I have a new 7940
I have set-up the network
And tried to tftp SIP ver. 2.1
And ever time it boots and starts the tftp download the 7940 reboots
Any input welcome
Regards Mick
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At 18:45 8-10-2003 +0200, you wrote:
Hi capi users :-)
you might also want to try chan_capi 0.3.0 which is already
in the downloads directory but not linked on the page.
The option echosquelch=1 now finally works.
Yeah, I found 0.3.0 recently and installed it, seems to be working fine.
What is
hi guys,
i'm using sept 30 cvs and oh323 5.5
i'm having 5 second latecy(on only 1 audio path)
when a call is transferred
the scenario is this:
sip-asterisk-h323:operator (who
then transfers the call)
h323:destination
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An advanced graphical user interface
2) An IAX2 hardware device
3) A Radius CDR report module
4) A live-method, robust SQL-based dialplan
5) LDAP/SQL/Radius authentication for SIP phones
6) Robust R2 signalling support
7)
I am guessing you are running without reinvite's, I'm running with reinvite's with
latest CVS release and 79x0 phones without any issues with conferencing...
-Original Message-
From: Adam Rothschild [mailto:[EMAIL PROTECTED]
Sent: 08 October 2003 15:49
To: [EMAIL PROTECTED]
John Todd wrote:
I was wondering if anyone else has had this problem. I have
purchused several Cisco 7940 and 7960 phones. Of the 5 phones so
far I have run accross 2 that that give me malformed TFTP and refuse
to upgrade to the latest version of SIP code -- 5.3. In fact some
of the other
All I get is
Version Error
When trying to tftp
Any ideas ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, 9 October 2003 5:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem
wan't to add DS3 and SS7 to that ?
also licensed g723.1 and working g729
softfax and softmodem
that's what comes to my mind on the spot ...
On Thursday 09 October 2003 9:51 am, John Todd wrote:
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An advanced graphical user interface
Hello All,
Is it possible to make asterisk to do authetication
of IAX client through database (mysql, etc) instead of creating all the client
username in iax.conf?
How hard is to implementthe feature i
describe above?
We plan to use IAX as part of our VOIP
infrastructure mainly because
Had those same problems with some 7960's but not with others. As
previously mentioned (below and by others on the list over the last
year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x
and somewhere around v4.x remove all the comment lines in SIPDefault.
It will work. The problem
Well I eventually got the 7940 loaded
Now does anyone have quick fix to get it to work with asterisk
Tar in advance
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Thursday, 9 October 2003 9:41 PM
To: [EMAIL
Hi,
echosquelch=1 enables Petr Michalek's echo canceler, which
compares RX and TX volumes and mutes the RX in an echo condition.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
wan't to add DS3 and SS7 to that ?
I dunno; I've provisioned at least a half dozen DS3s and physically seen one
SS7... :-)
Regards,
Andrew
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I have just (last month) brought up * with ISDN soft phones.
I am using
Workstation-SIP or iax-*-isdn4linux-hisax-EICON Diva ISDN (not
Pro)-uk(bt)isdn lines
I am currently trying SIP clients - the last is an evaluation of SJphone,
but this problem does not seem to depend on the Workstation end
Hi all,
When i receive a call from pstn ( calls from sip works well) my phone shows
asterisk and not the number of the phone.
How can i make asterisk show the phone number of the person who caled?
thanks!
Miklos
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yes, you're right, i tried to put a ast_cdr_answer when queue makes the
ast_cdr_setdstchan, but it didn't work, or maybe, apps like backgound and
playback should not issue an ast_cdr_answer, so it is ussued only when the
answer is actually answered by someone.
is you used the queue app, don't
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An advanced graphical user interface
2) An IAX2 hardware device
3) A Radius CDR report module
4) A live-method, robust SQL-based dialplan
5) LDAP/SQL/Radius authentication for SIP phones
6) Robust R2 signalling support
7)
On Thu, 2003-10-09 at 05:14, Klaus-Peter Junghanns wrote:
dont forget generic voice modem support. or even better
chan_modoss or chan_modalsa the combination of an
external modem (for signalling) and a sound card :)
This sounds like your wish list, not something anyone said they would
work on.
