I am trying to send calls from an AS5300 to Asterisk via e1 and I get this
bit of information in place of routing information
Going to extension s|1 because of Complete received
Accepting call from '' to 's' on channel 1, span 1
Here are the relevant zaptel and zapata pieces.
Patrick,
doe a google search for ISDN over IP, maybe that's your solution.
jo
Patrick Stuckenberger wrote:
Hi list,
is it possible to create something like a ISDN-WAN-WAN-ISDN bridge?
We have to change our location, but our number and the telephone
system should shoulb stay the same.
Hi,
if there is somebody working on Czech support please contact me off list,
so we can work together.
Petr Mosnicka
--
YieldTech - linuxova divize ATAX Group, spol. s r.o.
V zavetri 6 tel: +420-777-2LINUX
170 00 Praha 7 mailto: [EMAIL PROTECTED]
Ceska
While I wish the guys at Stealth the best of luck, I'll say again
that ENUM is _NOT_ the solution for VoIP routing in the current real
world. See the mailing list archives for more of my rants on why DNS
is not the answer for cost-based routing (where cost is monetary,
distance, qos, or any
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5
I know it's exciting to get things working, however,
there are some things wrong with your configuration, despite it perhaps
working ok.
Is it really? You can make outbound calls this way?
In your friends definition (friend-sipgate) you don't have a host
specified.
host=sipgate.de
Without that
You may be quite right, I have read parts of the rfc at least, I remember,
but the lure of using cheap existing infrastructure is probably to great.
KHB
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 04, 2004 03:20
Subject: RE:
Dear List members,
I have this problem with Mediatrix 24-FXS-line gateway and out-of-band DTMF.
It seems not working - the RTP mode is not working and when I select INFO
mode, the Mediatrix is behaving just the same as with RTP mode.
Here is a bunch of logs to explain this:
1. The RTP
Hi!
I have 2 grandstream budgetone 100 series. I can call allright, but I
cant do call transfer, park and call conference. (all features works
with tdm devices) the
1. Check if Asterisk is always in the media path, i.e. you need the t or
T option (or something similar) in your Dial
The SDK for the Siemens USB cordless phone was just released
a few days ago. I understand from a few people I spoke with when this was first
released that this could be ported to work for Asterisk. Does anybody have any
thoughts now they have seen the sdk information?
Cheers,
Dean
Le lun 03/05/2004 à 18:48, Jeremy McNamara a écrit :
Actually its cuz chan_h323 sucks like that.
Correct me if I'm wrong, but I browsed the archives and I got the
feeling that you (Jeremy) were one of the main developers of the
chan_h323... aren't you a little harsh about your own work? :-)
John Todd wrote:
TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum,
it seems. If anyone here on the list has $100,000 to put together a
real programming effort towards getting that implemented, y'all let me
know. The longer this waits, the more lame and broken become the
Hi there,
I have serious problems with MGCP and Swissvoice ip10s, and it appears
that recent CVS also introduced trouble for other MGCP users. Please
check and add comments in the bugtracker so that we can get a clearer
picture - thanks! Also comment if things are working fine for you.
uhm, strange but does this work on your setup? even with permit and
deny, if a user is not matched in the conf, it is allowed access to the
default context stated in the conf.
On Wed, 2004-04-28 at 16:12, James H. Thompson wrote:
I think the problem is that using permit= alone does nothing.
On Sunday 02 May 2004 08:07 pm, Kevin Walsh wrote:
Someone on IRC once pointed out the conflict between suggested and
must on a similar page and said that their TDM400P (FXS-only at the
time) was working on a PCI 2.1 system. Can anyone confirm whether a
PCI 2.2 bus is mandatory?
Yes, PCI 2.2
On Sunday 02 May 2004 08:56 pm, Jamin W. Collins wrote:
On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote:
The same Digium shop page suggests that two PCI slots would be required.
I'll assume the card is too fat, with the daughter board(s) fitted, to
fit into a single slot.
This
On Wednesday 28 April 2004 03:37 pm, Barton Hodges wrote:
I would like to use Asterisk for voicemail, connected to a Toshiba
dk280. Has anyone done this with this model or similar system? Are
there any documents available that could give me some insight into how
I can do this?
You may want
Hi
I am new to this forum can some body tell me how can i
configure and implement agents.
if there is any document available on agents
implementation plz forward me that
thanx
Salman
__
Do you Yahoo!?
Win a $20,000 Career Makeover at
So a guy shows up at the the office, after making an
appointment with the office manager / receptionist to talk
'phone systems'.
