[Asterisk-Users] Call transfer problem
My dial statement is (for testing purposes): 123,1,Dial(H323/192.168.1.55|20|tT) When a caller dials extension 123 I can connect and talk without difficulty. Both the caller and the callee can press # to drop back to asterisk. The caller can dial an extension and transfer the callee. When the callee tries to dial an extension, I get: Unable to find extension '' in context ' ' It seems to me that the callee is not given a proper context and therefore cannot dial extensions in asterisk without first calling the pbx. If the pbx calls an extension, that extension is in limbo. ManxPower from the Asterisk IRC has had the same problem but has not needed the transfer capability so has never looked into it farther. I know other people have gotten this to work because I've read testimonys in the mailing list archives saying that they did get it working. I'm wondering how this was accomplished? I am using a week old version of asterisk from cvs on an Athlon 600 with 512 Megs of RAM, Slackware 8.1, 2.4.20 kernel. John Fortman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer problem
For testing purposes, my dial line is:
Dial(${ARG2},20,tT)
When I call from one machine through asterisk to
another, I can press # from either side and hear "Transfer."
However, from the caller side I can continue on and
put people on hold by dialing '700'.
From the callee side, I can press # but if I try to
dial an extension I hear "I'm sorry. That is not a valid extension.
Please try again." Asterisk displays a message "Unable to find extension
'7' in context ' ' "
What this tells me is that if a VOIP client picks
up a line that has been Dial()ed from asterisk, that client is not given a
context and, therefore, cannot dial extensions. How can this be
fixed? Have I messed up the setup somehow? If so, can anyone give me
a working example?
John.
[Asterisk-Users] ast_channel_alloc() losing pvt struct
I don't understand the reasoning here so could somebody please help me out? chan_h323 is causing a segmentation fault when trying to connect a call. I tracked the problem back to chan_h323.c in the oh323_new() function. the code is: tmp = ast_channel_alloc( 1 ); After this point, tmp->pvt is not allocated (null pointer). HOWEVER, in the ast_channel_alloc() function right before the statement: return tmp; tmp->pvt IS allocated. Now between the statements: return tmp; and tmp = ... the pvt structure is getting lost and the tmp->pvt point is set to 0. Has anyone got a clue here? It doesn't make any sense to me at all. John.
[Asterisk-Users] chan_h323.c
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild.
[Asterisk-Users] # Transfer context problem
Setup:
Asterisk with chan_h323 (chan_iax was connecting
the two clients directly, dropping asterisk out of the picture)
Clients are two pentium class computers on the same
network with ohphone installed.
The idea is simply to have one client call into
asterisk (a client calling from outside) to be
answered by an operator then transferred to a
different extension.
Extensions.conf:
[default]
include => parkedcalls
...
exten =>
s,2,Dial(${ARG2}|20|tT) ; ARG2 = H323/192.168.1.55
The caller dials in.
The operator answers.
The operator dials #
Asterisk plays "transfer" (so
far, no problems)
The operator dials an extension
Asterisk replies "sorry, that is not a valid
extension. Please try again."
The message in asterisk -vvvc is "unknown extension
'X' in context ' ' "
or something to that effect.
What I've found is the following:
line 668 of chan_h323.c (cvs from Tuesday
08/19/2003 around 11:45am)
strncpy( ch->context,
i->context, sizeof( ch->context )-1 );
i is struct oh323_pvt that was generated shortly
beforehand by oh323_alloc() with
no reference in the function to ->context.
THere doesn't even appear be a search
to compare the requested IP address with an H323
user so the context is left blank!
Therefore, the operator cannot transfer because he
is out of context in asterisk.
There appears to be a similar problem with chan_iax
because I can dial to an IAX client
from an H323 client through asterisk and have the
same problem. (IAX client to IAX client
cannot dial # to return to asterisk. Asterisk
hangs up the connection after linking the two
clients.)
My question is: Have I setup extensions.conf
wrong or is this really a bug in Asterisk?
This has been bothering me for about four weeks now
and I cannot fix this correctly.
I have faked it by changing:
strncpy( ch->context,
i->context, sizeof( ch->context )-1 );
to
strncpy( ch->context,
"default", sizeof( ch->context )-1 );
which works to test the transfer function but is no
where near a good fix.
