => s,4,Playback(vm-goodbye)
That will answer and play back "thanksforcalling.gsm", dial SIP/ and
SIP/1112 with music. If not answered within 30 seconds, it will go to
v
On Wed, 2004-05-26 at 14:42 +1000, Simon Brown wrote:
> Can I do this with * ???
>
> S,1,answer call
> S,2,play "thanks for calling, we'll be with you soon"
> S,3,play music while caller waits and ring nominated extensions at same time
> S,101,if not answered go to voicemail
>
> I can't find a wa
On Tue, 2004-05-25 at 23:42, Simon Brown wrote:
> Can I do this with * ???
>
> S,1,answer call
> S,2,play "thanks for calling, we'll be with you soon"
> S,3,play music while caller waits and ring nominated extensions at same time
> S,101,if not answered go to voicemail
>
> I can't find a way to p
Can I do this with * ???
S,1,answer call
S,2,play "thanks for calling, we'll be with you soon"
S,3,play music while caller waits and ring nominated extensions at same time
S,101,if not answered go to voicemail
I can't find a way to play music and ring extensions at the same time.
Any help would
man cvs
for proper date formats.
bkw
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 25, 2004 9:36 PM
Subject: [Asterisk-Users] CVS checkout problem
>
>
> HI,
>
> Sort of new to linux. I am trying to update my asterisk and get this error
as
> b
By the way the error I'm getting in debug mode
Starting simple switch on 'Zap/1-1'
May 25 21:51:36 WARNING[327700]: chan_zap.c:4871 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
Can anyone help?
Dan
-Original Message-
From: Dan Cunningham
To: '[EMAIL PROTECTED]'
Sent: 5/
use realy IP address
and call is DIAL(h323/[EMAIL PROTECTED])
Igor Barsanti wrote:
I've setup an asterisk server H.323 compliant, with a GnuGk gatekeeper.
My h323.conf is:
[General]
port=1720
gatekeeper=127.0.0.1
context=default
[PABX]
type=H323
e164=PABX
prefix=0
context=default
my gatekeeper.ini
I use about 300 IP phone combination Welltech LP101, welltech LP102
welltech3502-8, Cisco 7905 and Cisco 7960
Steven Critchfield wrote:
On Tue, 2004-05-25 at 18:38, Jeff Gustafson wrote:
On Tue, 2004-05-25 at 14:06, Steven Critchfield wrote:
In theory Asterisk shouldn't have a lot of
HI,
Sort of new to linux. I am trying to update my asterisk and get this error as
below.
[EMAIL PROTECTED] test]# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] test]# cvs login
Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot
CVS password:
[EMAIL PROTECTED
Why do you consider kphone useless? Works pretty well on my debian laptop.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Girish
> Gopinath
> Sent: 26 May 2004 01:25
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Sip/IAX Clients for Linux
>
On Tue, 2004-05-25 at 18:38, Jeff Gustafson wrote:
> On Tue, 2004-05-25 at 14:06, Steven Critchfield wrote:
> > > In theory Asterisk shouldn't have a lot of load. It's simply routing
> > > calls (at 64kbit) and not converting them from one codec to another.
> >
> > While there is no codec, the
I'm running CVS-HEAD-05/06/04-17:52:43 . Do you think the problem is
fixed in this version, or do I have a different problem?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Tue, 25 May 2004, Eric Wieling wrote:
> Bruce Komito wrote:
>
> > I have a PRI wi
Well that certainly seems to say there can be glare, but a read of the
q.931 protocal stack seems to suport that it's not possible. If I were
a leser man, I could say a snide remark about that if you click the home
icon, it is apparent the document is referring to a driver and s/w for
the Microsof
Scott Stingel wrote:
A little more research on this:
I found a Dialogic flow diagram that seems to indicate what happens when
glare occurs on an IDSN line. So it looks like perhaps it can occur?
Re-phrasing my original question: "Does the asterisk PRI driver properly
re-try an outgoing call that
very very sure. pri's don't experience glare. channelized t1's do
(they are actually CAS lines, or carrier associated signalling)
pri's use out of band signalling (the D channel,which is usually the
24th channel)
Q.921 and Q.931 handle layer 2 and 3 call setup and tear down. Q.931
would not all
A little more research on this:
I found a Dialogic flow diagram that seems to indicate what happens when
glare occurs on an IDSN line. So it looks like perhaps it can occur?