After her eyes glaze over, with him talking T1 and
Frame-Relay I get to see him. He's from Norvergence. Well
dressed. Tells me they can do a T1 for $79, with
Found this:
http://w3.ripoffreport.com/reports/ripoff89155.htm
Many other nasty stories about them too.
--
Cheers,
Neil
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 04 May 2004 12:51
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] How does
Tilghman Lesher wrote:
On Monday 03 May 2004 13:56, Frank Mandarino wrote:
I have worked around this issue by storing the extension in a
variable, then restoring it using a Goto in the 'T' processing.
For example:
exten = 411,1,SetVar(ORIG_EXTEN=${EXTEN})
exten =
411,2,Dial(IAX2/[EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr =
On Thursday 29 April 2004 06:20 am, Andres wrote:
Sean Garland wrote:
I still have problems with beeps and clicks on all my calls. I have
polycom sip phones. I also can hear the beeps and clicks on some of my
messages, which would lead me to believe that it is more of a decoding
problem on
Title: Max TE410P card on an Asterisk
Hello,
Does
anybody know the max number of TE410P/TE405P card we can put in an asterisk
box?
Thanks.
Hi Philipp,
I havn't tried latest mgcp code but I can say that chan_mgcp has serious
problems with IP10S that are partially solved by my latest patch
http://lists.digium.com/pipermail/asterisk-users/2004-March/041615.html
I have received any feedback about it.
Regards,
Daniel ANDRE
Philipp
It depends entirely on the application: number of transcoders running etc.
There has been some discussion on this topic in the past - you might consult
the archives and the Wiki.
Assuming that you're running a fast processor (2.4GHz), I would think the
general answer is either one or two 4-port
On Friday 30 April 2004 11:48 am, Akshay Lamba wrote:
Hi Everyone,
Could someone direct me to a site that talks about Asterisk
implementation for Ethernet interfaces/SIP Implementation? I've done my
share of googleing and am only able to come up with sites that use
digium hardware only.
See:
Title: Max TE410P card on an Asterisk
Short
sarcastic answer: (just because I've seen this question 12 times in the last few
months!)
As
many as will fit on your motherboard, but don't expect to use them all
:)
Long
true answer:
2 quad
cards on a fast P4 system if you are doing very
Searched the
archives thoroughly...
Can't find this
specific problem...
Simple setup with
Asterisk on RedHat. No voice cards in the box, 2 SNOM 200
phones...
Phones seem to work
well, can leave VM, Message Waiting Indicator lights up but when I try to
retrieve messages the call
Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies!Neil Grant [EMAIL
I have an nec electra elite 192 with a t1 card; and am looking for
suggestions as to integrating them (can't throw out the system yet!).
I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a
hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart
signaling, and 2
I've been wondering what the difference is in the syntax of things,
like Dial.
Some examples show things like:
exten = 500,1,Dial,SIP/${EXTEN}|10
but other examples show:
exten = 500,1,Dial(SIP/${EXTEN}|10)
or
exten = 500,1,Dial(SIP/${EXTEN},10)
Which one is correct? Or most correct? Which one
I don't think your DTMF is set right look in sip.conf for the dtmf
directive for your phones.
cheers!
On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the
-Original Message-
From: Karl Brose
Sent: Tuesday, May 04, 2004 2:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Resolved: sipgate.de
You can make outbound calls this way? In your friends definition
(friend-sipgate)
you don't have a host specified. host=sipgate.de Without
On Tue, 2004-05-04 at 08:51, Tim Sailer wrote:
I've been wondering what the difference is in the syntax of things,
like Dial.
Some examples show things like:
exten = 500,1,Dial,SIP/${EXTEN}|10
but other examples show:
exten = 500,1,Dial(SIP/${EXTEN}|10)
or
exten =
Sorry, forgot to include that... Seems to be set right for the Snom phones
(from what I could gather).
[520]
type=friend
secret=blah
host=dynamic
callerid=Mike
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.0.12
mailbox=520 ; Mailbox for message
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
reacend wrote:
| Hi there, i have a working Microsoft ISA firewall with buildin
| H.323 Gatekeeper So Far, i got registerd the asterisk on the M$
| Gatekeeper...
|
|
| here is the h.323 configuration:
|
| ; Open H.323 driver configuration ;
I have a good friend who used to be a
sales rep for them. The entire sales pitch is based on making the customer believe
that they are lucky to have been offered the opportunity to beome a Norvergence
customer as they are extremely selective. If any technical question where to
come up, he
My customer is going to ask for a copy of Norvergence's contract to read
the details. He said he'd send me a copy when it arrives... From doing
a little goggleing it sounds like Norvergence is a scam business model
waiting to implode but what do I know. :)
Lance
Also, working this a bit more... if i do the echo
test (extension 600) i get sorta the same thing...
