[Asterisk-Users] a beginner's SIP question ..
Dan
dtoma at fx.ro
Tue Jun 3 05:29:57 MST 2003
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 -----
From: Dave Alan Caruana
To: asterisk-users at lists.digium.com
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 -----
From: Dan
To: asterisk-users at lists.digium.com
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/723 at 216,52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the destination.
BR,
Dan
----- Original Message -----
From: Dave Alan Caruana
To: asterisk-users at lists.digium.com
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 723 at 216.52.153.207
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/723 at 216.52.153.207") in new stack
-- Called 723 at 216.52.153.207
-- 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 call-1054307890-9 at 217.168.168.49 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 call-1054307890-9 at 217.168.168.49 for seqno 1 (Response)
-- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207") in new stack
-- Called 723 at 216.52.153.207
-- 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 call-1054307890-9 at 217.168.168.49 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 call-1054307890-9 at 217.168.168.49 for seqno 102 (Request)
-- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207") in new stack
-- Called 723 at 216.52.153.207
-- 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 call-1054307890-9 at 217.168.168.49 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 call-1054307890-9 at 217.168.168.49 for seqno 102 (Request)
----- Original Message -----
From: Dan
To: asterisk-users at lists.digium.com
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
To: asterisk-users at lists.digium.com
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:723 at 216.52.153.207
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/723 at 216.52.153.207)
When from my softphone I dial
sip:1303 at 217.168.168.51
on the console I get :
-- Executing Dial("SIP/sipphone-97b6", "SIP/723 at 216.52.153.207") in new stack
-- Called 723 at 216.52.153.207
-- 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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