On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot figure out what is wrong with my IAX2 trunk.
The latest chan_sip.c works for my budgetones with the following lines removed. YMMV.
I haven't bothered to dig in and see what those lines actually do. Did soneone just
get wacky with cut and paste from the peer while loop? Or am I breaking something
else.
Jon
--- chan_sip.c.broken Fri Nov
You need to have an exten entry that matches the number of digits the
telco is sending. You should be able to see this vrom the CLI with a
couple of verbose flags.
When we had a EM wink T1, the telco sent 4 digits of the phone number,
This meant our exten line looked like;
exten = 9004,1,answer
Shoval Tom wrote:
It's not MY dns, it's our ISPs one.
And as I've wrote in an earlier thread, I get the exact same error on four
more ISPs (two more here, and two at the US)
I'm more curious of the 5xx HTTP error message you got. Who sent that if
you can't reach the web server?
To me, it sounds
Actually the NAT is not done in a cisco device.
But even if so, is there a solution?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, November 07, 2003 3:44 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of
It just hoses NAT on incoming (from * to grandstream) calls.
Jon
Meant selecting from a list for dns entries, probably, as you only want to
check about your servers if you get a call in the middle of the night...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Friday, November 07, 2003 4:36 AM
To:
I know, but this is what I got in a reply earlier, so I tried it too, and of
course it isn't there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, November 07, 2003 4:37 AM
To: [EMAIL PROTECTED]
Subject: RE:
Nope, no proxy here, nor at ISP (so they say).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Friday, November 07, 2003 10:42 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Shoval Tom wrote:
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it
Sure, I will be happy to test it for you...
Please let me know more details.
Ta
SJ
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Is anyone having sound quality issues with the GS phones when calling
out to the PSTN through a X100P?
Basically I am doing this..
I call out to my cell phone and then with the cell phone to mu ear I
gently blow a constant stream of iar into the mic on the GS phone.. What
I should hear in my
Hi all,
Anyone know of a small H323 gateway that I can run on the * box or a cheap
PC under Linux or Windoze.
I have a Multitech MV110 FXO box and would like to get it talking to *.
Any help appreciated.
Regards
Dave
___
Asterisk-Users mailing list
1.
install h323 support on your *. All docs are in
/usr/src/asterisk/channels/h323. You MUST follow the instructions to the
letter.
2.
configure Multitech MV110 FXO box to send/receive calls from *
3.
configure * (in h323.conf) to send/receive calls to Multitech MV110 FXO
box
Ta
SJ
Asterisk is an H.323 gateway - if you want it to be ;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: 06 November 2003 07:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 Gateway
Hi all,
Anyone know of a small H323
Alternatively, you may use asterisk-oh323
(http://www.inaccessnetworks.com/projects/asterisk-oh323)
Michael.
Senad Jordanovic wrote:
1.
install h323 support on your *. All docs are in
/usr/src/asterisk/channels/h323. You MUST follow the instructions to the
letter.
2.
configure Multitech MV110 FXO
On Thu, 2003-11-06 at 17:24, Jared Smith wrote:
I've got a mysql+php based company directory for the Cisco 7960s at
http://www.jaredsmith.net/misc/cisco7960/Directory-0.1.tgz
Cool :)
Does the whole 'services.xml' thing apply to the lower-end Cisco IP
Phones like the 7905G, or is it only for
Lal, Deepak (Contractor) wrote:
When I get a SIP call, I get the following error:
*CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp'
Which client is used to place the call? I haven't seen
I am very new to this but can a modem be an extension I have a situation
where a customer is having to pay for Telco co-lines at several tower
locations where he has High-speed IP. He uses this line for 3 reasons 1
alarm system to call out which is easy I think if I am reading it right
but he also
Hello,
Did somebody already try to use ASTERISK together
with Festival to check
e-mail boxes for messages?
-- Bart
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with netmeeting through * ( * dial out with
Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked.
Before, there was chan_oh323 0.5.5 and
Hi,
- Original Message -
From: Tom Shoval [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 1:33 PM
Subject: RE: Iconnect and DIAX - WAS [Asterisk-Users] Questions from a total
beginner
-- Executing Dial([EMAIL PROTECTED]/7,
SIP/[EMAIL PROTECTED]|70) in
Thomas Haeger wrote:
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with netmeeting through * ( * dial out with
Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked.
