Hi,

i am trying to install the just from git cloned app_swift version.
Compiling works fine. Install as well. But if i try to load the module
at Asterisk it fails with.

Command ‘module load app_swift.so ‘ failed.
[Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module:
Error loading module ‘app_swift.so’:
/usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close
[Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module
‘app_swift.so’ could not be loaded.

My System Informations:

server*CLI> core show version
Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on
2012-06-20 08:55:14 UTC

root@server:~# uname -r
3.2.0-25-generic

root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so
linux-vdso.so.1 => (0x00007fff6d3ff000)
libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2010a65000)
/lib64/ld-linux-x86-64.so.2 (0x00007f2011041000)

root@server:~# cat /etc/ld.so.conf.d/swift.conf
/opt/swift/lib

root@server:~#ldconfig -v | grep swift
/opt/swift/lib:

libswift.so.6 -> libswift.so.6.0
libceplex_de.so.6 -> libceplex_de.so.6.0
libceplang_de.so.6 -> libceplang_de.so.6.0

root@server:~# swift -V

Cepstral Swift v6.0.1, March 2012

Default Voice: Matthias-8kHz v6.0.0
Language: German v5.1.0
Lexicon: unknown v0.0.0

Concurrency: 1 Port(s) Registered
0 Port(s) In Use

Distribution: No audio distribution license was found.
Saving audio to a file is disabled.

Copyright (C) 2000-20012, Cepstral LLC.

Do You have any Ideas why that won’t work?

Best Regards Jakob Böttger

smime.p7s

Hello,

I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like “number 12345 not available” I was only
hearing “345 not available”. Verbose level 5 on the asterisk console didn’t
give me any hint on this, it only shows that playback of the prompt started
correctly in every test case. Any hints on how I can debug this? I think
it’s some problem on my local configuration, I doubt it’s a problem with my
SIP provider or mobile phone provider, they are both very reliable (Sipgate
and T-Mobile).

Thanks for any hint!

Best regards
Stefan

Hi,

I’m just trying to install the DPMA on my Asterisk. I already made the
upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:

*mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
*
*compiling Asterisk-Cert2 1.8.11*
*./configure
make
make install
make config
*
Afther that i register the DPMA license, and finally copied the *
res_digium_phone.so* to */usr/lib/asterisk/modules *

When i try to load the module on asterisk console this is what i get>

**CLI> module load res_digium_phone.so
Unable to load module res_digium_phone.so
Command ‘module load res_digium_phone.so’ failed. *

With *tail -f /var/log/asterisk/message *

*[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module
‘res_digium_phone.so’: libavahi-client.so.3: cannot open shared object
file: No such file or directory
[Jun 11 18:53:26] WARNING[2554] loader.c: Module ‘res_digium_phone.so’
could not be loaded. *

Hope you can help

Hello,

I know this a little bit off topic but maybe someone of you has experience
with this topic.

I want to run a hylafax install in an openvz container with iptables.
My Problem is that I can load the kernel module for ftp connection tracking
(nf_conntrack_ftp) in the containers config file but no idea how to set the
parameter to track the right ports.

My vz config looks like:

IPTABLES=”ipt_REJECT ipt_tos ipt_TOS ipt_LOG ip_conntrack ipt_limit
ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl
ipt_length ipt_state iptable_nat ip_nat_ftp ip_conntrack_ftp ipt_conntrack
ip_tables ”

On the host machine the module nf_conntrack_ftp ports is loaded with the
parameter ports=21,4559.

With best regards

David Oppermann
Voip Engineer // voip@sil.at // Tel 059944-2440 //

Hi Tim,

Thanks for your response. Here is my topology as listing down below;

PSTN Line –> Cisco Voice GW –> IP Cloud –> Asterisk

Will Asterisk able to receive the fax (as in topology above) using its’ fax
module? In sip.conf I enabled fax detection and T.38. Actually I don’t
want
to use Hylafax + iaxmodem as per requirement.

Please advice.

