Re: [asterisk-users] Does Asterisk support DNIS?
Arriving late to this discussion, sorry if this has already been mentioned but DNIS and ANI can be variable length without confusion if the sender uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced Star ANI Star DNIS Star allows the receiver to identify the two values unambiguously and to find the trailing boundary (when the 3rd * has been received). We have a Channelized Voice T1 from a long distance provider that is set up this way into our non-Asterisk PBX where the provider sends us ANI as the full originating phone number and DNIS as the last 4 digits. So the DTMF string seen by our PBX for someone calling one of our toll-free numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313, would be *8085551313*4567*. The PBX parses this string and uses the last 4 digits DNIS to route the call from the T1 trunk group to the proper internal extension or hunt group. Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and DNIS? --Ron On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote: It will do so automatically if it is working. Asterisk will stuff those digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if you are expecting 3 digits. Until recently we had DID service from our telco on an EM Wink channelized voice T-1. The above is what we did. David Ruggles wrote: Yuan (and Matt), Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated with trying to configure Asterisk for what (in a proprietary PBX) is normally one of the easiest parts of configuration. With a wink start T1 the DNIS digits are transmitted in-band. The Network goes off hook, the PBX winks (goes off hook for 200ms) and then the network sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it answers the call (goes off hook). So you would tell the PBX to look for x number of digits and then after getting that number of digits it will answer the call. I have the Sangoma A101 configured for wink start, but I can't find anything that says how to specify the number DNIS digits to expect. If the PBX answers the call instead of just winking, the DTMF tones will be transmitted during the call which is what seems to be happening here. For more specific information a good overview of the wink start process can be found here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080 1123bb.shtml#topic2a Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a wink start T1? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:57 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Does Asterisk support DNIS? Matt already replied to your other posting of similar content. I'm also a bit confused. Do you mean you have observed that Asterisk is brought into the intended context, but start to react to digits in DNIS one after another? If so, can you estimate the interval Asterisk stays in each extension? If this is true, it seems to suggest that your provider is sending DNIS as a DTMF string after Asterisk has answered the call. Isn't this a bit weird? What does the card's manual say about DNIS (with wink start)? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support DNIS?
On Sun, 18 Feb 2007, Matt wrote: Why would the card care? This would be something you'd take care of in your dialplan. Right, the card wouldn't care. So does Asterisk know about how to send and receive delimited ANI and DNIS through a channelized voice T1? --Ron On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote: Arriving late to this discussion, sorry if this has already been mentioned but DNIS and ANI can be variable length without confusion if the sender uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced Star ANI Star DNIS Star allows the receiver to identify the two values unambiguously and to find the trailing boundary (when the 3rd * has been received). We have a Channelized Voice T1 from a long distance provider that is set up this way into our non-Asterisk PBX where the provider sends us ANI as the full originating phone number and DNIS as the last 4 digits. So the DTMF string seen by our PBX for someone calling one of our toll-free numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313, would be *8085551313*4567*. The PBX parses this string and uses the last 4 digits DNIS to route the call from the T1 trunk group to the proper internal extension or hunt group. Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and DNIS? --Ron On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote: It will do so automatically if it is working. Asterisk will stuff those digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if you are expecting 3 digits. Until recently we had DID service from our telco on an EM Wink channelized voice T-1. The above is what we did. David Ruggles wrote: Yuan (and Matt), Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated with trying to configure Asterisk for what (in a proprietary PBX) is normally one of the easiest parts of configuration. With a wink start T1 the DNIS digits are transmitted in-band. The Network goes off hook, the PBX winks (goes off hook for 200ms) and then the network sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it answers the call (goes off hook). So you would tell the PBX to look for x number of digits and then after getting that number of digits it will answer the call. I have the Sangoma A101 configured for wink start, but I can't find anything that says how to specify the number DNIS digits to expect. If the PBX answers the call instead of just winking, the DTMF tones will be transmitted during the call which is what seems to be happening here. For more specific information a good overview of the wink start process can be found here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080 1123bb.shtml#topic2a Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a wink start T1? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:57 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Does Asterisk support DNIS? Matt already replied to your other posting of similar content. I'm also a bit confused. Do you mean you have observed that Asterisk is brought into the intended context, but start to react to digits in DNIS one after another? If so, can you estimate the interval Asterisk stays in each extension? If this is true, it seems to suggest that your provider is sending DNIS as a DTMF string after Asterisk has answered the call. Isn't this a bit weird? What does the card's manual say about DNIS (with wink start)? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support DNIS?