This is a problem of
Thanks Michael,
for this very special detail :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Freitag, 7. November 2003 14:00
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Unable dial out with
Hi All
I have a snom 200 phone here which works perfectly when using the
handset to playback the voicemail messages etc.
However when I play back the voice using the speakerphone it sounds
choppy. Anyone had this problem before?
Regards
Mark
___
It works now - It seems I had a space after extension# and that was causing a
problem.
The client is a Cirpack (www.cirpack.com) softswitch. The sip debug output (AS
REQUESTED) is:
SIP debug output
*CLI sip debug
SIP
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
= xxx,x,playback(file)
Ing Javier
I did not get any answers to my earlier posting. I hope I have better luck this
time- The question: Is it possible to use Asterisk as media gateway controller?
I know * supports MGCP, but does that also imply that I can use * to control any
third party media gateway (such as one providing media
I am running ntpd on the same machine as asterisk in order for the
grandstream phones to display the time. After a while the time display
fails until the phone is re-booted. Has anyone run into this problem
before? Is it simply a bug in the GS firmware?
Sean
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
= xxx,x,playback(file)
Ing Javier
Shoval Tom wrote:
This problem exists with all of the DNS servers I tried.
I tried several ISPs in Israel and a couple at the US.
; DiG 9.2.1 www.voip-info.org
;; ANSWER SECTION:
www.voip-info.org. 3600IN A 192.168.168.3
This is mystical. If I dig in 192.116.202.99, the
Hi Hans,
b. Hooked up to ISDN PBX using an ISDN4Linux (chan_modem ?) or CAPI4Linux
(chan_capi ?) compatible ISDN controller.
We had real problems with chan_modem, where Asterisk would deadlock after
the first call was hung up. We saw this with both 0.5.0 and CVS from
late October. Other
I have the following setup:
AnalogPhone1--TDM400P-ASTERISK---via SIP---SoftswitchPOTS Phone2
When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well.
When making a call from Phone2, I get a dial tone but after dialing the number
I hear nothing (no ringing tone).
Lubomir Christov wrote:
Hello,
I have the same problem - I can't dial out using the new 0.5.6
chan_oh323 - chan_h323 is working BUT with only one way audio with g729
)
I'm using pwlib 1.4.11 and openh323 1.11.7 (because some problem in g729
support in the newest version...)
My * is
If you mean to use MGCP clients and an E1/PRI to the telco with asterisk
as the 'origo', that's what I do, so yes.
On Fri, 2003-11-07 at 14:24, Lal, Deepak (Contractor) wrote:
I did not get any answers to my earlier posting. I hope I have better luck this
time- The question: Is it possible to
--unique-boundary-1
Content-Type: application/ISUP;version=cp10isup;base=etsi121
Content-Disposition: signal;handling=optional
01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74
77 11 11 0f 06 01 10 00
--unique-boundary-1
Hi!
Content-type: application/ISUP
---
Mark Evans wrote:
Hi All
I have a snom 200 phone here which works perfectly when using the
handset to playback the voicemail messages etc.
However when I play back the voice using the speakerphone it sounds
choppy. Anyone had this problem before?
Regards
Mark
Yes, I think its the echo
On Fri, 2003-11-07 at 14:40, Sean Rodger wrote:
I am running ntpd on the same machine as asterisk in order for the
grandstream phones to display the time. After a while
minutes, hours, days... ?
the time display
fails until the phone is re-booted. Has anyone run into this problem
before?
after awhile? I have had mine running for the past week or so with no
problems. Although my NTP server is a cisco not the asterisk box.
Thanks,
Will
- Original Message -
From: Sean Rodger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 8:40 AM
Subject:
I've noticed the same problem on the BT-102. I would also like to know
this... (cc'ed grandstream to get their opinion)
--
Josh Roberson
Indigent Networks
1.877.677.9647 x1
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi !
Now I can hear nice mp3 through my phone... Great :P
And many thanks for your posts. Now it's working fine... hmmm almost !!!
In fact, I m using DialenMP3.agi. It's a real nice agi script...
