similar to: creating device=/dev/phone0

I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to

I want * to answer the phone when call comes-in. I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a command: mknod /dev/phone0 c 100 0 Though, when I start * I get: Parsing '/etc/asterisk/phone.conf': Found Sep 12 00:18:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' Sep 12 00:18:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Quick Summary: I am trying to join some Win2k (Service Pack 4) machines to a NT4 style domain with a Samba PDC. One machine successfully joins the Samba domain. One fails. Current configuration: SambaPDC: - FreeBSD 4.9-RELEASE - Samba 2.2.8a (from binary package) Host1 (Silv): - Windows 2000 SP4 (5.00.2195) Host2 (Plaid): - Windows 2000 SP4 (5.00.2195) All machines are

Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and reloading, it keeps the same ip open. The point is that i cant open all ips i need at the same

I'm trying to add an emergency dial to my context. However, when I try to dial it, I get caught in an endless loop. For debugging, I have pared out nearly all the control flow and just have ChanIsAvail() and Dial() called. Using two different extensions to call teh same number, I get two different actions by *. Here is the vvverbose output: -- Starting simple switch on

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

Hi, I'm having problems with setting up my ISA LineJack card on Linux machine... I've done everithing according to documentation available, by compiling the new ixj driver (v1.2.1), loading it, adding device node /dev/phone0 with major number 100 and minor number 0, adding aliases into the /etc/modules.conf: alias char-major-100 phonedev alias char-major-100-0 ixj and running depmod

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER

Hi guys, Does anyone have any ideas on my questions below ? Kindest regards David Wilson D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 MSN: dave@dcdata.co.za http://www.dcdata.co.za support@dcdata.co.za sales@dcdata.co.za KZN's first and only pure Linux solution provider ____________________________________________________ LinuxBox S.A.: Africa's shell

I am trying to setup GRE between two CentOS 4.5 boxes. I have tried several variations of what''s listed below, but none of them work. box1: modprobe ip_gre ip link set gre0 up ip tunnel add gretun mode gre local 66.1.1.161 remote 66.1.2.161 ttl 20 dev eth0 ip addr add dev gretun 10.253.253.1 peer 10.253.253.2/24 ip link set dev gretun up ip route add 10.2.0.0/16 via 10.253.253.2 box2:

Hello fellow asterisk users, I'm trying to use feature application defined in application map. it's defined as follows: lbxvml => 1,self/caller,Macro,Jump2Voicemail It's working properly when called party answers the call, but I'd like to have feature usable while call is still ringing in queue but this just does not work.. Is this a bug or feature? Is there a way to have

I saw your information on the internet and I have multiple opportunities available for QA Analysts. The opportunities are available in the Chicago area. A description of the opportunities can be viewed at www.parallelpartners.com. If you or someone you know is interested, please email me a copy of your resume. Thank you Raul Garcia Administrative Manager Parallel Partners 20 N. Wacker

Hello, All: What would it take to make ?iconv? portable? I ask, because I want to convert accented characters to vanilla ASCII, thereby converting, e.g., ?Ra?l? to ?Raul?, and Milan Bouchet-Valet suggested on R-help that I use 'iconv(x, ?", "ASCII//TRANSLIT?)?. This worked under Windows but failed on Linux and Mac. It?s part of the ?subNonStandardCharacters? function

Hi, I''d like to have translated languate-aware routes in my application, like: /projects/5/permissions => locale "PL" /projekty/5/pozwolenia => locale "PL" Is there any way to achieve it with RESTful routes? I have done it manually with named routes, but it''s overwhelming with over 300 routes :D Also tried namespace (yeah, didn''t work) and