== Spawn extension (default, asterisk, 1) exited non-zero
on 'SIP/520-a25e'May 4 09:15:51 NOTICE[1125329600]: chan_sip.c:5655
handle_request: Unknown SIPcommand 'PUBLISH' from
Howdy,
I appologise in advance if this is not the correct forum for this message.
Bought an ACT Networks NetPerformer SDM-9350 Voice/Data router off ebay
('cause it was cheap and has 4 EM/FXO/FXS configurable ports.), now I need
to get it to work.
2 problems:
1) No documentation - I've searched
Yes, I would like to see the contract.Michael Miller [EMAIL PROTECTED] wrote:
I have a good friend who used to be a sales rep for them. The entire sales pitch is based on making the customer believe that they are lucky to have been offered the opportunity to beome a
hi all,
i'd like to know if it would be possible with asterisk (and which
hardware would i need) to implement the following (or is it not possible
with asterisk - but possible with ...)
I'd like to set up something like a Mobile to Conventionel Network
Gateway - so that users (with there Mobile
On Saturday 01 May 2004 09:42 pm, Tom Scott wrote:
okay, will use ${EXTEN}.
it all seems to be working now. I think my problem was
understanding the flow of control using contexts, but
i also needed to do some reading on syntax and variables
-- and more to come.
the working commands that
I will scan it when I get home tonight and
post the url to download it from.
Michael
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J Poz
Sent: Tuesday, May 04, 2004 10:51
AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] How
does Norvergence do it ?
On Monday 03 May 2004 01:08 pm, Rich Adamson wrote:
I've been running * for eight months in production mode without the
init.d/zaptel script in place. Didn't know 'make config' from within
the zaptel src directory even existed, and have never seen/heard anyone
even mention that before. Its
Hello all,
I have googled a bit, but was not able to a definite answer (maybe there is
not one..)
The question is, how different would be the voice qualitiy, if you let
translate * from alaw (PRI) to gsm instead of using alaw as codec for sip.
And also how would echo and the processor load be
This is possible with asterisk. There several ways you can do this.
You would need a X100P from Digium to interface with the PSTN line
coming in. Then you could send the call over VoIP which doesn't require
anything more than broadband and a VoIP provider. You should have
caller-id on the PSTN
Die GSM Tailnehmer wählen nicht die eigentlich Auslandsnummer - sonder
unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP
Gateway sollte dann die Durchwahl(=Auslandsnummer) wählen und das
Gespräch verbinden.
So dachte ich mir das auf jeden Fall - obs möglich ist weiß ich nicht
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the
incoming X100P cards.
Did you put immediate=yes in your zapata.conf? I had similar problems
previously (I have T100p instead of X100p) and it is fixed when I put
immediate=no.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: Tuesday, May 04, 2004
Folks,
It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there
another IAX2 client that is usable under Linux (Debian preferred)?
Thanks,
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box 726
Hello,
I have a T1 (not PRI) plugged into my Asterisk
server with a T100P card.
Everything is working well, except I only get the
first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon)
tried switching to 7 digits, and I only got the first digit of the
7.
Can anybody help?
Lisa
Thanks for that, worked a treat.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lisa Xie
Sent: 04 May 2004 17:33
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar
On Tue, 2004-05-04 at 11:52, Pat Boyle wrote:
-- zaptel.conf --
span=1,0,0,esf,b8zs
em=1-8
loadzone=us
defaultzone=us
-- extensions.conf --
; Need an extension to pick up calls from the T1 that uses em wink
; This comes in as a 6 instead of 4 full digits
; then pass to the s extension
I was told the X100P will have issues if installed on a line
with a DSL connection. Is there a card
that will work correctly on a DSL connection?
Thanks!!
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is used in the context
ParkandAnnounce.
Can
Hi to all,
I added a second isdn fritz card to my asterisk box to manage a second isdn
line.
But when I start capi it sees only one controller.
How I can enable the second isdn card.
Thank you
Bye
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
We utilize an X100P on a DSL line provisioned
by Verizon with no problems. Just
make sure you place the filters in the right place and you wont have any
problems.
- Brent
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Blackman
Sent:
On Tue, 2004-05-04 at 12:21, John Blackman wrote:
I was told the X100P will have issues if installed on a line with a
DSL connection. Is there a card that will work correctly on a DSL
connection?
You were told wrong. There are a FEW people that are having problems
with their X100P on a DSL
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to
know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
Yes I think so. But you have to download libr2 and compile it, if I am not
mistaken.