> Date: Fri, 1 Jun 2012 10:19:12 -0400
> From: Ahmed Munir
> Subject: [asterisk-users] Fax over IP ?
> To: asterisk-users@lists.digium.com
> Message-ID:
> > >
> Content-Type: text/plain; charset=”iso-8859-1″
>
> Hi all,
>
> Couple of things I would like ask, does Asterisk provides free license for
> FoIP (for 1 channel) or need to purchase it? Couple of years back, I was
> able to send and receive the fax using Digium T1 card, in term of FoIP how
> can I able to receive fax from traditional telephone lines / T1 lines? As
> far my understanding, the functionality for FoIP is to send fax to email or
> receive fax from email i.e. using T.38 protocol.
>
> The thing I would like to know how I can implement this solution i.e.
> receiving fax via IP? Correct me if I’m wrong, while receiving fax from
> traditional telephone lines will the topology looks like as listed below;
>
> PSTN Lines –> Asterisk (mounted a T1/ analog card) –> IP –> Asterisk
> (receive Fax over IP)
>
> or else?
>
> Please advice.
>
>
> –
> Regards,
>
> Ahmed Munir Chohan
> ————– next part ————–
> An HTML attachment was scrubbed…
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20120601/3c861c56/attachment-0001.htm
> >
>
> ——————————
>
> Message: 15
> Date: Fri, 1 Jun 2012 09:29:19 -0500 (CDT)
> From: Tim Nelson
> Subject: Re: [asterisk-users] Fax over IP ?
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <2939915.57880.1338560959074.JavaMail.root@rockbochs.com>
> Content-Type: text/plain; charset=utf-8
>
> —– Original Message —–
> > Hi all,
> > Couple of things I would like ask, does Asterisk provides free
> > license for FoIP (for 1 channel) or need to purchase it? Couple of
> > years back, I was able to send and receive the fax using Digium T1
> > card, in term of FoIP how can I able to receive fax from traditional
> > telephone lines / T1 lines? As far my understanding, the
> > functionality for FoIP is to send fax to email or receive fax from
> > email i.e. using T.38 protocol.
> > The thing I would like to know how I can implement this solution i.e.
> > receiving fax via IP? Correct me if I’m wrong, while receiving fax
> > from traditional telephone lines will the topology looks like as
> > listed below;
> > PSTN Lines –> Asterisk (mounted a T1/ analog card) –> IP –>
> > Asterisk (receive Fax over IP)
> > or else?
>
> FoIP typically means the fax session traverses an IP link at some point,
> most commonly at the ‘last mile’. What happens to the fax after that is up
> to your requirements. The faxes can be emailed out, stored in a web
> application, printed to a printer, etc. The possibilities are endless.
>
> Asterisk does have a few options for faxing. Those are most notably:
>
> 1. Fax for Asterisk – Free license available for 1 channel, or paid
> licenses for 2+ channels
> 2. app_fax (I think this is the current module name) – Free fax module for
> Asterisk, no channel limit, based on SpanDSP
> 3. Hylafax+ and IAXmodem – Most complicated method of fax setup, but most
> robust and reliable (in my testing). Would require use of Asterisk 10 with
> T.38 gateway functionality for proper fax reception.
>
> Just keep in mind raw fax audio over VoIP is a bad idea, see here:
> http://www.soft-switch.org/foip.html
>
> If you can provide some additional details on what you’re planning to do,
> we can give more info.
>
> –Tim
>
>
>
> ——————————
>
> Message: 16
> Date: Fri, 1 Jun 2012 09:36:50 -0500 (CDT)
> From: Tim Nelson
> Subject: Re: [asterisk-users] Fax Server for Asterisk
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <13867195.58161.1338561410617.JavaMail.root@rockbochs.com>
> Content-Type: text/plain; charset=utf-8
>
> —– Original Message —–
> > Hi Tim,
> > Unfortunately i can’t reproduce the scenario because it was a long
> > time ago. But it would be nice to hear from you, what things can be
> > verified within fax and Asterisk? Any TIP on wireshark monitoring?
>
> Within Asterisk, the debug logs can be helpful for routing/connectivity
> diagnostics. With Hylafax, all of your details will be found in the session
> logs in /var/spool/hylafax/log. Here you can see each session’s interaction
> with the remote fax device. It is an art deciphering the various protocols,
> but the folks on the Hylafax lists are incredibly helpful until you’ve
> learned the magic of understanding the logs directly.
>
> –Tim
>