search for: troxlinux

Hi list, I install dahdi-linux successfully with the module of oslec for the echo, but when I specify it in the system.conf the echo canceller oslec it shows me errors: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) I see that the echo cancellers is supported: mg2, kb1, sec2, and sec because oslec is not supported?, but he has support to compile it with dahdi_linux! best

hello list, I am trying to arm an ivr for schedule of office and outside of office [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include =>outsideofschedule|*|sat-sun|*|* [scheduleofservice] exten

Hi list, I try to install asterisk on vps server , but fails when I want to install dahdi [root at shark dahdi-linux-2.6.3-rc1]# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing

somebody knows some application web that allows me to call to my internal extensions of my asterisk, example click to call. I was proving the click to call of this example but it doesn't work http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html greeting rickygm

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

hi list , is having problems when sending a fax with hylafax and a card sangoma A200D, when he sends it arrives to the destination but it paginates appears in white this is my log Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms, 20 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM WWW.SOFT-SWITCH.ORG spandsp/ Aug 20

Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX, generate tables in a couple of files in the folder realtime / mysql , voicemail_messages.sql and voicemail.sql the connection with mysql and odbc works well isql asterisk useradmin xxx +---------------------------------------+ | Connected! | | | |

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Hi list, I need some help to improve my cdr, now in my company are asking me how to know which of my phone numbers are most used when receiving calls from the PSTN and incoming the IVR was thinking about using userfield field, and I'm trying to do, I have at the moment 4 channel DAHDI ; DAHDI CHANNEL 3=23XXXXX6 context=in callerid=asreceived group=1 signalling=fxs_ks channel => 3

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

the schedule of my server this configured with -6:00, and this correct one with the normal hour of my country, I made the change but I don't work me [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include

when I make a call to the pstn it shows me this error: aximum retries exceeded on transmission 9d4a24f8-b673756b at 192.168.10.19 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 11 20:35:34] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging up call 9d4a24f8-b673756b at 192.168.10.19 - no reply to our critical packet (see doc/sip-retransmit.txt). bug? voicemail same

Hi is a list could be off topic ;) , but someone has installed the latest version of app_swift on centos 6 for asterisk 1.8 I'm trying to make with this manual, but have had no success http://www.cepstral.com/en/support/telephony/faq?os=linux&section=getting-started gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6 -D_AST_VER_1_8 -c -o app_swift.o app_swift.c

Hi list, I'm trying to attach a Avaya with Asterisk, call the extension 3241 to 1042 belonging to avaya, but only sounds rings and when I pick up the phone keeps ringing 08-14-13 07:26:17 AM-856ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 3241 at 172.16.8.40, To = 1083 at 172.16.8.5, Response = 100 Trying 08-14-13 07:26:17 AM-857ms Line = 18, Channel =