Is there any way that you could trigger events based upon actual pickup of
calls and hangups of lines in ALL cases(parked calls, queued calls, calls
triggerd by .call queue files)?
It seems like Asterisk needs something a little lower level to allow for
this, is it even possible?
MATT---
really, for that, CDR needs to be rewritten in some parts, cause one thing
you could use is to know the full path of a call, based on an identifier
or something, so you can now the cal last 10 seconds on the prompt, 15
seconds on a queue, the 20 seconds talking, after that was parked for 10
Rich,
Thank you for your response. I have tried to do that. Unfortunately the oldes
version of code available on the Cisco site is 3.2. The current code rev on the phone
is 3.3 MGCP. Unfortunately I get the same results.
I start with version 3.2 - Did not work, then tried 4.4 and that did
How do you transfer the call?
Michael.
Kelvin Chua wrote:
hi guys,
i'm using sept 30 cvs and oh323 5.5
i'm having 5 second latecy(on only 1 audio path) when a call is
transferred
the scenario is this:
sip-asterisk-h323:operator (who then transfers the call)
Mick,
Can you please provide more detail on specifically what you did / or did not do to get
it to work.
Thanks
Babak
[EMAIL PROTECTED] wrote:
Well I eventually got the 7940 loaded
Now does anyone have quick fix to get it to work with asterisk
Tar in advance
Regards Mick
This had to do with a revision of request_and_dial, where the real bug
lives.
It's fixed in CVS now and the hack mentioned here should no longer need to
be applied.
Mark
On Wed, 8 Oct 2003, Richard Lyman wrote:
same issue as previously noted...
look at lines 1628ish in chan_iax.c and line
TC wrote:
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An advanced graphical user interface
2) An IAX2 hardware device
3) A Radius CDR report module
4) A live-method, robust SQL-based dialplan
5) LDAP/SQL/Radius authentication for SIP phones
6) Robust R2 signalling support
On Thu, 9 Oct 2003, Mark Spencer wrote:
This had to do with a revision of request_and_dial, where the real bug
lives.
It's fixed in CVS now and the hack mentioned here should no longer need to
be applied.
Mark
On Wed, 8 Oct 2003, Richard Lyman wrote:
same issue as previously
Hi,
Where are the settings to access the demo server at Digium? I would like to setup and
test x-lite as well with a running asterisk until i get my box up and running.
Thanks
-- Original Message --
From: Chris Albertson [EMAIL PROTECTED]
Reply-To:
After about three hours
I just TFTP to 7940
I had that weird file issue
So renamed the file
Using Cisco tftp unticked the box that says transfer this file only
And after 50 or so attempts there you go
Honestly I would rather load an IOS on our big mother routers ( if you
know what I mean
Im having problems setting up a trunk between two locations. Heres the
setup I have:
Server A is connected to the PSTN at my datacenter
Server B is connected to a clients e1 line at his datacenter
I only want to route calls from Server B to Server A and out through the
PSTN. Server A has a
So whats the best way to find the maximum number of concurrent calls in
this setup:
IAX2 Trunk using GSM over a 512k internet line.
thanks
duncan
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show channels?
On Thu, 9 Oct 2003, duncan wrote:
So whats the best way to find the maximum number of concurrent calls in
this setup:
IAX2 Trunk using GSM over a 512k internet line.
thanks
duncan
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[EMAIL
So whats the best way to find the maximum number of concurrent calls in
this setup:
IAX2 Trunk using GSM over a 512k internet line.
show channels?
actually i meant how to find out how many i could push down the 512k line -
with regards to codec bandwidth and signalling etc...
duncan
-Original Message-
From: Babak Pasdar [mailto:[EMAIL PROTECTED]
I was wondering if anyone else has had this problem. I have purchused
several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run
accross 2 that that give me malformed TFTP and refuse to upgrade to
the
hmm, i'd gone back thru ...request_and_dial to get to it... weird
that i missed a simplier fix G
Mark Spencer wrote:
This had to do with a revision of request_and_dial, where the real bug
lives.
