similar to: Asterisk extension not found problem...

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing calls are done through a macro as follows: [macro-diallink2voip] exten => s,1,Dial(SIP/${ARG1}@link2voip-sw2,120) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Hangup exten => s-CONGESTION,1,Dial(SIP/${ARG1}@link2voip-sw1,120) exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1) exten =>

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

Hello, I'm having trouble getting finding a company that provides DID in Western Canada. More specifically in Edmonton, Alberta. We have tried getting in contact with Link2Voip and Calgary Telecom but neither seems to be answering their phones or email. I would appreciate it if anyone can point me in the right direction. Thank you, Nelson

I tried this [dongle0] ;audio=/dev/ttyUSB1 ; tty port for audio connection; no default value ;data=/dev/ttyUSB2 ; tty port for AT commands; no default value ; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345 ; imei and imsi must contain exactly 15 digits ! ; imei/imsi discovery is available on Linux only

Dear List, I frequently use the data() function to load csv files (with separator ";") into R session, typically data(myfile) loads myfile.csv from my working/data directory into R. Now, in 1.4.0 version, everything works as expected, but with one difference: The values readed in older versions in "num" mode are now readed as "int" mode, converting the values

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

Dear R users, I'd like to make this data rem.y = c(-1,0,2,4,5) from y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5). That is, I need to remove repeated values. Here is my code, but I don't think it is efficient. How could I improve this? #------------------------------------------------------------------------ y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5) n=length(y) for (i in 1:n) #

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

hi, i use xapian to index a txt file, it's size is 268M. i take each line as a document, and each line has two field like 13445511 | 111115151. the recored size is 10000000. the XAPIAN_FLUSH_THRESHOLD set 1000000. it takes 1026544ms to index the file, it is more slower than lucene. The lucene speed is about 40000 records per second. code: try { Xapian::WritableDatabase

Hi Dave, >> context=openbts >> callerid=4735202222 >I see you are using OpenBTS. To my understanding, OpenBTS does not >support caller ID, so I don't think it can work. >But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Hi list, I have a user, referenced by his IMSI (IMSI208300618462231), who is assigned to extension 2111 in /etc/asterisk/extensions.conf and sip.conf (see below).

Dear "someone who can help", I am having a problem with SAMBA and SWAT (although I'm not sure SWAT is part of the problem...) I get two instances of "smbd" when ever it is started on this system. From SWAT I can only stop it ONE time. After that, each time I attempt to stop the SMBD daemon from SWAT it simply starts TWO more instances of the daemon (or something

I want to know about asterisk and openBTS If anybody made any test and experience... Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/cbd08740/attachment.htm

Does anybody know anyone who offers encrypted IAX termination at reasonable rates? I googled, searched the WIKI, but didn't find a whole lot of information. Thanks, David

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple