Re: [asterisk-users] Conference problem
The CM is sending the BYE messages. Any ideas? Christian --- On Wed, 4/22/09, Martin asteriskl...@callthem.info wrote: From: Martin asteriskl...@callthem.info Subject: Re: [asterisk-users] Conference problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 22, 2009, 8:08 PM run a sip debug and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru cristi_icon...@yahoo.com wrote: Hello all, I have some issues with the MeetMe application. The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. The problem is that some users who are calling in from PSTN are getting disconnected from the conference room after a period of time. They can get in but after a while suddenly they are disconnected. The funny thing is that on the Asterisk CLI/logs no errors/retrans/etc. appeared. The Asterisk has no Zaptel hardware. All the necesary modules are installed. Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference problem
Hello all, I have some issues with the MeetMe application. The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. The problem is that some users who are calling in from PSTN are getting disconnected from the conference room after a period of time. They can get in but after a while suddenly they are disconnected. The funny thing is that on the Asterisk CLI/logs no errors/retrans/etc. appeared. The Asterisk has no Zaptel hardware. All the necesary modules are installed. Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 over NAT
Jeremy McNamara wrote: H.323 doesn't deal with NAT. Jeremy McNamara ayaz wrote: hello I am trying to configure h323 over NAT Can some one help me out Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users As far as I know there is one posibility if the equipment who is dealing with NAT can do so called H323 NAT... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
[EMAIL PROTECTED] wrote: I need to make a little IVR app and get/send the data into a MS-SQL database. As far as I know, it doesn't have driver for Linux. Anybody here already found here any workaround for this situation? Maybe, I can use an AGI interface to do that, maybe perl+ODBC? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users You can use the FreeTDS library to write your necesary code for authentication (and other queries) . A week or two ago I have wrote a module that worked well for me: cdr_sybase(for cdr insertion) and I was able to compile this using FreeTDS not only Sybase OpenClient. I didn't tested on MSSQL but I don't see a reason why not to work... Cristian Vasiliu [EMAIL PROTECTED] AccessNET ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Language
Mark Spencer wrote: Use setlanguage. Then organize the language files by directory e.g. /var/lib/asterisk/sounds/de /var/lib/asterisk/sounds/digits/de Also, say.c will have to be modified to support German style number handling. Mark On Sun, 27 Jul 2003, Peer Oliver schmidt wrote: Hi, I am in the process of recording voicemail prompts in german. How do I specify the language for the voice mail messages? I want to offer both language files, based on the calling party. Any and all help is greatly appreciated. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Why don't we (the ones who need this) contribute with code for supporting several languages into the saynumber application and say other in general. I have modify for romanian this code and it will be great if my work it will be used by ohers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer):Unable to forward voice
I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid = asreceived amaflags = billing usecallerid=yes overlapdial=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 ;context=h323 context=outbound signalling=pri_cpe channel = 63-77 ;channel = 79-93 ;group=5 ;context=h323 ;signalling=pri_cpe channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 extension.conf exten = ,1,Wait(1) exten = ,2,Answer exten = ,3,Playback(beep) exten = ,4,Dial(Zap/g3/0007352638) and I get this error You can see the output from pri debug span3: *CLI pri debug span 3 Enabled debugging on span 3 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 10309/0x2845) (Originator) Message type: SETUP (5) Sending Complete (len= 4) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (3) '0212318657' ] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ] -- Making new call for cr 10309 -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) -- Accepting call from '0212318657' to '' on channel 13, span 3 Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 43077/0xA845) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing Wait(Zap/75-1, 1) in new stack -- Executing Answer(Zap/75-1, ) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 43077/0xA845) (Terminator) Message type: CONNECT (7) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing Playback(Zap/75-1, beep) in new stack -- Playing 'beep' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 10309/0x2845) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing Dial(Zap/75-1, Zap/g3/0007894638) in new stack Making new call for cr 32771 Protocol Discriminator: Q.931 (8) len=45 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Display (len= 1) [ 1 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
[Asterisk-Users] Alphanumerical digits
Sorry Martin to bother you again! I have an ISDN flux with 100 numbers. The local PSTN is sending now the DNIS/DID (so they said!!!) (I have set the immediate=no in zapata.conf) but I have the same problem as before : NOTE : the number is alphanumeric-DID alphanumeric (I will make tests with numeric mumber!). WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 1 Into extensions.conf I have match everything: exten = _.,1,Wait(1) exten = _.,2,Answer exten = _.,3,Playback(beep) What should I expect into extension: DNIS or DID? Cristian Vasiliu ([EMAIL PROTECTED]) mail to:[EMAIL PROTECTED] web: cvasiliu.home.ro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alphanumerical digits
I see the following line into debug (pri debug span 1): 1. Progress Description: Calling equipment is non-ISDN. (3) ] 2. Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Into a ISDN that is working I got this : -- Making new call for cr 23274 -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) -- Extension '' in context 'outbound' Into this one I got only : -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) What parameters they not send! The complet debugging is this: Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 162/0xA2) (Originator) Message type: SETUP (5) Sending Complete (len= 4) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (1) '0788401422' ] -- Making new call for cr 162 -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32930/0x80A2) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 162/0xA2) (Originator) Message type: STATUS (125) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Protocol error, unspecified (111), class = Protocol Error (6) ] Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Present (6) -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 162/0xA2) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error (6) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] -- Processing IE 8 (Cause) -- Processing IE 30 (Progress Indicator) WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 1 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 162/0xA2) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Pub lic network serving the local user (2) Ext: 1 Cause: Recover on timer expiry (102), class = Protoco l Error (6) ] -- Processing IE 8 (Cause) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 32930/0x80A2) (Terminator) Message type: RELEASE COMPLETE (90) WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not f ound? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 1 Any ideas why this happening? ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Drop the call in 10min
[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Check the whentohangup value from cdr struct . Set the value to 600 . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Drop the call in 10min
[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ast_channel_setwhentohangup(chan,600); ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
[EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I was having the same problem because : 1 number for E1 and the local PTSN was not sending the DID to select the appropriate extension. Set the immediate=yes into zapata.conf and catch the call into s extension! Thanks to Martin Pycko ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on PRI channel : Call specified but not found!
Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 Cristian VASILIU mail: [EMAIL PROTECTED] www : cvasiliu.home.ro Soon sip address. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on PRI channel : Call specified but not found!
Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 In the zapata.conf configuration file I have the following line : immediate=no! I have to specify the with immediate=yes it is working even if it is asking for the s into the pri context! Cristian VASILIU mail: [EMAIL PROTECTED] www : cvasiliu.home.ro Soon sip address. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering on an zap device
How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 # Span 1 bchan=1-15,17-31 dchan=16 # Span 2 bchan=32-46,48-62 dchan=47 # Span 3 bchan=63-77,79-93 dchan=78 # Span 4 bchan=94-108,110-124 dchan=109 alaw=1-124 loadzone = nl defaultzone=nl zapata.conf: [channels] context=inbound switchtype=euroisdn signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=no rxgain=0.0 txgain=0.0 immediate=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=outbound signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 and extensions.conf: [general] static=yes writeprotect=yes ... [inbound] exten = 1,1,Ringing exten = 1,2,Wait,2 exten = 1,3,Playback(beep) exten = 1,4,Playback(agent-alreadyon) exten = _XXX,1,Ringing exten = _XXX,2,Wait,2 exten = _XXX,3,Playback(beep) exten = #,1,Hangup ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering on an zap device
Steven Critchfield wrote: First off, a E1 circuit is 32 channels, 1-32, 33-64, 65-96, 97-128. Also there is a D channel on 16 and 32 of each span. Then you need to add the D channel definition to your zapata.conf files. I believe all this was covered in the examples. On Tue, 2003-07-08 at 08:20, Cristi wrote: How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 # Span 1 bchan=1-15,17-31 dchan=16 # Span 2 bchan=32-46,48-62 dchan=47 # Span 3 bchan=63-77,79-93 dchan=78 # Span 4 bchan=94-108,110-124 dchan=109 alaw=1-124 loadzone = nl defaultzone=nl zapata.conf: [channels] context=inbound switchtype=euroisdn signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=no rxgain=0.0 txgain=0.0 immediate=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=outbound signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 and extensions.conf: [general] static=yes writeprotect=yes ... [inbound] exten = 1,1,Ringing exten = 1,2,Wait,2 exten = 1,3,Playback(beep) exten = 1,4,Playback(agent-alreadyon) exten = _XXX,1,Ringing exten = _XXX,2,Wait,2 exten = _XXX,3,Playback(beep) exten = #,1,Hangup ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122 ;channel = 2 ;callerid=CallerID Phone (256) 428-6123 ;callerid=CallerID Phone (630) 372-1564 ;callerid=CallerID Phone (256) 704-4666 ;channel = 3 ;callerid=Pac Tel Phone (256) 428-6124 ;channel = 4 ;callerid=Uniden Dead (256) 428-6125 ;channel = 5 ;callerid=Cortelco 2500 (256) 428-6126 ;channel = 6 ;callerid=Main TA 750 (256) 428-6127 ;channel = 44 It is the mapping between PTSN nr and channel numbers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering on an zap device
