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...sk.org/view.php?id=15971 I'd love to get feedback in the bug tracker in order to get this feature into Asterisk soon :-) Also don't hesitate in asking for help with the configuration. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090930/e6368f81/attachment.htm

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:

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 Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

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 =>

...he same time. Suggestions? Also, is there a way to force SMB to dump its cache and reread the file, other than unmounting and remounting the share? Thanks. -- James ---------------------------------------- James H. Puttick Kerr Vayne Systems Ltd. 1 Valleywood Drive, Unit 5A Markham, Ontario L3R 5L9 Canada +1 905 475 6161 office +1 905 479 9833 fax mailto:james.puttick@kvs.com ----------------------------------------

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

...Has anyone gotten this to work? Or does anyone have any ideas on how to debug why MoH stops immediately after starting? Thanks in advance! Alex Robar *___________________________________________* Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 alex.robar@gearytech.com www.gearytech.com Strategic management of technology for business. Have you seen our new Voice over IP telephone systems f...

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

Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue

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

I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...king with 32bit Windows 2003 (or was it Windows 2008?). I know KVM is granting the 4 CPU's since Linux VM can use 4 CPU's on the same computer. Thanks, Marcelino Marcelino Mata Engineering Technology Specialist Multimatic Technical Centre 85 Valleywood Drive Markham, ON Canada L3R 5E5 Tel.???+1 905 470 0025 ? ? ? CONFIDENTIALITY NOTE: This email (and attachments, if any) contain(s) information that is PRIVATE and CONFIDENTIAL, and may be LEGALLY PRIVILEGED or otherwise exempt from disclosure under applicable law. All copyright is reserved, and no right or license is granted...

Hi Everyone, We are using two Sangoma U100 (USB FXO) units connected to an Acer Aspire Revo (little PC running on Atom). The units work beautifully except for Monday :-) It maybe a conincedence or maybe the fact that Saturday/Sunday is off and something happens where one of these U100 modules goes into sleep and that's when all the 4 Dahdi channels are lost. So, I have been getting Monday

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

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