Re-phrasing my original question: "Does the asterisk PRI driver properly
re-try an outgoing call that is dropped due to a
> PRIs do not experience glare.
Never say Never... pigs can fly too!
bkw
PS: Pack a pig in a box and next day fedex it across the country... they
fly! :P
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asteris
Bruce Komito wrote:
I have a PRI with a bunch of DID numbers on it. When I dial one of the
DID numbers from the outside, the call is correctly routed, either to the
auto-attendant or to the correct extension. However, all the caller hears
until the call is answered is silence, i.e., no ringing.
Scott Stingel wrote:
Hi-
I have an upcoming application that requires use of PRI channels that are
primarily used for high-volume incoming traffic, but that are to be used for
outbound calling as well. Of course, one option is to have dedicated
outbound channels reserved, but this is an inefficien
No and Yes, Olle. But mostly NO.
What Asterisk is doing actually depends on how it is configured. If you
are, by design, accepting calls for a particular [user] through the
default context from the general section in sip.conf it will generate
the correct response, but this is not because aster
On Tue, 25 May 2004 15:47:12 -0400, Mag Gam wrote:
>Hello:
>
>I am new to the list. I am trying to set up asterisk with voicepulse. I
>have a voicepulse username + password, and SIP DID. When I login to
>voicepulse, I have this under my devices tab:
>
>Devices
>
>*Login:* Sysxxx
>*Password
> Its possible to get glare... but doesnt outbound start at
> channel 1 and inbound start at channel 24? Or something
> like that?
A lot of telcos and PBX installers configure it so that incoming calls hunt
sequentially from channel 1 upwards, and the PBX makes outgoing calls on
channel 23 (on a P
> What about the scenario where the CO switch offers you a call at the same
> time you try to set up an outbound call on the same channel?
Wouldn't BOTH requests fail then if it's a two-way negotiation?
Switch: I'd like to send a call to you on B17
CPE: I'd like to place a call on B17
Switch: NAK
> I upgraded to the latest HEAD version of asterisk, and all IAX calls started
> sounding choppy. It was suggested on the IRC channel that I go back to
> asterisk -stable to determine if that fixes it. Is downgrading as simple as
> upgrading? Because now, -stable builds fine, but I get an error
Its possible to get glare... but doesnt outbound start at channel 1 and
inbound start at channel 24? Or something like that?
bkw
- Original Message -
From: "Paul Crick" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 25, 2004 6:32 PM
Subject: RE: [Asterisk-Users] "Glare"
> You are correct... No glare on a PRI
Are you sure?
What about the scenario where the CO switch offers you a call at the same
time you try to set up an outbound call on the same channel?
We dealt with this at my previous company (developing with VOS/CT-ADE and
Dialogic hardware connected to PRIs
CAN SOMEONE PLEASE POST THIS CONVERT IN A HOWTO OR FAQ
This works
Small twist I had to use signed loads so it was P0S353 is the OS79XX.txt and
the SIPDEFAULT.cnf has the current load that I wanted to go to.
Every other Doc or person that replied there howto never worked it may have
been th
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of lists
> Sent: Wednesday, 26 May 2004 7:38 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] 79XX converting
>
> OK I just tried that with the pos030201 load still no go
>
> Should I try go
On Tue, 2004-05-25 at 14:06, Steven Critchfield wrote:
> Most devices you will be using will be in some multiple of 24 as that is
> the number of channels in a T1. An Adit 600 will allow you up to 48
> channels as it is capable of handling 2 T1s on the back and 6 x 8port
> cards. So your 100 phones
You also need a SIPDefault.cnf
Simon
A prose style is a metaphysics.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lists
Sent: Wednesday, 26 May 2004 9:26
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 79XX converting
Humm that SCCP to start so
The correct spelling of the string is BELLCORE - derived from the bellcore
telephony specifications.
-Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Waddington
Sent: Monday, May 24, 2004 4:59 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Humm that SCCP to start sorry
I went up to a signed load on the sccp NP using my CCM's but I can't get
the phone to load a SIP load. I am currently trying 7.1 as per cisco's
paper
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guide
s09186a008022a968.html#wp1048832
Still
remove the .so because it needs changes that aren't in the version you are
trying to downgrade to.
bkw
- Original Message -
From: "Nik Martin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 25, 2004 4:32 PM
Subject: RE: [Asterisk-Users] Downgrading Asterisk
> That's the
John Fraizer wrote:
It looks like your cellphone carrier is actually "answering" the call
before they ring your phone. In their switch, they probably have the
equiv of:
exten => your.cellphone.number,1,Answer()
exten => your.cellphone.number,2,Ringing
exten => your.cellphone.number,3,Dial(CELL/
Thanks everyone for your responses. While these tips and tricks did
infact help get asterisk compiled with the fax modules, it seems that *
still craps out on the app_dtmftotext.c when you first start it. I
can't seem to find a way to get rid of it. I'm not even totally sure
it's required to sen
I have recently installed a test Asterisk server but am having problems
getting this to work with a Vigor 2600V. I have no issues using X-Lite to
connect to Asterisk but if I configure the 2600V, it registers correctly but
I don't get any sound at all although calls do seem to connect. The 2600V
Hi,
I am unable to get any music or sounds played with the
MusicOnHold or SayDigits commands. I do get sound from
the Playback and Background commands.
I have gone through the process of installing mpg123
and putting the link in usr/bin (and usr/local/bin).
For the MusicOnHold command I can see t
I have a PRI with a bunch of DID numbers on it. When I dial one of the
DID numbers from the outside, the call is correctly routed, either to the
auto-attendant or to the correct extension. However, all the caller hears
until the call is answered is silence, i.e., no ringing. That's not so
bad wi
That's the exact error I'm getting on Slackware 9.1, 2.4.26. Any one gat an
idea?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of jo
> Sent: Tuesday, May 25, 2004 4:35 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Downgrading Asterisk
You are correct... No glare on a PRI
W. Kevin Hunt
CCIE #11841
www.huntbrothers.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Tuesday, May 25, 2004 3:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] "Glare" condit
I've setup an asterisk server H.323 compliant, with a GnuGk gatekeeper.
My h323.conf is:
[General]
port=1720
gatekeeper=127.0.0.1
context=default
[PABX]
type=H323
e164=PABX
prefix=0
context=default
my gatekeeper.ini contain:
[RasSrv::GWPrefixes]
PABX=0
When i call, from a gatekeeper registere
Hi Stefan,
that's been the solution. Thanks a lot!
(I still wonder why I was able to compile against the stable 1 without
this patch.)
jo
Stefan Tichy wrote:
On Tue, May 25, 2004 at 07:40:51PM +0200, jo wrote:
Since upgrading from stable to latest cvs I can't place CAPI calls (AVM
Fritz/chan
Or just add /usr/local/lib to your /etc/ld.so.conf file.
-wade
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus
Darilion
Sent: Tuesday, May 25, 2004 1:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] spandsp hylafax asterisk and confusion
Ok.
I have downloaded the latest CVS and lo and behold theres the new SMS stuff
so I answered my own question.
Thanks me!
Still not sure how to edit my extensions.conf.
But I'll probably post back an answer to myself soon.
Sorry for being such a dumbell.
I want to implement SMS.
Do I need to dow
OK I just tried that with the pos030201 load still no go
Should I try going up on the MGCP then down to pos30201 and work my way up?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer
Sent: Tuesday, May 25, 2004 3:29 PM
To: [EMAIL PROTECTED]
Sub
Sorry, no solution but same problem. Downgrading brings this message on
Suse9.0, 2.4.21:
[app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so:
undefined symbol: ast_get_txt
May 25 23:28:42 WARNING[16384]: loader.c:408 loa
Hi Scott,
SS> Normally PBX's are designed to have the CPE yield to an incoming
SS> call if a particular channel is seized by both ends at the same
SS> time (a condition known as "glare"), but I'm wondering if anyone
SS> has real-world experience with asterisk to say how well this is
SS> handled.
If you know the date when things worked in cvs head, I thought you could
just (using XX April as an example):
cvs update -D "XX April 2004" asterisk zaptel libpri (and...asterisk-addons,
if used)
Then make clean; make install in each directory.
