similar to: g729 problems

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

As I understand it (and my understanding is obviously incorrect) the switch => statement sells the Asterisk box to resolve (aka lookup) extensions by querying the remote Asterisk server defined in the switch => statement. The switch => statement is used to centralize dialplans. I've not used the switch => statement yet, I'm just trying to understand the ramifications of using

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Is there any way on the iaxtel.com web site to see if my asterisk is registering and what 700 number is associated with my iaxtel account? I registered many months ago but never used it. My asterisk shows registered, but I can't seem to receive any calls (callers get a the user is not registered message) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental)

I received a message from Packet8 last night telling me they added some new area codes. I took a look at their new area code/rate center finder page and it looks like the added a LOT of new area codes. They now have phone numbers available in almost every state. The URL for finding out which area codes and rate centers are available with Packet8 is at http://www.packet8.net/about/areacodes.asp

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

What's the proper way to specify the allowed codecs in iax.conf? It doens't like allow=ilbc,gsm but if I put two allow= lines, one for ilbc and one for gsm it seems to always to want to use gsm. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

I'm starting to write an AGI script. I want to pass more than one parameter to the script, but seem to be unable to. extensions.conf: exten => 85,1,AGI(/etc/asterisk/agi/args.agi,myarg1,myarg2) args.agi: #!/usr/bin/perl print STDERR "FNORD prog = $0\n"; print STDERR "FNORD arg 1 = $ARGV[0]\n"; print STDERR "FNORD arg 2 = $ARGV[1]\n"; print

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip