I'm trying to connect my asterisk 1.4.6 to a system that provides video
content (through SIP).
Problem is my video system only speaks H263-2000 version (aka H263++).
As far as I can see, * only understands H263 and H263+ in and sdp.
Can anybody tell me how to extend asterisk so it'll support
Hi list,
I'm trying to get channel gtalk working in asterisk 1.4.5
I have it built and configured as follows:
*jabber.conf:*
[general]
debug=yes
autoprune=no
autoregister=no
[myaccount]
type=client
serverhost=talk.google.com
[EMAIL PROTECTED]/Talk
secret=mypassword
port=5222
usetls=yes
Would you be so kind to share your experience?
I can read most of C language, but writing it is another thing.
And I'm not familiar with the internals of Asterisk...
Or maybe you could already confirm that my problem is related to NAT (client
or Asterisk side, not sure)
On 6/21/07, [EMAIL
I haven't changed rtp.conf from original installation.
So the values are:
rtpstart=1
rtpend=2
I should maybe give it a try with a lower rtpstart.
What do you mean by turning on NAT?
Are you referring to parameter bindaddr in gtalk.conf? (found that on
Gregory,
I know there is something called SIP CTI TR87.
It's used by Nortel to integrate with Microsoft's Live Communication Server.
Don't know if something similar exists for Asterisk.
This links could be helpfull:
http://www.ecma-international.org/publications/techreports/E-TR-087.htm
Regards,
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon as
separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood as
a separator for
Lindquist [EMAIL PROTECTED] wrote:
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon
as separator.
I was thinking the 'CUT' function
All,
The last Peter got it right! :-)
The final solution:
exten = s,n,Set(sep='\;')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Thanks for you input and have a very nice day!
Koen
On 11/30/06, Peter Boehm [EMAIL PROTECTED] wrote:
_The functions:_
exten = s,n,Set(sep=';')
exten =
Afer running
./configure
with whatever options you need, you should run
make menuselect
That will give you a menu to select the required modules.
Modules marked with XXX are disabled, mostly because of a missing dependency.
I think jingle requires iksemel.
Good luck!
Koen
On 9/28/06,
Sounds like a nice setup you have in mind.
All I can tell is that you might have trouble with clocking on your PRI's if you use multiple cards in one system.
I've read about it somewhere, but can't find the source. Have a look at the wiki.
Syncing clocks on one card happens on the card level. But
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Kenny, you should give it a try!
K
On 7/28/06, Filip Drągowski [EMAIL PROTECTED] wrote:
asterisk does daily log rotate all along ? i didn't know that it is posiiblei create file in
Maye you should use the 'D' option in the Dial application to proceed when the call is answered.
Not sure, and I don't have time to test myself, but give it a try!
K
On 6/16/06, Frederik Fix [EMAIL PROTECTED] wrote:
Hi,I'm trying to setup a system where incoming faxes are received usingSpanDSP
We use dynamic conferences with MeetMe.
As far as I can tell, the 'e' option is not needed.
We use a global var as counter for the conference number.
You provide it with the MeetMe command. This way you always know which conference to join.
K
On 6/16/06, Miles Scruggs [EMAIL PROTECTED] wrote:
Why still use mpg123?
Start using format_mp3 from asterisk-addons and your * will play mp3 by itself...
K
On 6/13/06, Marc Rohlfing [EMAIL PROTECTED] wrote:
Hi,I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) andAsterisk (to
1.2.9.1) at the same time. Now, when trying to
Muhammad,
I have been struggling with M1 and * over an E1 for a while myself, but know it's running fine.
Here's my d-channel config:
ADAN DCH 18 CTYP MSDL CARD 08 PORT 1 DES Asterisk1 USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC EURO CNTY BEL PINX_CUST 0
ISDN_MCNT 300 CLID
Use format_mp3 from asterisk-addons.
It will enable your * to play mp3 without the use of an external process... (if I got it right)
On 6/8/06, Richard Reina [EMAIL PROTECTED] wrote:
Turby,Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH
Is there a way to use 'application mapping' from features.conf without the built in features (pickup, blind transfer, etc.) nor call parking?
I have been trying to comment out everything in features.conf, but my asterisk stills shows the defaults...
Koen
If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls?
More in details: could a global var be used to build a counter that will be incremented by every call that passes.
I think when 2 calls come in almost sumiltaneously, they could both be
Sounds like a reasonable explanation.
But this means that I should limit the incrementing stuff to one line in the dialplan.
This would be bad:
exten = s,1,Set(Chan_Var=${GlobalVar})
exten = s,2,Set(Chan_Var=$[${Chan_Var} + 1])
exten = s,3,Set(GlobalVar=Chan_Var,g)
Better:
exten =
depending on your zapata.conf file, you should use
exten = _9X.,1,Dial(Zap/r1/${EXTEN:1})
The little 'r' means round robin, starting at the next highest channel than last time.
Have a look in extensions.conf from the samples for more options.
Make sure you have your 4 channels in one group
I'm running pretty much the same config in Belgium.
Here's what I use:
zaptel.conf:
span=1,1,0,ccs,hdb3 # no CRC4 used here
bchan=1-15,16-31
dchan=16
zapata.conf:
[trunkgroups]trunkgroup = 1,16spanmap = 1,1,1
I have a TE110P connected in euroisdn as pri-cpe.
When I dial out from a sip phone to a number over the pri, I get an error
Unable to set channel 1 (index 0) to linear mode
On the destination phone, I only get a terrible noise when answering the call.
There doesn't seem to be a speech path...
Hi,
I wonder if anyone is using Digium's TE110P card on an E1 connection.
I have been try to, but so far it wasn't much of a success.
It only works more or less in EuroISDN as PRI CPE.
And even that config gives me some trouble with channel negotiation.
My current config:
zaptel.conf:
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