Cheers
Scott
Scott M. Stingel
President,
Emergin
I have installed two HFC PCI A-cards running zaphfc from bristuff-0.0.2,
which seems to work quite fine, but I continously receive the messages
"D-Channel on span 1 up" followed by "D-Channel on span 1 down" with a few
seconds interval. Why is that? Bri intense debug log and configuration
files bel
On Tue, 2004-05-25 at 14:53, Jeff Gustafson wrote:
> On Mon, 2004-05-24 at 19:33, Paul Mahler wrote:
> > I have had good experiences with Adit. Their customer service and
> > documentation are excellent.
>
> Sounds good. So they have chassis that can handle >= 100 analog
> phones? I looke
Not sure on the kernel but My voicepulse worked 1st try by following these
instructions:
http://www.voicepulse.com/kb
The following settings are for VoicePulse Connect! users only, not
VoicePulse Broadband Phone Service users. Please note that these examples
assume you are setting up Asterisk
Hi Nik
Nik Martin wrote:
I upgraded to the latest HEAD version of asterisk, and all IAX calls started
sounding choppy. It was suggested on the IRC channel that I go back to
asterisk -stable to determine if that fixes it. Is downgrading as simple as
upgrading? Because now, -stable builds fine, bu
Nick Grindley wrote:
Hi All,
When receiving or making a call about 60 seconds or so into the call we
develop choppy/stutter audio problems. It then seems to clear itself only to
return again, and so the pattern carries on! This has got me stumped!
I can confirm similar behavior on the IpDialog Si
Amaury Jacquot wrote:
Tony Hoyle wrote:
Guess what... BT actually charge a monthly rental fee (£1.50 or ~$2.40
per month) for providing this information. Gotta love state
monopolies hmm? Hope they don't take too long provisioning it...
heh, they'll probably mess up some other setting on that li
Hello!
How can one select outgoing MSN when dialing out from ttyI-interfaces?
I have successfully done this with CAPI e.g...
exten => _X.,2,Dial,CAPI/60:bBYEXTENSION
...in extensions.conf.
Currently correponding for my ISDN modem interface is...
exten => _X.,2,Dial(Modem/g1:${EXTEN})
On Tue, May 25, 2004 at 07:40:51PM +0200, jo wrote:
> Since upgrading from stable to latest cvs I can't place CAPI calls (AVM
> Fritz/chan_capi-0.3.1)
> Did I miss something that has to be changed in configfiles?
> Also tried to recompile chan_capi which run into an error.
Did you apply the patc
lists wrote:
I have a done google seaches on convertion and so far they all failed.
Rich adamson and wheely-bin.co.uk
Here is what I have
Laptop running solarwinds tftp with the following files
OS79XX.txt <- POS30201
SIP.cnf.xml
That should be SIP.cnf
John
_
On Tue, 2004-05-25 at 13:53, Scott Stingel wrote:
> Hi-
>
> I have an upcoming application that requires use of PRI channels that are
> primarily used for high-volume incoming traffic, but that are to be used for
> outbound calling as well. Of course, one option is to have dedicated
> outbound ch
Hey Markus
> The question now is: how do I tell Asterisk to send
> everything starting with 011 as "unknown numbering plan"?
You can use the "pridialplan=unknown" option in zapata.conf but that will
then apply to everything.. which should be fine.. but I'm not sure about 11
digit dialing for local
HAHA so true.. I have had to fire customers before.. It felt great!
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of David Boreham
> Sent: Tuesday, May 25, 2004 2:22 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Nufon
Sorry for being such a dumbell.
I want to implement SMS.
Do I need to download the latest CVS to get SMS functionality?
I only have one asterisk box and it's in production. A bit risky seeing as I
know feck all. I am kaking myself that when I get the latest code it will
all go pear shaped.
who d
Rana Dutt wrote:
Let's say you have a 256 Kbps Internet connection and you're using it for
voice calls. With mu-law (G.711), each call uses about 80 kbps, so you
really can't have more than 3 calls active at one time. Does Asterisk
support any kind of Call Admission Control where it would prevent y
On Mon, 2004-05-24 at 19:33, Paul Mahler wrote:
> I have had good experiences with Adit. Their customer service and
> documentation are excellent.