For those which would not know, you will find it here :
There would have to be a corresponding change in the SIP dialog or in the
actual audio sent both ways. Can you provide some information on how it
has changed?
Mark
On Fri, 7 Nov 2003 [EMAIL PROTECTED] wrote:
Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the
I had the same problem!!
But with H323 protocol.
Try to put 'r' for ringing, let me know if this helped you.
- Original Message -
From: Lal, Deepak (Contractor) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 10:33 AM
Subject: [Asterisk-Users] No ringing tone
Hi !
Now I can hear nice mp3 through my phone... Great :P
And many thanks for your posts. Now it's working fine... hmmm almost !!!
In fact, I m using DialenMP3.agi. It's a real nice agi script...
For those which would not know, you will find it here :
Hello,
question is can we use today's * MGCP function for trunk routing instead of
the MGCP client.
e.g.
exten = _X.,1,Dial,MGCP/[EMAIL PROTECTED]
where other-sfotSwtich-ip is the IP address of another softSwitch, which
support MGCP protocol.
Please advise.
Thanks,
George
-Original
So, what are you suggest for a SIP/IAX client over a browser ?Lee Goodman [EMAIL PROTECTED] wrote:
ActiveX support PLEASE!Lee- Original Message -From: "Dan" <[EMAIL PROTECTED]>To: <[EMAIL PROTECTED]>Sent: Thursday, November 06, 2003 10:53 AMSubject: Re: [Asterisk-Users] IAX/SIP Client
Hello,
I have a problem with asterisk when dial out to a SIP
provider.
Asterisk send a INVITE with no credentials, the
provider reply with a 401 Unauthorized.
However, Asterisk DOES NOT resend the invite again
with credentials. But it hangs there (maybe waiting
for a ok)
It is this a bug in
I did put r as in (in extensions.conf):
Exten = 514777,1,Dial,Zap/2r2|10
But this still does not help!!
Thanks - DL
-Original Message-
From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Sent: Friday, November 07, 2003 9:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] No
To be more accurate: Can I use asterisk as a Call Agent/Media Gateway Controller
for a third party Media Gateway? E.g. if I have
PABX (not *)---[ Some 3-party Media Gateway ] -- AAL2 --to somewhere
Can I use Asterisk to manage the 3-party Media Gateway using MGCP ? In other
words:
PABX (not
minutes, hours, days... ?
It happens after about 6-7 hours. The problem is very consistent.
I am also running 1.0.3.81 firmware on the phones.
Perhaps this is a config problem with ntpd?
here is my ntp.conf:
server clock.isc.org
server time.nist.gov
restrict clock.isc.org mask
Dan,
You are doing a fine job with your DIAX project. Just in case your
English is a second language and you haven't heard this colloquialism
On Wed, 2003-11-05 at 12:12, Dan wrote:
Hi Gary,
- Original Message -
From: Gary [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday,
Or maybe like this:
(I do not know)
Exten = 514777,1,Dial,Zap/22|10|r
- Original Message -
From: Lal, Deepak (Contractor) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 12:07 PM
Subject: RE: [Asterisk-Users] No ringing tone
I did put r as in (in
Hello,
Probably that would help you :
http://www.voip-info.org/wiki-Asterisk+cmd+Musiconhold
Cheers,
Areski
On Fri, 2003-11-07 at 16:37, Nick Knight wrote:
Hello all,
I am trying to suss out music on hold feature - hopefully to integrate
nicely with SIP phones and the hold button you
anyone around here have the ability to terminate
a .NL phone number to IAX or SIP ??
if so please contact me off list.
thank you
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At 07:01 AM 11/7/2003, you wrote:
Hi !
Now I can hear nice mp3 through my phone... Great :P
And many thanks for your posts. Now it's working fine... hmmm almost !!!
In fact, I m using DialenMP3.agi. It's a real nice agi script...
For those which would not know, you will find it here :
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
,,4,incoming,,Zap/1-1,Zap/4-1,Voicemail,u,2003-11-07
17:43:04,2003-11-07 17:43:04,2003-11-07
17:43:22,ANSWERED,DOCUMENTATION
,,19373693874,incoming,,Zap/1-1,IAX[Voicepulse]/1,Dial,IAX2/[EMAIL
mtm spm wrote:
Hello,
I have a problem with asterisk when dial out to a SIP
provider.
