No need for the pri debug span, the problem is the duration of the tones
when using dtmfmode=rfc2833. It is way to short. A lot of IVR's just
don't get enough of the tone to work. The info method still has the correct
duration.
Simple to test just deal another phone and hit keys, you will see
Leave off the softkey xml tags, This should get you working
You can telnet to the phone and type debug http and you will get better
errors.
Maik Schmitt wrote:
struggling with localization issues (so the script is not German only)
took me a week longer than expected. (Did anybody ever get
Big issues for sip: (Please note I use both Asterisk and Vocal between
the two you can have a fairly scalable sip environment with a fair
amount of call features.)
Pluses for Vocal:
For sip switching Vocal is much more scalable, You can have a cluster of
UserAgents and Gateways. It never
Dave Alan Caruana wrote:
hi ..
I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it ..
1st question:
when i phone out
or receive a call from one of the SIP phones onto the PSTN, there is
a LOT of local echo in the handset .. the
I also had the same problem with sip, I also moved back a couple of
weeks in cvs.
I also use a AS5300 Cisco in my call chain.
I got a bunch of Ignoring this request in debug. I have not had time
to trace the call path on this problem yet.
Low, Adam wrote:
All,
I've found problems in my setup
Also using the -p (real time option) to start asterisk, may also help.
James Taylor wrote:
Carlos,
You may have something here.
Dan - you might try to connect via the virtual network adapter to your host machine's hard drive. (just map a drive) it could be that networking from your VM to the
Yes I have seen it. I had to change the digit time in the voicemail app
and recompile.
There is a new voicemail2 app. I have not used it, but maybe it fixes
this problem.
If you test it out, let me know how it works for you.
Brian Borders wrote:
I have a problem with using voicemail on the
Can I use a WILDCARD TDM400P to connect to
four Telco circuits aka FXO? Or will I need
four Wildcard X100P?
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What does -p do?
Mark Spencer wrote:
You can try using the -p option to Asterisk.
Mark
On Wed, 2 Apr 2003, Jeff McClure wrote:
Good points. This system currently does not use any SIP or IAX channels (or
any other form of VoIP) and only deals with 1 call at a time (the single
FXO channel is
The pending route patch may fix you (I'll be testing it once it is merged)
but if not, try using a b2bua on out going calls this should hide Asterisks
ugliness from Iconnect.
A mostly working b2bua can be found here:
http://www.vovida.org/downloads/vocal/1.5.0/rh73/b2bua-1.5.0-20.i386.rpm
I don't
crypto keys at startup
-c Provide console CLI
-d Enable extra debugging
-p = Realtime Priority
Andre
- Original Message -
From: James Sizemore [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, April 05, 2003 12:26 PM
Subject: Re: [Asterisk-Users
Yes.
Lele Forzani wrote:
Has anybody noticed that # transfers aren't working anymore when SIP is used
with rfc2833 dtmfmode? They work as espected with inband dtmf.
lele
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I agree, whole heartily, No XML please! I suggest the requester,
take a look at Vocal if he thinks XML is a good ideal for any-e-thing
at all. I am glad most Unix configuration files have avoided XML hell.
Problem will all XML configs:
1. They are nearly imposable for a human to read, for any
Chris you seem to like the things I hate the most! LOL
About the only thing I hate worse then XML for config
files is using M4 for any-e-thing at all!!! grin Your
a sick man, you just seem to love needless steps in editing
a config file! I'll stick to Macros, myself.
Chris Albertson wrote:
I got some time this week end to play with
this. By add the pickup lines in
extensions.conf:
;
;Parked calls
;
extern = 701,1,ParkedCall(701)
extern = 702,1,ParkedCall(702)
extern = 703,1,ParkedCall(703)
extern = 704,1,ParkedCall(704)
extern = 705,1,ParkedCall(705)
extern = 706,1,ParkedCall(706)
Thanks,
P.S. What is the URL to your wish list? smile
Mark Spencer wrote:
SIP does not yet support parking unless you do #transfer support. The
reason is that once you have done a transfer in SIP, the original call is
gone, so there is no way to announce where the call has been parked.
Mark
I send example XML files a week or so ago, as well as
an impotent line for you dhcp server.
Mike Reiling wrote:
Anyone know if it is possible to load your own XML scripts on to the
phone, bypassing the Cisco CallManager? I am still waiting for my
phone to arrive, but I have been playing with
, Mar 13, 2003 at 06:47:44AM -0600, James Sizemore wrote:
Message waiting indication work fine you
just need to set up DDNS for phones Asterisks
Needs to know how to reach the phones!
[2114]
type=peer
username=2114
insecure=yes
canreinvite=no
context=default
Just check-out asterisk from cvs, It compile but
crashes right off with?
# Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe
Any ideal how far back I need to go to get a working
build?
___
Hell
rm -rf asterisk
cvs checkout asterisk
make samples
Same thing!
Mark Spencer wrote:
make clean ; make install?
Mark
On Sun, 9 Mar 2003, James Sizemore wrote:
Just check-out asterisk from cvs, It compile but
crashes right off with?
# Ouch ... error while writing audio data: : Broken
The only problem I can think that you would have with the
ztdummy would be that to used a kernel source other
then the one your running when you build it...
So what errors did you get when you build ztdummy?
Rattana BIV wrote:
No I use chan_capi and H323 but not zaptel device.
So can I use it ?
Uncomment ztdummy from zaptel/Makefile
make clean ; make install
modprobe ztdummy.
Restart asterisk, all fixed.
Rattana BIV wrote:
Hi,
I try the application MeeMe but i Have a problem when I call a conference.
It show me : Unable to open pseudo channel
Does anyone can help me ?
regards
A few note about each file.
OS79XX.TXT: Should always have this old version of the code for the phone.
Ringlist.xml: Lets you have custom ring tones (not a good to have Bart
saying eat my shorts as your ringer)
SIP-MAC-ADDRESS.cnf : Phone setup goes here. Set You telnet password if
you want to
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