It's fixed in CVS now and the hack mentioned here should no longer need to
be applied.
Mark
Hi all,
I do have the same problem.
Does this problem appear with the last versions of TDM's board (TDMx0B)?
I have seen a bug in feedback state (see
http://bugs.digium.com/bug_view_page.php?bug_id=087), but the
description it's not equal.
I work with:
Linux:Debian woody
try to read this:
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
senad
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duncan wrote:
actually i meant how to find out how many i could push down the 512k
line - with regards to codec bandwidth and signalling etc...
Measure the data rate on one call and divide 512k by it for a rough
estimate..
If you want more accuracy make one call and measure the data rate,
Rafael Gonzalez Lomeña wrote:
Hi all,
I do have the same problem.
Does this problem appear with the last versions of TDM's board (TDMx0B)?
I have seen a bug in feedback state (see
http://bugs.digium.com/bug_view_page.php?bug_id=087), but the
description it's not equal.
I work with:
On Thu, 9 Oct 2003, WipeOut wrote:
actually i meant how to find out how many i could push down the 512k
line - with regards to codec bandwidth and signalling etc...
Measure the data rate on one call and divide 512k by it for a rough
estimate..
If you want more accuracy make one call
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for trunking to work the type has to be peer..
When I set mine up I did so using type=friend just to make it simple..
So
WipeOut wrote:
duncan wrote:
actually i meant how to find out how many i could push down the 512k
line - with regards to codec bandwidth and signalling etc...
Measure the data rate on one call and divide 512k by it for a rough
estimate..
If you want more accuracy make one call and measure
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for trunking to work the type has to be
peer.. When I set mine up I did so using type=friend just to make it
simple.. So
Hi Jeremy,
The handbook says:
user: A user can place calls to or through the Asterisk server.
peer: A peer receives calls from the Asterisk server, but does not
place them
friend: A friend both sends and receives calls through the Asterisk
server. This makes the most sense for handsets or
Jeremy McNamara wrote:
A friend is both a user and peer. However, I would discurage the use
of a friend as it will severely restrict your dialplan, espcially once
you are dealing with more than just a couple Asterisk boxes.
Jeremy, Can you elaborate on how using type=friend would restrict the
John Todd wrote:
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for trunking to work the type has to be
peer.. When I set mine up I did so using type=friend just to
TeleSIP wrote:
Hi Jeremy,
The handbook says:
user: A user can place calls to or through the Asterisk server.
peer: A peer receives calls from the Asterisk server, but does not
place them
friend: A friend both sends and receives calls through the Asterisk
server. This makes the most sense for
tcpdump is the easiest way. From 1 call to 50 calls the number of packets
should be about the same, and they should just get larger.
Mark
On Thu, 9 Oct 2003, Jared Smith wrote:
On Thu, 2003-10-09 at 11:39, WipeOut wrote:
[snip]
He states that in order for trunking to work the type has to
I am looking into the possibility of buying a Cisco
7960 with a 7914 expansion module. I know a lot of
people are using the 7960, but I haven't read much
about the 7914 and I was wondering if anybody has used
this module with Asterisk?
-- Thank you for your time
I've been told that the SIP firmware cannot deal with the 7914, however
I've never been able to try it for myself as the few 7914s I have laying
around here have no interface cable and I am unable to find the pinout.
Even TAC couldn't help me :(
Jeremy McNamara
jerk face wrote:
I am
WipeOut wrote:
Jeremy, Can you elaborate on how using type=friend would restrict the
dialplan.. Just so I am aware of the pitfalls.. :)
Mark's words to me, when I was a newbie:
[00:08] kram a user is to authenticate an incoming call
[00:08] kram a peer is someone you send a call to
[00:08] kram
canreinvite=no in the appropriate sip.conf user or peer.