On Sun, 18 Feb 2007, Matt wrote: BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all the intelligence you need to do it right. You may not have that option. For example, you want to split a T1 from a legacy PBX to 12 channels to a proprietary IVR system and 12 channels to an Asterisk box. Can't do that with with PRI and a single T1 because you only have one control channel. --Ron On 2/18/07, Matt [EMAIL PROTECTED] wrote: Why would the card care? This would be something you'd take care of in your dialplan. On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote: Arriving late to this discussion, sorry if this has already been mentioned but DNIS and ANI can be variable length without confusion if the sender uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced Star ANI Star DNIS Star allows the receiver to identify the two values unambiguously and to find the trailing boundary (when the 3rd * has been received). We have a Channelized Voice T1 from a long distance provider that is set up this way into our non-Asterisk PBX where the provider sends us ANI as the full originating phone number and DNIS as the last 4 digits. So the DTMF string seen by our PBX for someone calling one of our toll-free numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313, would be *8085551313*4567*. The PBX parses this string and uses the last 4 digits DNIS to route the call from the T1 trunk group to the proper internal extension or hunt group. Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and DNIS? --Ron On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote: It will do so automatically if it is working. Asterisk will stuff those digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if you are expecting 3 digits. Until recently we had DID service from our telco on an EM Wink channelized voice T-1. The above is what we did. David Ruggles wrote: Yuan (and Matt), Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated with trying to configure Asterisk for what (in a proprietary PBX) is normally one of the easiest parts of configuration. With a wink start T1 the DNIS digits are transmitted in-band. The Network goes off hook, the PBX winks (goes off hook for 200ms) and then the network sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it answers the call (goes off hook). So you would tell the PBX to look for x number of digits and then after getting that number of digits it will answer the call. I have the Sangoma A101 configured for wink start, but I can't find anything that says how to specify the number DNIS digits to expect. If the PBX answers the call instead of just winking, the DTMF tones will be transmitted during the call which is what seems to be happening here. For more specific information a good overview of the wink start process can be found here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080 1123bb.shtml#topic2a Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a wink start T1? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:57 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Does Asterisk support DNIS? Matt already replied to your other posting of similar content. I'm also a bit confused. Do you mean you have observed that Asterisk is brought into the intended context, but start to react to digits in DNIS one after another? If so, can you estimate the interval Asterisk stays in each extension? If this is true, it seems to suggest that your provider is sending DNIS as a DTMF string after Asterisk has answered the call. Isn't this a bit weird? What does the card's manual say about DNIS (with wink start)? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DevKitLite compiles but won't load modules or run asterisk
I'm trying to install on a 233Mz PII running RH7.1 kernel 2.4.2 The astinstall script pulled files from CVS ok and compiled with no warnings. When I try to run modprobe wcfxo and modprobe wcusb I get the following identical set of error messages: 15. Creating sample environment...SKIPPED = Installation Completed! [EMAIL PROTECTED] asterisk-devkit-lite]# modprobe wcfxo /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol add_wait_queue_Re67723dc /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol __pollwait_Rf55e743f /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol proc_mkdir_R7e88f9ad /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol register_chrdev_R0ec734c6 /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol remove_proc_entry_R6be8391a /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol create_proc_entry_R4231d820 /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol remove_wait_queue_R9ce3 /lib/modules/2.4.2-2/misc/zaptel.o: insmod /lib/modules/2.4.2-2/misc/zaptel.o failed /lib/modules/2.4.2-2/misc/zaptel.o: insmod wcfxo failed [EMAIL PROTECTED] asterisk-devkit-lite]# [EMAIL PROTECTED] asterisk-devkit-lite]# modprobe wcusb /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol add_wait_queue_Re67723dc /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol __pollwait_Rf55e743f /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol proc_mkdir_R7e88f9ad /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol register_chrdev_R0ec734c6 /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol remove_proc_entry_R6be8391a /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol create_proc_entry_R4231d820 /lib/modules/2.4.2-2/misc/zaptel.o: unresolved symbol remove_wait_queue_R9ce3 /lib/modules/2.4.2-2/misc/zaptel.o: insmod /lib/modules/2.4.2-2/misc/zaptel.o failed /lib/modules/2.4.2-2/misc/zaptel.o: insmod wcusb failed [EMAIL PROTECTED] asterisk-devkit-lite]# Running ztcfg returns the following: [EMAIL PROTECTED] asterisk-devkit-lite]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 5: Unable to open master device '/dev/zap/ctl' [EMAIL PROTECTED] asterisk-devkit-lite]# Running asterisk after creating the 2 sample mailboxes returns a bunch of activity that I've reduced to the banner and lines containing warning: [EMAIL PROTECTED] asterisk-devkit-lite]# asterisk -vvvc Parsing '/etc/asterisk/asterisk.conf': Not found (No such file or directory) Asterisk CVS-12/23/03-01:41:05, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = WARNING [1024]: File res_musiconhold.c, Line 508 (moh_register): Unable to open pseudo channel for timing... Sound may be choppy. WARNING [3076]: File res_musiconhold.c, Line 165 (spawn_mp3): Found no files in '/var/lib/asterisk/mohmp3' WARNING [3076]: File res_musiconhold.c, Line 227 (monmp3thread): unable to spawn mp3player WARNING [1024]: File chan_iax2.c, Line 6044 (load_module): Unable to open IAX timing interface: No such device WARNING [1024]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now WARNING [9226]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable WARNING [1024]: File config.c, Line 579 (cfg_process): No '=' (equal sign) in line 90 of extensions.conf WARNING [1024]: File chan_zap.c, Line 628 (zt_open): Unable to open '/dev/zap/channel': No such device WARNING [1024]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING [1024]: File loader.c, Line 407 (load_modules): Loading module chan_zap.so failed! I had previously installed asterisk from CVS before the Digium hardware arrived and was able to run asterisk from the console, dial into the the sample environment ( but not leave voicemail ) and run the IAX demo at Digium. Can anyone tell me what may be running amiss here? Also, is there a script or makefile target that will fully un-install asterisk, zaptel, zapata and libpri so that I can try again? Thanks, Ron -- Ronald FoxEmail: [EMAIL PROTECTED] Diagnostic Laboratory Services, Inc. Phone: (808) 589-5172 Honolulu, Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users