similar to: invalid extension -> missing the original ${EXTEN} value

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.

Hello All, I always used a compiled version for a x86 system >From http://kvin.lv/pub/Linux/Asterisk/ , and Works fine. At this time , I'm using a Pentium IV Dual core, Running Centos 4.3. I tried to install the 64 bits Compiled version but has a translation time > 20ms. Is it correct ? I also tried to make my self compilation and apply The patch but the patch doesn't work on ICC

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Sorry, it seems I have only replied to Lakhdar, not to the newsgroup. Below is my reply to Lakhdar, and I would like to make it more clear now, using some pseudo values for simplicity: I read bytes 1 to 124 from my encoded spx file. I decode themt and get the values: ---Frame 1---- -293 -8234 2134 17 ---Frame 2---- -9323 -732 189 2329 Both frames are just perfect as I need them. But now when I

Hi Friends, Is Apple IPhone mobile is released in India? Is Apple IPhone mobile is released in USA? If IPhone is released in India, Can you tell me any Apple authorized showroom in Hyderabad (Andhrapradesh, India)? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew

I'm trying to add 2 FXO cards but when the second is added, asterisk stop to respond (zap channel): app_dial.c:545 dial_exec: Unable to create channel of type 'Zap' modprobe zaptel and modprobe wcfxo didn't return any error. ztcfg -vv is reporting only 1 card: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels

Most of my outbound calls are handled by macro contexts: [macro-pstn] exten => s,1,Wait(.5) exten => s,2,PlayBack(beep) exten => s,3,PlayBack(silence/1) exten => s,4,Dial(${ARG1}/${ARG2}${ARG3}) exten => s,5,Playtones(Congestion) exten => s,6,Wait(3) exten => s,7,Hangup which is called like exten => _920[1289]XXXX,1,Macro(pstn,Zap/g1,9,${EXTEN:${TRUNKMSD}}) Because of

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

I found out something: The more frames I decode before the frame that I actually want to decode, the better the quality becomes. For example when I basically want to decode frame #100, I read frame #80 to #100, and then frame 100 has the quality that I need. Why? Is there any information on this behaviour? Thank you. Hermann Am 23.12.2011 18:37, schrieb Hermann Weber: > It would be nice

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), "manually" patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

Hello, I have a vector (numeric) v-> c(a,b,c,d,e) and I want to create the vector n->c(b-a,c-b,d-c,e-d). How can I do that? Thank you Hermann [[alternative HTML version deleted]]

My file is 3 hours long, so decoding takes around 5 minutes on an average computer. That is a bit too long unfortunately... Am 23.12.2011 20:38, schrieb Steve Checkoway: > On Dec 23, 2011, at 10:54, Hermann Weber<hermie.weber at gmx.de> wrote: > >> And how many frames does Speex need to "recover"? >> Or is that not predictable? > No idea. My guess is not

Hello, I wish to count how often zero (0) appears in the vector test. test [1] 1 1 1 1 1 1 2 1 1 1 0 2 0 1 1 0 0 0 1 1 1 0 1 2 1 1 1 1 1 1 I think of something like ... > sapply (test, function (x) if (x==0 ... ... but actually I dont know how to carry on. Could anybody give me a hint? Thanks Hermann [[alternative HTML version deleted]]

And how many frames does Speex need to "recover"? Or is that not predictable? Greetings, Hermann Am 23.12.2011 19:17, schrieb Steve Checkoway: > > On Dec 23, 2011, at 10:03 , Hermann Weber wrote: > >> I found out something: >> >> The more frames I decode before the frame that I actually want to >> decode, the better the quality becomes. >> For

Hello, >From: "Hermann Wecke" <hermann@wecke.com> >Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern => >extensions.conf >Date: 8 May 2004 22:03:57 +0000 > >Is it possible to strip some numbers from the *end* of a number? > >I know that ${EXTEN:1} will remove 1 position from the beggining... but >how to remove N numbers from