Asterisk send a INVITE with no credentials, the
provider reply with a 401 Unauthorized.
However, Asterisk DOES NOT resend the invite again
with credentials. But it hangs there (maybe waiting
for a ok)
It is
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
Take a look in the root of the source (/usr/src/asterisk) at the
README.cdr and in the /cdr/cdr_csv.c file for descriptions of the fields..
please help me. i want to store these into mySQL
I though to it also, but really I don't know how can I get the pid of a
process ran by asterisk.
I mean, the only think I do it's :print EXEC MP3Player \$key\\n;
Then asterisk take the hand with mp3player applications that will launch
mpg123, etc...
How can I get this pid of the good mpg123
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Sent: 03 November 2003 19:18
To: Asterisk Users
Subject: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for
downlaod...
as promise, at:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
As an
Hi Olle,
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was something
I have been trying to get the DTA310 to work properly with Asterisk for the
last couple of weeks. It seems to connect but it does not play back any
sound and I
cannot dial it by using x-lite. Sip debug looks pretty good. I was
wondering if someone has a working config that they could post so that
Can I use a modem and a soundcard as an fxo
?
I've read in the documentation something , but how
can I do that ?
Regards
Alex
Thanks for that - but how does that plug into a sip client - all this
will do as I understand it is if I forward a call to that extension it
will play music - how do I get it back - and how do I tie it into the
hold button on a sip client??
Thanks again.
Nick
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm file or an output from
festival.
I need to call this numbers on demand(from another
program), since
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
- alsa.conf
- enum.conf
- modem.conf
- modules.conf
- oss.conf:
see sample.call in the Asterisk source directory
On Fri, 2003-11-07 at 11:52, mtm spm wrote:
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm
Alexandru Coseru wrote:
Can I use a modem and a soundcard as an fxo ?
I've read in the documentation something , but how can I do that ?
Don't bother, you will waste more time which equates to money. Buy a
X100P from Digium and support Asterisk.
Jeremy McNamara
server:/usr/src/linux/drivers/isdn# patch -p0
../../../isdn-kernel-dtmf-dsp-patch.diff
patching file isdn_tty.c
patch: malformed patch at line 9: (info-emu.vpar[1]))
what can be this??
Matthew Enger wrote:
And a working patch for linux kernel.
On Fri, 2003-11-07 at 09:30, Matthew Enger
On Fri, 2003-11-07 at 11:43, Alexandru Coseru wrote:
Can I use a modem and a soundcard as an fxo ?
I've read in the documentation something , but how can I do that ?
NO YOU CAN'T. Read the archives, it isn't implemented. Read the flame
wars to see why it most likely will never be
On Fri, 2003-11-07 at 11:57, Mireia Munoz de jesus wrote:
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
Thanks William,
Works fine now.
Wim
- Original Message -
From:
William Carlson
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 9:43
PM
Subject: Re: [Asterisk-Users] Grandstream
problem
try
disallow=all
allow=ulaw
under the general
A simple question.
I found an old Xten Xphone Beta 1.01 that has Sip capabilities.
So i try with my asterisk, but he always try to login using this format:
sip:[EMAIL PROTECTED]:5060
And say that registration failed.
Someone see this kind of thing before ?
I try using Xten Lite and everything
Does everything work fine now? I am still having
problems with SayUnixTime. Voicemailmain2 works
though. The one simple AGI script I wrote doesn't do anything. Asterisk starts
playing and the grandstream just rings. Both work fine on other channels/sip
phones.
Thanks,
Will
-
Thanks for that - but how does that plug into a sip client - all this
will do as I understand it is if I forward a call to that extension it
will play music - how do I get it back - and how do I tie it into the
hold button on a sip client??
The call parking feature may be more what you want.
At 09:20 AM 11/7/2003, you wrote:
I though to it also, but really I don't know how can I get the pid of a
process ran by asterisk.
I mean, the only think I do it's :print EXEC MP3Player \$key\\n;
Then asterisk take the hand with mp3player applications that will launch
mpg123, etc...
You're right,
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using EXEC Dial
channel/number the problem with this is that the script then completes
and does not wait for the call to end..
