similar to: ATA-186 Dialplans

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with

I am currently a college student looking for ways to prepare myself for any Server Administration job once I get out of college. I've been going back and worth between Freenode IRC channels (#linux and ##windows) trying to decide what to learn. On one hand, we have Windows Server 2008 R2, people in #linux keep on telling me to just go ahead and use it because "samba can't provide

Hi, I am running glm logit regressions with R and I would like to test a linear combination of coefficients (H0: beta1=beta2 against H1: beta1<>beta2). Is there a package for such a test or how can I perform it otherwise (perhaps with logLik() ???)? Additionally I was wondering if there was no routine to calculate pseudo R2s for logit regressions. Currently I am calculating the pseudo R2

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hi. If I use a Cisco as a PSTN termination GW and need to route all incoming isdn calls to my asterisk and all outgoing calls from asterisk via the cisco out to pstn, how do I do that ? in the cisco I have this: dial-peer voice 1 pots destination-pattern [0-9]T no digit-strip direct-inward-dial port 3/0:D ! dial-peer voice 50 voip destination-pattern [0-9] voice-class codec 1 session

Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex

Friends, Last week, Hans Petter Selansky alerted us of a potential security issue in all releases of Asterisk. In fact, it doesn't involve the code, but the most common way to construct dialplans. If you have something like this in your Asterisk, you need to update your dialplans: [incoming-from-voip] exten => _X., 1, dial(SIP/${EXTEN}) Many VoIP protocols support a large character set,

Dear list, dear Cecilia and Daniel, sorry for coming in ten days late, I've been very busy lately so I came across this email only today. This is just to make some points clearer re: fixed effects and r2 in package 'plm', to both you and the list. In particular, to make you aware of some additional features. Please see my comments below, with '##'. Best, Giovanni

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get

On Sat, Jan 11, 2014 at 12:12:46AM +0100, Hilko Bengen wrote: > --- > images/README | 14 ++++++++++++ > images/mkzero/Makefile | 9 ++++++++ > images/mkzero/mkzero.c | 59 +++++++++++++++++++++++++++++++++++++++++++++++++ > images/special | Bin 0 -> 8192 bytes > 4 files changed, 82 insertions(+) > create mode 100644 images/mkzero/Makefile >

<digitmap =20 dialplan.digitmap=3D"[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxx= x x|[2-9]xxxT" dialplan.digitmap.timeOut=3D"3|3|3|3|3|3"/> Don't think it's been modified from the original supplied. ...brig -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

Hello! I'd like to Flash() a modem line (BRI) with Asterisk. It is a passive ISDN-card connected to a hardware PBX. I use ISDN4Linux. I recognised that unfortunately the Flash() application flashes Zap devices only. Now I am wondering how I could flash Modem/ttyI0. The source code chan_modem.c doesn't contain anything about flashing a modem line. So I tried to simply put the AT-command

Asterisk 1.2.12.1 The "*" key and the "0" key do not seem to be detected in my dialplan. I am using "a" and "o" to detect them. It simply "falls thru" to my "i" where it says, "I am sorry that is not a valid extension". joe a.