Jeremy McNamara
Anton Tinchev wrote:
I have incoming calls from cisco AS5350 that are placed in queue.
Queue rings on agents with SIP phones, and native bridge cousing some problems(no call
at all).
When i go to queue from iax client
and the peer simply has the required information:
[NuFone]
type=peer
secret=his_secret
context=NANPA
host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...? Aren't
those used
only for inbound anyhow?
Thorsten
Jeremy McNamara wrote:
WipeOut wrote:
Jeremy, Can you elaborate on how using type=friend would restrict the
dialplan.. Just so I am aware of the pitfalls.. :)
Mark's words to me, when I was a newbie: [00:08] kram a user is to
authenticate an incoming call
[00:08] kram a peer is someone you
Here is my Configuration
PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186
When I call from the pstn to the ATA, the ATA rings but I don't hear
anything on the calling side until the call is picked up.
When I call from the ATA, everything seems to work fine.
When I bypassed ASTERISK,
If I do something like
exten = 1,1,Record(/someplace/somefile|gsm)
It does not record I end up getting
-- Executing Record(SIP/mlh-04d0, |gsm) in new stack
exten = 1,1,Record(filename|gsm)
it works great!
Is there anyway that I can set the path in the record app...if not is
there an easy
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote:
and the peer simply has the required information:
[NuFone]
type=peer
secret=his_secret
context=NANPA
host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...? Aren't
those used
Hello All,
Thanks to those that responded to my problem of compiling on SUSE 8.2. I
was not able to get the compile done so decided to put RedHat 9 on this
system. After getting a RedHat supported NIC and RedHat installed,
Asterisk compiled cleanly, one SIP phone is connected and voice mail
You can send a fake ring by using something like:
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r)
Assuming the ATA is in the sip.conf as [1234]
However, this does NOT solve the underlying problem.
On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote:
Here is my Configuration
PSTN - Cisco
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote:
and the peer simply has the required information:
[NuFone]
type=peer
secret=his_secret
context=NANPA
host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...?
Aren't
those
That does make a ringing sound, but any idea what's causing the problem?
Stephen
Subject: Re: [Asterisk-Users] No Ringing from PSTN
You can send a fake ring by using something like:
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r)
Assuming the ATA is in the sip.conf as [1234]
However, this
On Thu, 2003-10-09 at 15:51, George Lin wrote:
Hi list,
Can anyone suggest us what kind compression tool is best to compress a GSM
file.
And what kind compression ratio can be?
This is a hard message to write with out unleashing the flame thrower.
On this list it has been discussed many
According to cisco it's just a serial cable, can you just use a straight
through cable? It looks like a standard phone-headset type cable,
though shorter.
Nick
On Thu, Oct 09, 2003 at 12:57:50PM -0700, jerk face wrote:
Well that sucks.
What about using SCCP
--- Jeremy McNamara [EMAIL
We have a DMS100 that does not have PRI.
So we're using a channelized T1 using WU-LAW, ESF and B8ZS coming from the
DMS100 that's plugged into a Tormenta2 Quad T1 Card on my Asterisk Box
running Debian 3.01(woody) with Kernel 2.4.22.
The Link is up but according to the DMS100, Channel_1 goes
Record(/tmp/testing:gsm)
Thats what I use.. and it works.
bkw
On Thu, 9 Oct 2003, Lists wrote:
If I do something like
exten = 1,1,Record(/someplace/somefile|gsm)
It does not record I end up getting
-- Executing Record(SIP/mlh-04d0, |gsm) in new stack
exten = 1,1,Record(filename|gsm)
What hardware are you using to connect to the PSTN?
G
At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote:
Hi all,
When i receive a call from pstn ( calls from sip works well) my phone shows
asterisk and not the number of the phone.
How can i make asterisk show the phone number of the
Thanks Steve.
In fact, I am looking for a ZIP tool to zip a GSM file. currently I found
that winzip ONLY compress 10% of a WAV file.
