search for: chan_phon

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 box, etc. all work but there is...

Hi, I have a latest asterisk cvs sources on my box, the rest of enviroment is: gcc version 3.3.3 20040412 (Red Hat Linux 3.3.3-7) linux 2.6.7 and after checkout I get: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/local/src/avm/asterisk/asterisk/channels' make: *** [subdirs] Error 1 I have symlink "/usr/src/linx". It's a _fresh_ checkout...

...hread -lncurses -lm -lresolv -lssl gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC -c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 [root@sip asterisk-1.2.10]# Whats happening? I already tried with 3 di...

...gt; (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0'...

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 to register channel '/dev/phone0' What am I doing wrong? -- #Joseph

Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel 2.6.8.1-12mdk The error I get is: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/software/asterisk-1.0.0/channels' make: *** [subdirs] Error 1 Any ideas what might be wrong? ___________________________________________...

...98 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) O/S: Linux 2.4.27; IXJ driver for Linux Rev. 3.5, gcc 3.01 I applied the patch for dialtone as per my issue on 1.2.40. Build log contains no warnings for chan_phone. Build tested using install to DESTDIR=/tmp/asterisk; make samples; edited 'extensions.conf' to uncomment extension '1265' (Phone/phone0); added skinny extension to test Cisco 7920 (which worked except for audio stalling): exten => 1266,1,Dial(Skinny/1266 at 7920-1) exten =&gt...

Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [root@sss asterisk-1.2.13]# so I have ' make clean' ------------------ Then I downloaded the &...

...0 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.000000.3 wavelengths) Wavelength 1 (in samples): 16.66667 Minimum samples (1): 50 (3.000000.3 wavelengths) Need 200 samples Wrote ringtone.h [CC] chan_oss.c -> chan_oss.o [LD] chan_oss.o -> chan_oss.so [CC] chan_phone.c -> chan_phone.o [LD] chan_phone.o -> chan_phone.so [CC] chan_sip.c -> chan_sip.o [LD] chan_sip.o -> chan_sip.so [CC] chan_zap.c -> chan_zap.o chan_zap.c: In function ?process_zap?: chan_zap.c:11149: internal compiler error: Segmentation fault Please submit a full b...

...so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] => (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable t...

...Result from running ztool : Only one item on the list : OK HFC-S PCI A ISDN card 0 [NT] layer 1 AC Here are the last lines of my log (disregard the date and time, date and time on my server is wrong) ----------------- START LOG ------------------------ Aug 2 04:03:46 VERBOSE[1552]: [chan_phone.so]Aug 2 04:03:46 VERBOSE[1552]: [chan_phone.so] => (Linux Telephony API Support) Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': Found Aug 2 04:03:46 VERBOSE[1552]: == Registered channel t...

...ght with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) My sound card information: Vendor : Intel Corp. Model : 82801CA/CAM AC'97 Audio Controller Module : i810_audio A...

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 WARNING[1209214400]: file.c:537 ast_readaudio_callback: Failed t o write frame -- Playing 'vm-login'...

Hi, I've just been going through the various modules that are autoloaded to see what I need and what I don't and came across chan_phone.so which loads /etc/asterisk/phone.conf. I did a lookup on voip-info and google and came across this article in Linux Journal from 2001. Anyone know why it isn't being used much (from what I can tell) and what's happening with it today? Thanks, Mike http://www.linuxjournal.com/arti...

...> (OSS Console Channel Driver) == Console is full duplex May 28 13:59:42 WARNING[16384]: chan_oss.c:352 setformat: Requested 8000 Hz, got 48000 Hz -- sound may be choppy May 28 13:59:42 WARNING[147466]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) [chan_zap.so] => (Zapata Telephony w/PRI) May 28 13:59:42 WARNING[16384]: chan_zap.c:8030 setup_zap: Ignoring usercallerid May 28 13:59:42 WARNING[16384]: chan_zap.c:704 zt_open: Unable to specify channel 1: No such device or address May 28 13:59:42 ERRO...

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

...for backward compatibility. * autoconf.h is now included via -imacros on the commandline */ #include <linux/autoconf.h> #endif but it doesn't exist in the FC6 kernel tree for 2.6.18-1.2798.fc6 ... Likewise, attempting to build Asterisk-1.2.13 fails thusly: chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/asterisk/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [root@gate asterisk-1.2.13]# and the path: /usr/include/linux doesn't hav...

...shmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) after [root@voip root]# asterisk -vvvvcp command: [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jan 6 14:57:50 WA...

...-info]: [modules] autoload=yes noload => chan_agent.so noload => chan_iax2.so noload => chan_local.so noload => chan_mgcp.so noload => chan_modem.so noload => chan_modem_aopen.so noload => chan_modem_bestdata.so noload => chan_modem_i4l.so noload => chan_oss.so load => chan_phone.so noload => chan_sip.so noload => chan_skinny.so load => codec_a_mu.so load => codec_adpcm.so load => codec_alaw.so load => codec_g726.so load => codec_gsm.so load => codec_ilbc.so load => codec_lpc10.so load => codec_ulaw.so load => format_g726.so load => form...

...: [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered...