similar to: MOH and Licensed G729 codec

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

Hi Nitesh, Take a look at this http://www.microappliances.com/site/html/index.php?section=Products&page =clienthowto.php I've never implemented it though so I would appreciate some feedback on if it works. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Saturday,

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

Hello Anton, Thanks for replying. The message "as: unrecognized option `-mcpu=cortex-a8'" is the same message I get when I just enter: as -mcpu=cortex-a8 So I think I llvm-gcc is using the Linux i386 assembler to assemble ARM code. Not good. What assembler is it supposed to use? The one in CodeSourcery? > But how's assembler actually called? Just try to compile

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Hi, Neel > I suspect it is using the wrong assembler. But when I do llvm-gcc -v I > can see it has: > >  --with-gnu-as=/home/neel/CodeSourcery/Sourcery_G++_Lite/bin/arm-none-linux-gnueabi-as But how's assembler actually called? Just try to compile something with extra -v option added -- With best regards, Anton Korobeynikov Faculty of Mathematics and Mechanics, Saint Petersburg

I suspect that my llvm-gcc cross compiler is using the wrong assembler because it does not recognize "-mcpu=cortex-a8". I was trying to build a cross compiler for a Mac host. Now I am trying to build on x86_64 Linux. I am targeting a Beagle board with an ARM Cortex-A8 and Angstrom Linux. TRIED: to use the script in llvm/utils/crosstool/ARM/build-install-linux.sh I used the recommended

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 -------------- next part -------------- An HTML

Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks.

Hello, I am trying to cross compile LLVM. Build and compile on Mac OS X 10.5.7 Target is an Angstrom Linux Cortex-A8 Beagle board. I would like to avoid GPLv3. I can compile. But when I run my executable on Beagle crt0.S is calling an AngelSWI to setup the heap and stack. But the AngelSWI seems to just be a stub because it returns zeros for the pointers. That leads to a segfault. If I hack in

Hello All, Is there any way to clear the SIP Channels? When I run "sip show channels" on CLI I see +500 SIP Channels active with "unknown" codec. But thats false information, because when I restart my Asterisk and run "sip show channels" I will see the actual active channels with correct codec info. Anyways to clear the sip channels without restarting the

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

Hi all, I''m thinking of useing sub-domains as account keys, something I''ve played with previously. I have accounts and users, fairly standard, except that users can belong to many accounts, and accounts can have many users. So I figure I need to have an area on the main domain, that allows users to view their account memberships and things. Sort of like a dashboard for each

On Nov 9, 2009, at 7:34 PM, Neel Nagar wrote: > I tried to speed up Dhrystone on ARM Cortex-A8 by optimizing the > memcpy intrinsic. I used the Neon load multiple instruction to move up > to 48 bytes at a time . Over 15 scalar instructions collapsed down > into these 2 Neon instructions. > > fldmiad r3, {d0, d1, d2, d3, d4, d5} @ SrcLine dhrystone.c 359 > fstmiad

Hello, I am trying to REGISTER asterisk to a SIP server, which is listening on Port 6060 (not 5060). The sip.conf file contains register=18474201111:quintum@192.168.2.94:6060/18474201111 maxexpirey=3600 defaultexpirey=120 But the REGISTER message is sent to Port 6060, but the Request-URI still contains, 5060. This is being rejected by SIP server. REGISTER sip:192.168.2.94

I tried to speed up Dhrystone on ARM Cortex-A8 by optimizing the memcpy intrinsic. I used the Neon load multiple instruction to move up to 48 bytes at a time . Over 15 scalar instructions collapsed down into these 2 Neon instructions. fldmiad r3, {d0, d1, d2, d3, d4, d5} @ SrcLine dhrystone.c 359 fstmiad r1, {d0, d1, d2, d3, d4, d5} It seems like this should be faster. But I did