Sounds good. So they have chassis that can handle >= 100 analog
phones? I looked at something called the Adit 600. But I wasn't sure I
coul
I upgraded to the latest HEAD version of asterisk, and all IAX calls started
sounding choppy. It was suggested on the IRC channel that I go back to
asterisk -stable to determine if that fixes it. Is downgrading as simple as
upgrading? Because now, -stable builds fine, but I get an error on the
a
Hello:
I am new to the list. I am trying to set up asterisk with voicepulse. I
have a voicepulse username + password, and SIP DID. When I login to
voicepulse, I have this under my devices tab:
Devices
*Login:* Sysxxx
*Password:* xx
*Context:* VPWS
*Connects to:* gw5.voicepulse.com
W. Kevin Hunt wrote:
I'll add $1k to that bounty, and will put another bounty out for $3k for
ss7 integration w/ full isup / imt support...
John Bittner wrote:
I am also looking for the SMDI support. I am willing to put up a bounty
of 2K to get this writen. Anyone interested please email me off lis
Yes this can happen on phones with older firmware (in my experience). Cisco
has a few references to problems with the filenames being longer then
eight(?) characters, although making them shorter did not help me. What I
have done is flashed the phone with a newer skinny image (regular cisco
image
> Jeremy, you are only making the problem worse when you reply here quickly
yet
I'd like to advance a contrary view. I particularly enjoy the
Nufone threads here. It's rare indeed that one can see reponses
from the horse's mouth so to speak on this kind of subject,
presented without the veil of su
Karl Brose wrote:
If the response to an OPTIONS is generated by a proxy server, the
proxy returns a 200 (OK), listing the capabilities of the server.
The response does not contain a message body.
Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
fields SHOULD be presen
That may be the case in Australia, but at least here in the US of A, the
telco accepts what is sent. I only have it set up to spoof on prefix 8
to call friends, but they already know that if they see their number,
odds are pretty good that it is me. :-)
The main "legit" way that is used, is when
Andrew Kohlsmith wrote:
In short, you're making this problem worse. Answer the damn support emails
quickly and people won't see the need to post here. I get the "we got your
support question, your ticket # is .." email quickly but then it tends to
languish for a while. I've only had a few sup
> On Tue, 25 May 2004, Bartosz Jozwiak wrote:
>
> >Hello,
> >
> >I have just received Adtran TSU 600 with 24 FXS ports.
> >I have installed sucessfuly T100P card.
>
> Sucessfully?
> Did you load the module for the card?
Yes
> What does 'ztcfg -v' show?
ast05:~# ztcfg -vvv
Zaptel Configuration
=
> We do.. I personally responded to him in less than one hour on a friggin
> SUNDAY. We are not superhero's here.
I don't know the details of his particular interaction with you; I am speaking
from my own experiences. Most times (Yes most, which is why I really don't
have much of a beef with
> In short, you're making this problem worse. Answer the damn support emails
> quickly and people won't see the need to post here. I get the "we got your
> support question, your ticket # is .." email quickly but then it tends to
> languish for a while. I've only had a few support questions th
Hi-
I have an upcoming application that requires use of PRI channels that are
primarily used for high-volume incoming traffic, but that are to be used for
outbound calling as well. Of course, one option is to have dedicated
outbound channels reserved, but this is an inefficient use of channel
res
Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They
advertise "no new software features", but it does include bugfixes for a
number of things. I know there was a discussion about the 0.4sec delay,
which is said to be resolved in this firmware (CSCed48311: Media takes 0.4
sec to
Just send it all as unknown.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Markus Mayer
> Sent: Tuesday, May 25, 2004 12:57 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Telus: Overseas calling
>
> Hi,
>
> We ran into
for those who want to patch their SIP, here is a quck fix to make
Asterisk do a little better:
--- chan_sip.c 2004-05-16 01:33:06.0 -0400
+++ chan_sip.c_OPTIONS 2004-05-17 14:30:36.0 -0400
@@ -5916,6 +5916,7 @@
/* Initialize the context if it hasn't been already */
I have a done google seaches on convertion and so far they all failed.
Rich adamson and wheely-bin.co.uk
Here is what I have
Laptop running solarwinds tftp with the following files
OS79XX.txt <- POS30201
SIP.cnf.xml
RINGLIST.DAT <-Ringer1.pcm
Ringer1.pcm
The phone (7940) is hardcoded to
On Tue, 25 May 2004, Bartosz Jozwiak wrote:
>Hello,
>
>I have just received Adtran TSU 600 with 24 FXS ports.
>I have installed sucessfuly T100P card.
Sucessfully?
Did you load the module for the card?
What does 'ztcfg -v' show?