Is there an alternate way to dial the call and then when the call is
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using EXEC Dial
channel/number the problem with this is that the script then completes
and does not wait for the call to end..
Is there an
John Todd wrote:
exten = 514777,1,Dial,Zap/2|10
Try:
exten = 514777,1,Dial(Zap/2,10)
I think these two versions of giving arguments are confusing. Reading docs
and show application texts, both variants are used, sometimes even in the
same text.
Is the first syntax old, to be
So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I
have the following gripes, which I've sent to a very clueful Cisco
person already. Mind you, I love the Cisco 79xx series phones, and
currently they are what I recommend to anyone who wants a 'real' IP
phone. I just cringe
-
I have following setup:
AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]POTS-AnalogPhone_2
I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine.
When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT
I hear no ringing tone AND when someone picks up
Steven Critchfield wrote:
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using EXEC Dial
channel/number the problem with this is that the script then completes
and does not wait for the call to end..
Interesting. Can you point to where this is documented? I rooted around
thru the Digium online manual, whitepaper, etc, couldn't find any doc.
http://www.voip-info.org/wiki-Asterisk+variables
/O
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[EMAIL PROTECTED]
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE:
Follow the instructions on line below and do NOT issue a make clean
install in asterisk/channels/h323 as indicated elsewhere, just issue a
make and then in /usr/src/asterisk (or wherever you source is), issue
a make install and
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot figure out what is wrong
On Fri, 2003-11-07 at 14:38, Yelson Vivas wrote:
HI Asterisk Gurus
I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P
card as T1 to connect my server to the channel bank (i use a cross cable),
the zplex doesn't show any alarm neither the server. I'm trying to make
On Fri, 2003-11-07 at 14:39, WipeOut wrote:
Steven Critchfield wrote:
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using EXEC Dial
channel/number the problem with this is that the script
Steven Critchfield wrote:
On Wed, 2003-11-05 at 15:03, Steve Murphy wrote:
Everyone--
Here's a cost analysis, rather crude and inspecific, of using Asterisk
to implement a phone system. I'm really quite naive and new to all this,
so I'd appreciate any corrections, tips, pointers,
Michael Manousos wrote:
when i try to make a call with netmeeting through * ( * dial out with
Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked.
This is a problem of OpenH323 1.12.0. Use this dial string:
Dial,OH323/h323:[EMAIL PROTECTED]
Or, even better, use the latest (it has been fixed).
Thank you guys very much,
it work great based on sample.call.
Have a nice weekend.
MTM
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Asterisk-Users
Hi Olle,
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was something like
Yep, we have 8 SNOM200's in an installation and none of them have a usable
speakerphone. There is something frelled up with the voice detection on
those phones when using speakerphone. We talked to kevin in technical
support for the American Distributor and he told us to try the latest beta
On Fri, 2003-11-07 at 15:02, Olle E. Johansson wrote:
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?),
Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two
incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall
wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer
config in sip.conf each works seperately, and I'm trying to use
Yes,
ASTERISK1 has 2x TDM400P and ASTERISK2 has 3 x100P cards in it.
I'll try to dork with the timer, but as long as wcfxo or wcfxs is loaded
shouldn't that take care of these issues?
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday,
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
,,4,incoming,,Zap/1-1,Zap/4-1,Voicemail,u,2003-11-07
17:43:04,2003-11-07 17:43:04,2003-11-07
17:43:22,ANSWERED,DOCUMENTATION
,,19373693874,incoming,,Zap/1-1,IAX[Voicepulse]/1,Dial,IAX2/[EMAIL
Whenever I have had this problem it was certainly a SIP/Firewall issue.
Calls will go through but audio (RTP) will not go through.
Second, using iconnect I have to add an option r (I think) to my dial
command in extensions.conf to make asterisk send ring back to the
originating phone, because I
mtm spm wrote:
Hi Olle,
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was
mtm spm wrote:
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm file or an output from
festival.
I need to call this numbers on demand(from another
Dragan Mickovic wrote:
I have couple of questions about the following. Currently I have 2 phone lines going
into my house, and I would like to have both of those coming into asterisk. I also
want to have a backup asterisk, so here are the main questions (I am knew to this so
I apologize if I ask
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