similar to: Problem with sounds DTMF's phone keys

Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from

Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be?

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

Now that everything seems to rock I've hit the next hurdle. In my extensions.conf I have the extension: [agi-async] exten => _01XXXX,1,Agi(agi:async) and I can see that the context is "hit" when dialing into *. However my java app that's supposed to receive async agi events get no such events at all, but it does receive other manager API events. * version is 1.6.1.4

Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the

After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx->pstn works fine. The tones simply don't make it through. Tiny brief fragments are all. Please

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we

- we used the C4/C5 (first prefix) byte instead of the apparent ModR/M one as the second prefix byte - early decoding normalized vex.reg, thus corrupting it for the main consumer (copy_REX_VEX()), resulting in #UD on the two-operand instructions we emulate Also add respective test cases to the testing utility plus - fix get_fpu() (the fall-through order was inverted) - add cpu_has_avx2,

On Sat, Mar 07, 2020 at 07:16:51AM +1300, Gavin Stephens wrote: > FYI Chrome just sits there saying establishing secure connection. I've just > installed Ice on another machine and copied over the icecast.pem file and > un-remarked the port and path areas for SSL on port 443 (not 8443).  Same > result. I have: <listen-socket> <port>8443</port>

I'm dealing with an idea to exchange data in a socket connection style or a sort of ftp transfer with IAX2 as the transport medium. An IAX client on e.g. a notebook could establish a connection to any remote machine (also client) via any Asterisk Server where both clients are registered. Due to the unique properties of IAX2 one could connect quite easily to any "hidden" remote

Hello ! I`m trying to find a way to use lzo compression for the data being transferred by rsync. rsync only supports gzip (or maybe ssh -c, which is also zlib), which gives cpu intensive workload - and this cannot be exchanged by a commandline param like we have with tar (--use-compress-program=....) i like to use lzo because it does much faster compression than gzip. I read the following in

Hello list, The dovecot version is 1.2.6 running on Solaris x86 11 (nv-b91). The relevant configuration lines are: passdb ldap { # LDAP database (doc/wiki/AuthDatabase.LDAP.txt.) args = /pfx/etc/dovecot/dovecot-ldap.conf } The file dovecot-ldap.conf is correct and LDAP authentication is working well. We would like to make it possible for users with a X.509 client certificate to log in

On 08/03/2020 16:24, Gavin Stephens wrote: > Sorry for the late update. I didn't realise the replies came and went > off list regarding this. > > I had another suggestion made to me it could still be bad pem files > and to try converting them to pfx then back to pem files, which I did, > and fortunately this fixed it. > > The error message I was getting in the Icecast