Is asterisk running? Does asterisk see the ports? (zap show channel
Hello
Can somebody send me please config files of zaptel.conf and zapata.conf for
adtran fxs ports.
I cannot make it work.
I do not get a dial tone on Adtran and when I am trying to call from sip i
get:
app_dial.c:674 dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy at this
Jeremy Hall wrote:
If by authentication by mobile number you mean the caller ID received,
that is not secure at all. CallerID is very easy to spoof when you have
a digital line (certain types, of course.) For example, when I call out
from my Asterisk box, if I prefix the number with 9, it sends
RFC 3261 states:
11.2 Processing of OPTIONS Request
The response to an OPTIONS is constructed using the standard rules
for a SIP response as discussed in Section 8.2.6. The response code
chosen MUST be the same that would have been chosen had the request
been an INVITE. That is, a 200 (
Andrew Kohlsmith wrote:
Answer the damn support emails
quickly and people won't see the need to post here.
We do.. I personally responded to him in less than one hour on a friggin
SUNDAY. We are not superhero's here.
Jeremy McNamara
___
Asterisk-
Hi there.
Graham Turner wrote:
was wondering if anyone could give us a run through an explanation of the
wiki and other examples of connecting to iptel's sip express router using
asterisk pbx so i can understand better the call processing ..
given the example i work from on john todd's www.loligo.
Let's say you have a 256 Kbps Internet connection and you're using it for
voice calls. With mu-law (G.711), each call uses about 80 kbps, so you
really can't have more than 3 calls active at one time. Does Asterisk
support any kind of Call Admission Control where it would prevent you from
originati
Hi,
We ran into a little problem recently with our phone provider (Telus
Canada): we are unable to dial numbers outside North America.
This is what happens: the phone number 011... is sent out over our
T1, Telus sees the correct number on their switch. However the
switch thinks it's a North Ame
Kevin wrote:
Please do us a favor and respond to the support emails so this can be
addressed in the proper forum.
I did respond to you:
Ticket number: 1558
Date: Sun May 23 20:41:42 2004
Jeremy McNamara
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Asterisk-Users mailing list
[EMAIL PROTECTED]
h
Hi All,
Fairly new at Asterisk ... looking for some help getting a new Asterisk PBX
to play nice with an old Lucent Partner ACS system. If anyone has done
this before and is in a helpful mood please pop me an email:
rob[at]coastside.net
Thanks,
Rob
_
On Tue, 25 May 2004, Klaus Darilion wrote:
>
>
> Brian D'Arcy wrote:
>
>
> > ast_load_resource: libspandsp.so.0: cannot open shared object file: No
> > such file or directory
>
> I copied the libspan* files from /usr/local/lib to /usr/lib and then
> asterisk started!
Do you have /usr/loca
Since upgrading from stable to latest cvs I can't place CAPI calls (AVM
Fritz/chan_capi-0.3.1)
Did I miss something that has to be changed in configfiles?
Also tried to recompile chan_capi which run into an error.
capi info shows me:
Contr1: 2 B channels total, 2 B channels free.
Any suggestions
> As Brain has suggested, this conversation should not have to be on this
> forum. I know of others who are having this same issue with Nufone and
> were having difficulties in getting you to provide support or a
> response. I resorted to the Asterisk community for suggestions. I notice
> you are
> Would it be insane to recommend plugging analog handsets into those
> devices that convert the analog extension into either a SIP or IAX phone?
I don't see why that would be insane, it just seems that they're signficantly
more expensive and harder to maintain than a cheap * box and channel bank
> Or... What about having something similar to the tone plans in
> indications.conf that would allow someone to either choose one of
> several canned Vertical Service Code plans or roll their own?
A+ YES BABY PLEASE THAT WOULD KICK SOME SERIOUS ASS
Hardcoded values = teh suck. Full stop. End o
Well I am getting the phones to ring but have no voice. When someone
dials an IP number does this circumvent the * server? I was trying to
make a capture of the call with ethereal but saw no traffic at the server
for the call. Unfortunatly I have no way to set the dtmfmode on the phone
side so I
Good afternoon,
I haven't set up a pre-paid system myself, so I can't answer to details
on what system to use, etc. But I can give you some advise regarding
your authentication scheme.
If by authentication by mobile number you mean the caller ID received,
that is not secure at all. CallerID is
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