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...5 to leave a message, press * to return to the main menu. 5. Extension: 6000 6. Please leave your message after the tone, when done, please hangup or press the pound key (it segfaults right after it says pound key) 7. Segmentation fault == Using SIP RTP CoS mark 5 -- Executing [*61 at line1:1] VoiceMailMain("SIP/line1-01d646b0", "6001") in new stack -- <SIP/line1-01d646b0> Playing 'vm-password.gsm' (language 'en') DTMF begin '1' received on SIP/line1-01d646b0 DTMF begin ignored '1' on SIP/line1-01d646b0 DTMF end '1'...

...r, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1: 192.53.103.108 time server2: 192.53.103.104 time format: 1 date format: 5 sip line1 auth name: AWirth sip line1 password: AWirth493 sip line1 user name: 47 sip line1 display name: AWirth sip line1 screen name: "Alfred Wirth" sip line1 proxy port: 5060 sip line1 registrar ip: 192.168.8.7 sip line1 registrar port: 5060 sip line1 outbound proxy port: 5060 sip line1 registra...

What are these dns queries for? I'd like to disable them but I cant find any obvious reference to them in the asterisk source. I'm running Asterisk 1.4.21.2 I call voicemail and immediately hang up: I called from a sip client called line1, but I have no idea where 08c5b9e0 is coming from... 14-Jun-2009 12:37:07.926 queries: info: client 127.0.0.1#41105: query: line1-08c5b9e0.shadowmage.org IN A + 14-Jun-2009 12:37:08.002 queries: info: client 127.0.0.1#38238: query: line1-08c5b9e0.shadowmage.org IN A + 14-Jun-2009 12:37:15.716 quer...

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac", "SIP/0114477xxxxxxxx@simpleconnect-sip|60|r") in new stack -- Called 0114477xxxxxxxx@simpleconnect-sip May 12 18:45:14 WARNING[24525]: chan_sip.c:694 retrans_pkt: Ma...

...tracey mailbox= context=internal callerid="tracey" <868> <aastra.cfg> live dialpad: 1 suppress dtmf playback: 1 time server1: pro5.pbzinc.loc sip dial plan: "X+#|XX+*" sip proxy ip: phone.pbzinc.loc sip proxy port: 5060 sip dtmf method: 1 sip out-of-band dtmf: 0 sip line1 auth name: tracey sip line1 user name: 868 sip line1 display name: "Tracey" sip line1 screen name: "Tracey" sip line1 proxy ip: phone.pbzinc.loc sip line1 proxy port: 5060 sip line1 dtmf method: 1 sip use basic codecs: 1 handset list version: 1 handset1 name: "Tracey's...

...em in two pastes, which I''ll paste below. ----1 ### This was already saved in my database, pre 1.8.3 -> 1.8.4, and pre Rails 1.1RC1 >> c.lawsuit.customer_service_location => "--- !ruby/object:Address \nattributes: \n created_on: 2006-02-24 12:30:19\n city: Alex City\n line1: 441 Saint Bernard Drive\n zip: \"35086\"\n line2: ''''\n id: \"124\"\n contact_detail_id: \"128\"\n detail_key: Home Address\n state: AL" ### And I created this YAML object from the same data. >> ay => "--- !ruby/object:Addres...

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Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqbala at improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqbala at improvise:/etc/asterisk$ cat extensions.conf [default] exten => s,1,Answe...

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002 Asterisk settings: users.conf:...

...ing. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel => 1 signalling = fxs_ks context = line1 channel => 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => s,4,BackGround(demo-congrats) exten => s,...

Dear all, Can someone from Netherlands who has SPA3000 send me the Regional & Line1 settings. I'm not sure what is wrong but I can call from asterisk to phone attached to spa300 FXO, but not the other way. I tested three phones: siemens gigaset, tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk, other two just dial but nothing happens. I guess this has so...

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congesti...

...label, and thus I have been using "/n" to break it into two lines. However, to make it technically correct for publication, I also need to use superscript in the label. For example: par(oma=c(0,0,2,0),mar=c(5,6,0.25,2),lheight=1) plot(1:10, ylab="14C-glyphosate line1\n line2") will provide the text in two lines as I would like it. However, I am trying to keep those same line breaks when using expression() to get my superscript number. This will not work, as it aligns the "14C" section with the bottom line of the expression making little sense...

Hello, Let's say we have the following function: foo <- function(x) { line1 <- x line2 <- 0 line3 <- line1 + line2 return(line3) } And that we want to change the second line to be: line2 <- 2 How would you do that? The two ways I know of are either to use fix(foo) And change the function. Or to just write the function again. Is...

...clear the the line called "25" is smaller at both temperatures than the other lines. but when I check the contrasts with summary(mod1) I get: Fixed effects: area ~ line * temp Value Std.Error DF t-value p-value (Intercept) 165417.32 3102.751 336 53.31312 <.0001 line1 2631.71 4387.952 8 0.59976 0.5653 line2 -2603.27 4387.952 8 -0.59328 0.5694 line3 -4667.61 3102.751 8 -1.50435 0.1709 temp -2614.39 142.160 8 -18.39045 <.0001 line1:temp 96.39 201.045 8 0.47946 0.6444 line2:temp 95.74 201.045...

...ok some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel => 1 signalling = fxs_ks context = line1 channel => 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => s,4,BackGround(demo-congrat...

...uot; } else colo<-"black" txt <- format(c(r, 0.123456789), digits=digits)[1] txt <- paste(prefix, txt, sep="") cex.cor <- 0.8/strwidth(txt) text(0.5, 0.5, txt, cex = abs(cex.cor * r),col=colo) } panel.line1 <- function (x, y, col = par("col"), bg = NA, pch = par("pch"), cex = 1) { points(x, y, pch = pch, col = colour, bg = bg, cex = cex) ok <- is.finite(x) & is.finite(y) if (any(ok)) abline(lsfit(x,y,intercept = TRUE), col= "r...

Thanks Geoff, it's becoming more clear to me now... > So, assuming it does, you could try: > aumix -w r [yann@raglou yann]$ aumix -w R [yann@raglou yann]$ aumix -q vol 100, 100 pcm 100, 100 speaker 0, 0 line 0, 0, P mic 4, 0, P cd 0, 0, P igain 0, 0, P line1 0, 0, R phin 0, 0, P phout 0, 0 video 0, 0, P No "R" on the pcm line, so that is probably the problem : my soundcard can only record line1... > Having said all that, it's probably far simpler to use Oddcast for XMMS if > you're playing the music with XMMS anyway. Get it f...