Bart
___
Thanks for the reply.
If I delete the "6" extension and leave the 6020
extension, asterisk won't answer it and I get the invalid extension message from
asterisk. I suspect that for some reason, the zaptel driver is only
passing forward "6" of the full four digits "6020."
Any thoughts on
i tried using music on hold option in the dial command
exten = ,1,Dial(zap/1/,20,m)
when someone calls me and i picked up the phone, the call will
be suddenly dropped. however, if i use a sip client instead of
zap (also changing the dial statement to sip), i can answer the
incoming call
I found similar posts regarding this error but none
that answered my question.
My zaptel.conf reads:
fxsks=1-2
fxoks=3
loadzone=us
defaultzone=us
and /proc/interrupts:
CPU0
0: 5542402 IO-APIC-edge timer
1: 2 IO-APIC-edge keyboard
2: 0 XT-PIC cascade
8: 3 IO-APIC-edge
If the new FXO doesn't have a filter built in then you will still have to
install a filter and might actually still have the same problem. I work
with DSL in large quantity on a daily basis... but that makes me far from an
expert! :P So I may be wrong...
bkw
-Original Message-
From:
What signaling are you using in /etc/asterisk/zapata.conf (em, em_w,
featd)?
When I use a DTMF based signaling, I can see the actual DTMF tones as
they are received in my 'full' log. Here is an example of what I see
(not real phone number) using a signaling type of 'featd':
Apr 30 17:02:46
Ahh, just like my momma told me, if it sounds too good to be true, it
usually is.. :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andre
Normandin
Sent: Monday, May 03, 2004 4:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] How does Novergence
Tim Sailer wrote:
Does anyone still have the 0.7.2 debs hanging around? I need to revert
a recent upgrade. We're having too may flaky problems (like softphones
being able to dial out fine, but GrandStreams failing to dial every
other time), and iaxcomm not working with gsm.
Why not diagnose
I second this.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jeremy McNamara
Sent: Tuesday, May 04, 2004 2:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 0.7.2 debs
Tim Sailer wrote:
Does anyone still have the
First thing you must is read next url
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
and if you hav done this, please attach your capi.conf.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de massimo
Enviado el: martes, 04 de mayo de 2004
On Tue, May 04, 2004 at 03:14:43PM -0400, Jeremy McNamara wrote:
Tim Sailer wrote:
Does anyone still have the 0.7.2 debs hanging around? I need to revert
a recent upgrade. We're having too may flaky problems (like softphones
being able to dial out fine, but GrandStreams failing to dial every
Is it possible to buy some kind of signalling converters from R2 to PRI ?
again.
please search the archives... this question
has been asked answered N*N*N^N times ...
no.
r2 support in asterisk in far from being complete
and it can do only 10% of the work.
you can try libr2 from the
Tim == Tim Petlock [EMAIL PROTECTED] writes:
Tim Be very careful about them. Search the archives of
Tim comp.dcom.telecom for details - focus on the last twelve months.
Ah, yes. I knew the name sounded familiar.
-JimC
___
Asterisk-Users mailing
On Tuesday 04 May 2004 11:37 am, salman khan wrote:
Hi
I am new to this forum can some body tell me how can i
configure and implement agents.
if there is any document available on agents
implementation plz forward me that
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Agents
What is the best free stun server out there? The one that I have looked at
from vovida requires two NICs. Is this neccessary?
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To UNSUBSCRIBE or
On Friday 30 April 2004 10:36 pm, CW_ASN wrote:
yes
- Original Message -
From: Steven Kalcevich [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 01, 2004 7:30 PM
Subject: [Asterisk-Users] Multiple music's on hold?
Hey there,
Is it possible to have multiple music on
AJ Grinnell wrote:
What is the best free stun server out there? The one that I have looked at
from vovida requires two NICs. Is this neccessary?
Asterisk does not require STUN.
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
STUN can be nice when connecting to Asterisk behind
NAT in some situations. X-Lite/Pro softphones,
Grandstream Budgetones and a few other clients make
great use of STUN.
That being said, the only good (free) STUN server I've
seen is the Vovida one that requires two NICs. It
works very well, if
I have a few SIP phones, Cisco 7960s, and was looking into implementing
some compression, ala G.729. I'm looking into purchasing a g729
licenses just to get an idea of performance and voice quality, over
lans, wireless and single channel isdn.
Does anyone have positive/negative experience w/
I have a few SIP phones, Cisco 7960s, and was looking into implementing
some compression, ala G.729. I'm looking into purchasing a g729
licenses just to get an idea of performance and voice quality, over
lans, wireless and single channel isdn.
