Is anyone working on getting Vonage to work with asterisk? I currently
use an X100P to connect to the ata186 Vonage sent me. With this setup
asterisk has never been able to receive caller id from the ata186. Any
ideas on what I could do?
___
I have noticed when I add dtmfmode=inband under the [general] section
in sip.conf I get flooded with warnings on the console after asterisk
answers a sip call...
WARNING[16401]: File dsp.c, Line 1106 (ast_dsp_process): Unable to
detect process 2 frames
WARNING[16401]: File dsp.c, Line 1106
does this mean?
On Sunday, March 9, 2003, at 02:52 AM, Jeremy McNamara wrote:
Well, someone is attempting to delete a object that has never been
allocated.
You might give chan_h323 a try, you might have better luck.
http://asterisk.nufone.net
Jeremy McNamara
Ben Clark wrote:
this is what
returned 1 exit status
make: *** [chan_h323.so] Error 1
On Monday, March 10, 2003, at 02:18 AM, Jeremy McNamara wrote:
Ben Clark wrote:
cc: cannot specify -o with -c or -S and multiple compilations
What does this mean?
Means I'm a dumbass. I left trailing whitespace in the Makefile
environment
I am also getting a seg fault when asterisk tries to load
chan_oh323.so. What should I try to get it to work?
[chan_oh323.so] = (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/oh323.conf': Found
Segmentation fault
On Saturday, March 8, 2003, at 09:06 AM, Krzysztof Bujak wrote:
...I tried
:39 PM, Ben Clark wrote:
I applied this patch and have been unable to get it to work... I get
NOTICE[1024]: File chan_sip.c, Line 3954 (reload_config): Ignoring
unknown SIP general keyword 'inbanddtmf'
Does the inbanddtmf=1 go somewhere other than the [general] section?
On Saturday, March 1
I have a blind transfer feature available to me from my telephone
provider and was wondering if asterisk can take advantage of this so
that when a certain extension is called the user is asked for the 11
digit pstn number they wish to call then asterisk flashes the line,
dials the transfer