Steven Critchfield wrote: Please trim unnecessary lines from the post. On Tue, 2003-07-08 at 09:23, Cristi wrote: What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122 ;channel = 2 ;callerid=CallerID Phone (256) 428-6123 ;callerid=CallerID Phone (630) 372-1564 ;callerid=CallerID Phone (256) 704-4666 ;channel = 3 ;callerid=Pac Tel Phone (256) 428-6124 ;channel = 4 ;callerid=Uniden Dead (256) 428-6125 ;channel = 5 ;callerid=Cortelco 2500 (256) 428-6126 ;channel = 6 ;callerid=Main TA 750 (256) 428-6127 ;channel = 44 It is the mapping between PTSN nr and channel numbers? This is to make internal phones show callerID. Notice the Green phone, Black Phone, These are to specify what callerid to use when the channel is used to place a call. In my office and at home I have set each channels caller ID to specify the user at the end of the phone. This way you know who it is inside your office calling you when you see the callerid. Also, when used appropriately, you can specify either your main trunk number or specific DID numbers for the callerID display on a PRI line. This way you could set up a 9+ dialing for DID display, and 8+ for main line display. This is helpful when you are doing generic calling or specific calling. First let me thank you for all the answers that I got from the asterisk people! Another question : Let say that now I have a E1 ISDN into a 4E1 card and I don't have any errors :. What configuration I have to do get a call ? Tring a nr associated with the channel I got only a busy line!From where the asterisk know what nr to answer ? It is specified into the extension config file for the context associated with the channel group? == D-Channel on span 1 up == D-Channel on span 3 up -- B-channel 1 successfully restarted on span 3 -- B-channel 2 successfully restarted on span 3 -- B-channel 3 successfully restarted on span 3 -- B-channel 4 successfully restarted on span 3 -- B-channel 5 successfully restarted on span 3 -- B-channel 6 successfully restarted on span 3 -- B-channel 7 successfully restarted on span 3 -- B-channel 8 successfully restarted on span 3 -- B-channel 9 successfully restarted on span 3 -- B-channel 10 successfully restarted on span 3 -- B-channel 11 successfully restarted on span 3 -- B-channel 12 successfully restarted on span 3 -- B-channel 13 successfully restarted on span 3 -- B-channel 14 successfully restarted on span 3 -- B-channel 15 successfully restarted on span 3 -- B-channel 17 successfully restarted on span 3 -- B-channel 18 successfully restarted on span 3 -- B-channel 19 successfully restarted on span 3 -- B-channel 20 successfully restarted on span 3 -- B-channel 21 successfully restarted on span 3 -- B-channel 22 successfully restarted on span 3 -- B-channel 23 successfully restarted on span 3 -- B-channel 24 successfully restarted on span 3 -- B-channel 25 successfully restarted on span 3 -- B-channel 26 successfully restarted on span 3 -- B-channel 27 successfully restarted on span 3 -- B-channel 28 successfully restarted on span 3 -- B-channel 29 successfully restarted on span 3 -- B-channel 30 successfully restarted on span 3 -- B-channel 31 successfully restarted on span 3 -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1 -- B-channel 4 successfully restarted on span 1 -- B-channel 5 successfully restarted on span 1 -- B-channel 6 successfully restarted on span 1 -- B-channel 7 successfully restarted on span 1 -- B-channel 8 successfully restarted on span 1 -- B-channel 9 successfully restarted on span 1 -- B-channel 10 successfully restarted on span 1 -- B-channel 11 successfully restarted on span 1 -- B-channel 12 successfully restarted on span 1 -- B-channel 13 successfully restarted on span 1 -- B-channel 14 successfully restarted on span 1 -- B-channel 15 successfully restarted on span 1 -- B-channel 17 successfully restarted on span 1 -- B-channel 18 successfully restarted on span 1 -- B-channel 19 successfully restarted on span 1 -- B-channel 20 successfully restarted on span 1 -- B-channel 21 successfully restarted on span 1 -- B-channel 22 successfully restarted on span 1 -- B-channel 23 successfully restarted on span 1 -- B-channel 24 successfully restarted on span 1 -- B-channel 25 successfully restarted on span 1 -- B-channel 26 successfully restarted on span 1 -- B-channel 27 successfully restarted on span 1 -- B-channel 28 successfully restarted on span 1 -- B-channel 29 successfully restarted on span 1 -- B-channel 30 successfully restarted on span 1 -- B-channel 31 successfully restarted on span 1
[Asterisk-Users] Debug PRI!
This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for [EMAIL PROTECTED]:1 (Retry 2) -- Making new call for cr 32781 Protocol Discriminator: Q.931 (8) len=28 Call Ref: len= 2 (reference 13/0xD) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len= 5) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '10' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32781/0x800D) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 10, span 1 got hangup -- Hungup 'Zap/10-1' NOTICE[28690]: File pbx_spool.c, Line 195 (attempt_thread): Call failed to go through, reason 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