I am wondering is there any good ZIP tool for a GSM file and or WAV file.
Thanks,
George Lin.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
When a call is placed connecting to the X100P, the volume of the call is
very low. I have played with the gain settings without many results.
Any suggestions?
Thanks,
Kevin
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Hi all,
I am trying to setup an E100P for use on Swisscom E1-PRI here in
Switzerland.
Swisscom seems to use Siemens hardware.
Here are my configs (cvs from a few hours ago) :
zaptel.conf
loadzone=fr ; tried de but got warning at modprobe
defaultzone=fr
span=1,1,0,cas,hdb3,crc4,yellow ; Seems
Sorry for asking for it, but it is nowhere documented.
There is no maches in the mailing list or the whole google.
I found it just in sources - conf parser of chan_iax.c.
Thanks
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On Wed, 2003-10-08 at 18:12, Eric Wieling wrote:
Does this work?
festival_client --tts_mode Do you want to play a game?
Yes. But since I dont have a soundcard in the box I use another tts
command. I quote mi first email:
Also I tested with festival_client executing the same
On Wed, 2003-10-08 at 18:12, Eric Wieling wrote:
Does this work?
festival_client --tts_mode Do you want to play a game?
To be more specific I tried this command.
festival_client --output jj.wav pp
where pp is a file with the following command:
Quoting Marcel Prisi [EMAIL PROTECTED]:
Swisscom seems to use Siemens hardware.
Here are my configs (cvs from a few hours ago) :
zaptel.conf
loadzone=fr ; tried de but got warning at modprobe
defaultzone=fr
span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens
On Thu, 2003-10-09 at 18:26, George Lin wrote:
Thanks Steve.
In fact, I am looking for a ZIP tool to zip a GSM file. currently I found
that winzip ONLY compress 10% of a WAV file.
I am wondering is there any good ZIP tool for a GSM file and or WAV file.
Don't expect to get much
What is the proper method to install/configure an X100P FXO
card?
On Thu, Oct 09, 2003 at 03:28:21PM -0400, Jeremy McNamara wrote:
I've been told that the SIP firmware cannot deal with the 7914, however
This is correct, I just tried it, and there's no support for the 7914
expansion module in the SIP image. All I got is steady read light on the
buttons.
I just talked with a friend that is a computer teacher at the local
collage. He heard about my experiments with asterisk and some grandstream
phones, and he wants to get a small setup going as a class project, which
will hopefully expand to cover the whole building. Right now if a teacher
Hi PJ-
I specialize in large volume IVR systems both here and in Europe. (please
see my web site Case Studies for more info)
If it's just a simple IVR with database, I can likely do the demo very
cheaply, to get a chance at the bigger job. Already have the AGI's and Perl
script to accomplish
Hi Tom-
Someone may have already answered you on this, but if not:
Didn't you receive a quick start sheet with your demo kit? It should cover
this. If not, what works for me (also running Red Hat 9.0) is to add the
following lines to the /etc/rc.d/rc.local file:
rmmod usb-uhci
modprobe
Thorsten Lockert wrote:
and the peer simply has the required information:
[NuFone]
type=peer
secret=his_secret
context=NANPA
host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...? Aren't
those used
only for inbound anyhow?
No, you actually
I set mine up like this
exten = 1234,2,Dial(sip/[EMAIL PROTECTED],20,r)
And everytime it rings I get
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp
line:872
And * falls over
This is with a voicetronix openline4 card
Any ideas ???
Regards Mick
-Original
Hi all,
I'm having a problem with * being very finicky about the length of
DTMF key-presses during menus, voicemail, etc. Basically, short (100
ms) tones are ignored, anything between 100ms (or so) and about 300ms
is correctly detected, and anything 300ms is interpreted as multiple
presses of
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 09, 2003 2:57 PM
Subject: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2
Trunking confirmation?)
WipeOut wrote:
Jeremy, Can you elaborate on how using
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