Thank you,
John.
Re: [Asterisk-Users] trying to make a X100P work
You did run "ztcfg -vv" after you modprobed for wsfxo and zaptel, right? I recently got a X100P and it wouldn't show up until I ran the config. John. - Original Message - From: "John Brown" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 19, 2003 4:12 PM Subject: [Asterisk-Users] trying to make a X100P work > Hi List, > > Could some kind soul post me a quick config > that makes use of a Wildcat X100P > > when I do a show channels nothing is there > > lsmod shows the wcfxo and related drivers loaded > and with no errors > > zap show channels is blank as well > > > mucho thanks > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323.c
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but Slackware requires ldconfig to be run)
3) /root/src/openh323: configure, make, make install, ldconfig
4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
card)
5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
do anything here)
6) /root/src/asterisk/channels/h323:
- edit Makfile
- set PWLIBDIR = $(HOME)/src/pwlib
- set OPENH323DIR = $(HOME)/src/openh323
- make, make install (installs openh323.a) (make samples if you do not
have h323.conf in /etc/asterisk when done)
7) /root/src/asterisk: make, make install, make samples
8) asterisk -vvvc
- the last section should load chan_h323
I haven't had any problems compiling this from CVS for almost a month on at
least three different systems with some version of Slackware. I have had
problems with other things like transferring calls but that's a different
issue.
John.
- Original Message -
From: "Sean Figgins" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
> On Mon, 18 Aug 2003, Mark Spencer wrote:
>
> > It's up one directly. It just moved.
> >
> > Run "make" in h323 then do "make install" on asterisk again.
> >
> > On Mon, 18 Aug 2003, John Fortman wrote:
> >
> > > What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
> > > ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
> > > not created so no h323 support in asterisk.
> > >
> > > Just wondering when to expect it again because I was stupid and didn't
> > > make a backup of the asterisk code before wiping the directory for a
> > > rebuild.
>
> Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
> I get a ton of errors from ptlib. I'm about ready to punt this sucker out
> the door. I really like what I have seen out of asterisk so far...
>
> Example of errors:
>
> In file included from /usr/include/ptlib/contain.h:218,
> from /usr/include/ptlib.h:137,
> from ast_h323.h:29,
> from ast_h323.cpp:27:
> /usr/include/ptlib/object.h:585: parse error before `(' token
> /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
> /usr/include/ptlib/object.h:1201: parse error before `(' token
> /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
> /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
> /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
>PObject::BOOL'
> /usr/include/ptlib/object.h:1214: parse error before `(' token
> /usr/include/ptlib/object.h:1265: syntax error before `operator'
> /usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
> /usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
> return
>type
>
>
> Any help would be appriciated, even if it's a recommendation to another
> flavor of linux.
>
> Thanks
> Sean
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Re: [Asterisk-Users] Question on setting up MeetMe conference bridge
I was able to connect two H323 clients in a conference call with only one X100P card. ztdummy did not compile correctly so I'm only using zaptel.o and wsfxo.o. The conferencing worked without any special setup on my part so I can't say that it's anything that I've done. - Original Message - From: Lee Goodman To: [EMAIL PROTECTED] Sent: Thursday, August 21, 2003 10:06 AM Subject: [Asterisk-Users] Question on setting up MeetMe conference bridge So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1 port FXO card). So, do you have to have mulitple Zapata devices to enable multiple users in a conference? Do I have to enable the zdummy (dummy zapata code) to run a multiuser conference? Thanks Lee Goodman PS, if someone could send an example configuration, that would be great
Re: [Asterisk-Users] GS on ebay...
The seller looks a little dubious. There has only been one buyer "juggsbunny." I wonder. What about the quality of these phones? Anyone had any experience here? - Original Message - From: "Brian West" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, August 25, 2003 12:42 AM Subject: [Asterisk-Users] GS on ebay... > 101's for 68.00 > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066051&category=11175 > > 102's for 79.95 > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066389&category=11175 > > Just passing what I find along > > bkw > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GS on ebay...
> Hi John, > > I'm the seller:) I got burned on Ebay once so I'm cautious. 8) Now I will have to look into the phones more closely and see if my company is interested. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
