Topic: AsteriskPBX

[Franco]

Very nice, I wonder if xPL alone would work with SME. Time to build another box and try  

[bovnet]

The hub certainly should compile on a SME, the installer includes a compiled version and installer. It fails on my SME. The service is reported to start but status says its stopped.

Got any automation project in mind ?

I would like to move my VB6 stuff to linux but have no c skills and havent seen any other language that does sockets and serial that i can grasp easily.

[Texasboy]

I got asterisk working on my 6.0.1 server today!!!! has anybody loaded asterisk on the new 6.5?

thanks
Texasboy

   

[Skydiver]

I got asterisk working on my 6.0.1 server today!!!! has anybody loaded asterisk on the new 6.5?

thanks
Texasboy

   

It works a treat on SME 6.5


Cheers

[tag]

Careful with asterisk sources at the moment - there is an error in chan_zap.c:2879

Line reads if (p0->tranfer && p1->transfer

Spot the deliberate???

It should, of course, read

if (p0->transfer && p1->transfer

Otherwise you will get the error==>

[snip]

chan_zap.c: In function zt_bridge':

chan_zap.c:2879: structure has no member named tranfer'

make[1]: *** [chan_zap.o] Error 1

make[1]: Leaving directory

[snip]

Other than that it all runs fine on 6.5 provided you install the correct rpms in the correct sequence....

Just for completeness - here they are....

kernel-headers-2.4.9-34.i386.rpm
cpp-2.96-113.i386.rpm
glibc-devel-2.2.5-44.i386.rpm
gcc-2.96-113.i386.rpm
kernel-source-2.4.20-37.7.legacy.i386.rpm
bison-1.35-1.i386.rpm
bison-devel-2.0-4.i386.rpm
ncurses-devel-5.2-26.i386.rpm
openssl-devel-0.9.6b-36.7.legacy.i386.rpm
zlib-devel-1.1.4-8.7x.i386.rpm

then you can get the latest asterisk sources from CVS:

export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot

CVS login (passsword=anoncvs)

cvs checkout zaptel libpri asterisk

fix the bug which you will find in

 asterisk/channels/chan_zap.c

then do your make clean ; make install for zaptel then libpri and, finally, asterisk.

Phew  

[Drifting]

I have followed this thread with interest.
Having played with Asterisk@home and being very impressed, I thought how nice it would be to have it on SME. So are you all saying it is now working? are your SME boxes setup for gateways or servers? As it seems to me (Who is stuck behind a MS ISA server that won't pass SIP) that a solution with SME with some type of firewall would be good (I am saying this out of total lack of knowledge on SME)

Interested to know the latest..

Drift.

[Franco]

Imho, have SME in Server/Gateway mode and asterisk@home behind it.
If more than one system isn't an option, edit things in asterisk by hand instead of using AMP.

[Drifting]

Imho, have SME in Server/Gateway mode and asterisk@home behind it.
If more than one system isn't an option, edit things in asterisk by hand instead of using AMP.

Thanks for the info, I did read about the problem with AMP and having access to the shell. I really must make the time to read up on Asterisk,I assume you are saying that SME will pass the required ports? or does that require configuring on SME? Most of the problems I have had is that I have an *@home setup and we at work are behind an MS ISA server, which flatly refuses to play nice with SIP.(MS say it won't work) So it looks like your suggestion is the route to go.


Thanks again.

Drift.

[Franco]

The ports would still need to be open/forward for SIP, and SIP uses a lot of ports (may be the reason they say it won't work?). But again *@home isn't limited to sip and IAX2 works right after setup, even behind the ISA Server, without the need to open ports.

[Drifting]

The ports would still need to be open/forward for SIP, and SIP uses a lot of ports (may be the reason they say it won't work?). But again *@home isn't limited to sip and IAX2 works right after setup, even behind the ISA Server, without the need to open ports.

Mmm very thought provoking, I just downloaded the DOcs for Asterisk proper, and will start to have a read up on IAX2, sounds like that will solve the problems with ISA.

Thanks ever so much.

Drift.

[Franco]

It works a treat on SME 6.5


Cheers

Do you an example of your setup you can share? (extension,sip,oh323,zapata,..., conf files)
I see you were able to load OH323 as well! (An How-To on that would be great)
I'm trying to have the same setup running on a 6.5 system, zapata gets loaded but I'm in a point where you once got stuck before

Code: [Select]

ast_request: No channel type registered for 'Zap'.
Thanks,

[wallyrp]

Good Afternoon,

Still pulling hair here due to the fact that I can't find some of the files needed by various aspects of Asterisk. The links are broke and/or the files are no longer available. The how-to and files provided by duncan are straightforward even though Asterisk has been updated to 1.2, I'm willing to settle with the older version to learn it.

I'm trying to get asterisk and amp working together. When I went through all the stuff last night, I finally got down to step 18 but ran into errors with the script apply_conf.sh when it attempted to apply some permissions to a folder in the var folder. I've reformatted my SME box for the third time and am going at it again. I'm very appreciative of all the effort by others that have more knowledge and such to get as far as we have with Asterisk. So, round the mulberry bush we go again.

Here are a few files from the AMP how-to that I'm not able to get so I went up to pre11 instead of pre4. I don't know if that will affect the whole installation or not.

wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_rxfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_txfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/apps_makefile.patch

The links above are not working and/or I'm not able to get to opencall.org to get these files. Anyone know of any other location where I might be able to get them?

[wallyrp]

Good Evening,

OK, I need the following files:

cd /usr/src/asterisk/apps
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_rxfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_txfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/apps_makefile.patch
patch < apps_makefile.patch

These are mentioned in the AMP how-to. I've gotten all the way down and compiled asterisk but it gets the following when I execute it because I'm using pre11 files instead of the pre4 mentioned in the how-to:

[app_rxfax.so]Ouch ... error while writing audio data: : Broken pipe

Any help??

[wallyrp]

Good Afternoon,

Arrgggh, I've tried to strip out all of the references to those files I need from opencall.org but evidently haven't found it yet. Still pecking at it.

[Franco]

I sent on email!