Hi,
> I also have OH323
installed, supposedly correctly, and the same
> gateway I want to connect to on SIP also
supports H323
To check if is correctly installed, set locally a
Netmeeting client with your Asterisk server as a gateway (in advanced
options)... then try to call one of your local extensions (enter the extension
number in the dial field in NM).
OH323 can use an external gateway, which can be
your 216.52.153.206
I'm afraid that I cannot help you with that.
BR,
Dan
----- Original Message -----
Sent: Tuesday, June 03, 2003 3:12
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
sorry i'm sending so many emails, I always think
of something
exactly after i've pressed Send .. please be
patient with me :)
I also have OH323 installed, supposedly
correctly, and the same
gateway I want to connect to on SIP also supports
H323, however
i do not know what the dial command line for
H323 is .. i'm trying
exten => 1304,1,Dial(OH323/216.52.153.206)
;ring
but I actually want to dial extension 723 on the
remote end,
so this is surely not right.. current messages
i'm getting
from Asterisk are these :
*CLI> dial 1304 --
Executing Dial("OSS/dsp", "OH323/216.52.153.206") in new
stack *CLI>
0:03.623
H323 Cleaner H323 Connection ip$localhost/9771
terminated. ERROR[1232188736]: File chan_oh323.c, Line 610 (oh323_call):
H323:0: Could not call 216.52.153.206. -- Couldn't call
216.52.153.206 -- Hungup 'H323:0' == Everyone
is busy at this time
help *very* welcome ;)
cheers
Dave
----- Original Message -----
Sent: Friday, May 30, 2003 7:50
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi Dave,
If you have registered the SIP phone with
Asterisk, then you must have a line like:
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
----- Original Message -----
Sent: Friday, May 30, 2003 6:21
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
I have included a dump of the debug info
...
what I am trying to do is route a call from
sipphone 217.168.168.49
If i dial direct from the sip phone to the
gateway it works fine .. so
I do not think there is any incompatibility
there.
Calls don't go through though
...
please help!!!
cheers
Dave
*CLI> -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-eca2 answered
SIP/217.168.168.49:5060 -- Attempting native bridge
of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-eca2 WARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060' WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-1418 answered
SIP/217.168.168.49:5060 -- Attempting native bridge
of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-1418 WARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060' WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-11ed answered
SIP/217.168.168.49:5060 -- Attempting native bridge
of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-11ed WARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060' WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
----- Original Message -----
Sent: Thursday, May 29, 2003 8:15
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi,
Check to have a common set of
codecs.
If X-Lite is used and at the other end is a
phone without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
----- Original Message -----
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a
beginner's SIP question ..
I am trying to get asterisk to dial this
address :
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice
prompt ..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration
:
When from my softphone I
dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in
new stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-7c3b answered
SIP/sipphone-97b6 -- Attempting native bridge of
SIP/sipphone-97b6 and SIP/216.52.153.207-7c3b
but on my headset all I get is silence ..
the call doesn't drop though.
What am I doing wrong ?
many thanks,
Dave
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