Does anyone have positive/negative
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi some one can give me information about a good and ship ip phone IAX
or SIP
Thanks
- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33) 36309553
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Hi all,
I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway.
There's no printed documentation shipped with the unit, but I have a piece
of software for windows that shipped with a different model (which I haven't
tried configuring yet), that uses snmp to set misc variables (ip
I just put multiple IPs on the same interface and use -a eth0:1 ip.
Seems to work fine.
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I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway.
There's no printed documentation shipped with the unit, but I have a piece
of software for windows that shipped with a different model (which I haven't
tried configuring yet), that uses snmp to set misc variables (ip
Hi Paul,
To my knowledge, you can't change the image on them. I recently bought 3
of them, and we help from this list, I was able to connect them to my
asterisk server. However, they are not fully functional. I can make
calls and hear calls, but I'm muted. I'm looking for a solution.
The protocol
On Tuesday 04 May 2004 06:13 pm, Jet Bagadion wrote:
i tried using music on hold option in the dial command
exten = ,1,Dial(zap/1/,20,m)
Did you mean exten = ,1,Dial(zap/1,20,m) ?
Anon
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[EMAIL PROTECTED]
Rich et alia,
Seems all of the Mediatrix stuff is configured through snmp
only. Finding
and changing the parameters is a royal pain,
Yer tellin' me!
however others have posted to
the list using that same model.
Really? I wasn't able to come up with anything googling, other than
Ryan Laginski ([EMAIL PROTECTED]) wrote:
Hi Paul,
To my knowledge, you can't change the image on them. I recently bought 3
of them, and we help from this list, I was able to connect them to my
asterisk server. However, they are not fully functional. I can make
calls and hear calls, but I'm
Bartosz Jozwiak wrote:
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to
know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
Yes I think so. But you have to download libr2 and compile it, if I am not
mistaken.
You are mistaken.
Regards,
Make sure you have canreinvite=yes in all peers in sip.conf that the
call goes through.
Also making sure that you don't have tT on any of your Dial
statements in extension.conf.
But your real problem is that you have some type of network problem use
mii-tool eth0
at a bash prompt, and make
At 7:14 PM +1000 on 5/4/04, Duane wrote:
John Todd wrote:
TRIP (RFC 3219) is the answer, but I'm the only one pounding that
drum, it seems. If anyone here on the list has $100,000 to put
together a real programming effort towards getting that
implemented, y'all let me know. The longer this
1. Check if Asterisk is always in the media path, i.e. you need the t or
T option (or something similar) in your Dial statement. Alternatively you
could introduce a canreinvite=no in sip.conf for the GS phones.
2. Check your context setup in extensions.conf and make sure that in call
cases
I am trying to forward an inbound call to go out through another X101P
and I get nothing but a noise like a helicopter sound...
Inbound and outbound are ok if done separately.
I already checked IRQs and they are fine.
Updated the drivers and asterisk and they seem to be ok too.
Turned on and off
John Todd wrote:
I strongly disagree with your summary that TRIP doesn't help the smaller
user. In fact, the reason I'm so strongly an advocate of some type of
TRIP development is that it removes the barriers for small entities in
the pursuit of better call rates for TDM offload and VoIP
Isamar Maia wrote:
I am trying to forward an inbound call to go out through another X101P
and I get nothing but a noise like a helicopter sound...
Inbound and outbound are ok if done separately.
I already checked IRQs and they are fine.
Updated the drivers and asterisk and they seem to be ok too.
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is used in the context
ParkandAnnounce.
Can
On Tue, 4 May 2004 12:32:30 -0400, Tim Sailer [EMAIL PROTECTED] wrote:
Folks,
It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there
another IAX2 client that is usable under Linux (Debian preferred)?
Thanks,
Tim
Did it work before you upgraded asterisk, or you can't get it
didn't encounter the sudden call hangup when i add Answer before
that.
exten = ,1,Answer
exten = ,2,Dial(zap/1,20,m)
On Tuesday 04 May 2004 06:13 pm, Jet Bagadion wrote:
i tried using music on hold option in the dial command
exten = ,1,Dial(zap/1/,20,m)
Did you mean
Hello List,
anybody knows the sip url of vonage ???
like [EMAIL PROTECTED] ??
regards.
-Neo
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I had a similar problem after a CVS update and had to set the rxgain to
-2 to reduce the time the echo canceller kicked in...
The problem is that my settings now only work well with
rxgain=+15
txgain=+15
Setting rxgain to -10, the noise disappeared but I can hear only one side
of the line.
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