search for: phone0

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 Dial("Phone/phone0", "Phone/phone0) in...

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:11...

...ial 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 Phone/phone0(1) (Provisional) Stopping retransmission (but retaining packet) on '1bf802c37038448b795a4dc8300e0627@12.218.223.74' Request 102: Found (Provisional) Stopping retransmission (but retaining packet) on '1bf802c37038448b795a4dc8300e0627@12.218.223.74' Request 102: Found -- SIP/fwdpu...

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?...

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. would the sip debug help? ill post if you need Thanks Dave Packham I have this...

...llerID,${FWDCIDNAME} exten => _2431XXXXXXXXXX,2,Dial,IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/**${EXTEN},60,r) exten => _2431XXXXXXXXXX,3,Playback(invalid) exten => _2431XXXXXXXXXX,4,Hangup exten => _2431XXXXXXXXXX,103,Bus The output is: -- Executing SetCallerID("Phone/phone0", ""Chris"") in new stack -- Executing Dial("Phone/phone0", "IAX2/FWDNUM:FWDPASS@iax2.fwdnet.net/**2431XXXXXXXXXX") in new stack -- Called FWDNUM:FWDPASS@iax2.fwdnet.net/**2431XXXXXXXXXX -- Call accepted by 65.39.205.121 (format ulaw) --...

...ring with Asterisk and want to setup it up (at the moment) as simple answering machine different message will be plaid between 8am - 5pm and different after hours. I'll add extensions later on. I've Supira 3000. I was reading wiki page and to my understanding I have to place device=/dev/phone0 in phone.conf (actually it is there so I uncommented it). Though I couldn't find any instructions how to setup correctly the device phone0 in /dev directory. Any hints or good links would be appreciated. What other configuration file do I need to edit at this point? -- #Joseph

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 afterwards... The kernel is 2.4.21, compiled without ISA PnP support and with (presumably older) ixj and phonedev drivers enabled as mod...

...y fresh install of 1.2.40 using sample configs; in dialplan (uncommented extension ext. 1265); in 'phone.conf' uncommented the following: [interfaces] mode=dialtone format=slinear echocancel=medium context=local txgain=100% rxgain=1.0 device => /dev/phone0 -- dialplan works, ext. 1265 rings, has two-way audio, call progress tones heard, sends dtmf, can dial out, but has no dialtone. 2. temporary 'update' (bininstall) install of 1.2.40 overlaying existing CVS HEAD installation: -- existing dialplan works, Phone/phone0 works as...

...ompile 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 error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/aster...

...NG[1209214400]: file.c:537 ast_readaudio_callback: Failed t o write frame -- Playing 'vm-login' (language 'en') May 21 04:25:10 WARNING[1209214400]: app_voicemail.c:2748 vm_execmain: Couldn't read username == Spawn extension (demo, 4000, 2) exited non-zero on 'Phone/phone0' -- Hungup 'Phone/phone0' or if I do a EXEC festival test I get May 20 17:43:05 WARNING[1217602880]: chan_phone.c:417 phone_read: Error reading: Resource temporarily unavailable May 20 17:43:05 WARNING[1217602880]: app_festival.c:181 send_waveform_to_channel : Null frame == h...

I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. ----------...

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 phone, then I can continue playing with the system. Starting asterisk with -vvvvcd gets me to the CLI prompt. This is what happens on incoming calls: Asterisk Ready. *CLI> We have caller ID Urgent h...

...m card), now that I have found out that Phonejack should use the Linux Telephony Devices and its configuration file is phone.conf, but the question is I do not know how to configure the extensions.conf to call out from sip client to PSTN line. I tried using the exten => _9NXXXXXXX,1,Dial(Phone/Phone0/${EXTEN}) I keep getting the unable to register channel phone error message. Can anyone please paste out a sample extensions.conf file that uses the Quicknet PhoneJack card. Thanks. Yahoo! Messenger - Log on with your mobile phone! -------------- next part -------------- An HTML attachment was...

I have a Quicknet Linejack. Actually, I want to make a standalone(no internet connections) call to my asterisk box running GNU/Linux. When I call the system it says: -- Executing BackGround("Phone/phone0", "demo-congrats") in new stack NOTICE[14349]: File channel.c, Line 1212 (ast_set_write_format): Unable to find a path from 2 to 1 WARNING[14349]: File file.c, Line 553 (ast_streamfile): Unable to open demo-cong rats (format 1): No such file or directory == Spawn extension (default...

...ny Interface ; ; Configuration file ; [interfaces] ;mode=immediate ;mode=fxo mode=dialtone ;format=g723.1 ;format=slinear format=ulaw silencesupression=no ; ; List all devices we can use. Contexts may also be specified ; context=local context=default ; txgain=100% rxgain=100% device => /dev/phone0

On 6/15/20 2:19 PM, Luca Bertoncello wrote: > Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere: > > Hi Jeff, > >> We are working on a product to analyze pcap files of VoIP calls.  So far >> it does a reasonable job of analyzing the frequency distribution of >> packets in both directions, pointing out which direction packet loss / >> bad jitter occurs.  If you can

...en => 200,Dial(IAX/aullox_gdl@200.64.35.58/200@aullox) 200.64.35.58 is my ip 200 is tehe xtension and aull is the context and at veracruz i have: [aullox] exten => 200,1,Wait,2 exten => 200,2,Playback(transfer,skip) ; "Please hold while..." exten => 200,3,Dial(Phone/phone0) exten => 200,104,Voicemail(u1234) but, I get this when I call gdl->ver *CLI> NOTICE[10251]: File chan_iax.c, Line 3628 (socket_read): Rejected connect attempt from 200.67.99.127, request 'exten=200;language=en;context=aullox;username=aullox_gdl;formats=2;capability=65283;version=1;a...

...terisk won't answer pstn ring", but I didn't find any follow up to get it working. My asterisk setup is like this: iptelephony:/etc/asterisk# cat phone.conf | grep -v \; [interfaces] format=slinear echocancel=medium silencesupression=yes context=local mode=dialtone device => /dev/phone0 mode=fxo device => /dev/phone1 phone0 operates fine. With regards to the following piece of code, I do not see the error message. if (mode == MODE_FXO) { if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN)) ast_log(LOG_DEBUG, &...

...Or, is that command simply a way of viewing the channels on a T1/E1 if you have that type of interface? If the latter, is there an equiv. command to list the single FXO interfaces? If the answer is yes, all configured devices should have something in channels, how do you map a linejack from /dev/phone0 to channel 1? I can see in the zapata.conf file where channels may be set, but not in phone.conf. I'm trying to get * to answer inbound calls. * reports seeing the caller ID for the incoming call, but never seems to try and actually answer the call. My "show channels" is also empty,...