Hi,
Dave Alan Caruana wrote:
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,
First, make sure to specify a codec type, in oh323.conf, that is supported by the gateway. If a gatekeeper is used and the gateway and Asterisk are registered on this gatekeeper, then you should do:
exten => 1304,1,Dial(OH323/723)
If there is no gatekeeper involved, do:
exten => 1304,1,Dial(OH323/[EMAIL PROTECTED])
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
Michael.
----- Original Message ----- *From:* Dan <mailto:[EMAIL PROTECTED]> *To:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *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:
exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207
<mailto:SIP/[EMAIL PROTECTED],52,153.207>)
in extensions.conf file
Then call 555 from the SIP phone to access the destination.
BR,
Dan
----- Original Message ----- *From:* Dave Alan Caruana <mailto:[EMAIL PROTECTED]> *To:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *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
through asterisk 217.168.168.51 onto a gateway
[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
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] <mailto:SIP/[EMAIL PROTECTED]>") in new
stack
-- Called [EMAIL PROTECTED] <mailto:[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]
<mailto:[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]
<mailto:[EMAIL PROTECTED]> for seqno 1 (Response)
-- Executing Dial("SIP/217.168.168.49:5060",
"SIP/[EMAIL PROTECTED] <mailto:SIP/[EMAIL PROTECTED]>") in new
stack
-- Called [EMAIL PROTECTED] <mailto:[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]
<mailto:[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]
<mailto:[EMAIL PROTECTED]> for seqno 102 (Request)
-- Executing Dial("SIP/217.168.168.49:5060",
"SIP/[EMAIL PROTECTED] <mailto:SIP/[EMAIL PROTECTED]>") in new
stack
-- Called [EMAIL PROTECTED] <mailto:[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]
<mailto:[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]
<mailto:[EMAIL PROTECTED]> for seqno 102 (Request)
----- Original Message ----- *From:* Dan <mailto:[EMAIL PROTECTED]> *To:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *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 ----- *From:* Dave Alan Caruana <mailto:[EMAIL PROTECTED]> *To:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *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 :
exten => 1303,1,Dial(SIP/[EMAIL PROTECTED]
<mailto:SIP/[EMAIL PROTECTED]>)
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing Dial("SIP/sipphone-97b6",
"SIP/[EMAIL PROTECTED]
<mailto:SIP/[EMAIL PROTECTED]>") in new stack
-- Called [EMAIL PROTECTED] <mailto:[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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