[asterisk-users] Conversations Mix
Hello, I have a problem with an adit 600 and a T400P card. This equipment was in a shelf for 2 years and when we connected an install it asterisk everything worked fine. But then we started receiving complaints that a person pick up their phone and will hear some other conversation. It happens with internal calls and outbound calls too. The channel bank has 4 fxs cards and one fxo card. From the console it seems that for some phones asterisk does not register the hook off nor the dialing. Any hints how to solve this ? regards,Iván ___ --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] ISDN DID
Hello , Thank you for every response. It was the telcos fault. They told me they were sending it, but they wer not. regards, ia On 8/10/05, Johann Steinwendtner [EMAIL PROTECTED] wrote: There is no called party ie but sending complete ie included in the setup message. Hence, it tries to terminate. Best regards Hans Paul Belanger schrieb: Where are your calls coming from? Are you connected to the Telco or PBX? PB Panitaxx wrote: Hi, thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ 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) [18 03 a9 83 8d] 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 ] [1e 02 84 83] Progress Indicator (len= 4) [ 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) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8d] 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 ] -- Accepting call from '915451900' to 's' on channel 0/13, span 1 Asterisk*CLI -- Executing Playback(Zap/13-1, vm-intro|noanswer) in new stack Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: PROGRESS (3) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Playing 'vm-intro' (language 'es') Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8d] 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 ] [1e 02 81 82] Progress Indicator (len= 4) [ 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) ] -- Playing 'vm-goodbye' (language 'es') Asterisk*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing NoOp(Zap/13-1, ) in new stack -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/13-1' Asterisk*CLI On 8/9/05, jj [EMAIL PROTECTED] wrote: What does pri debug span 1 show? On Aug 9, 2005, at 5:02 PM, Panitaxx wrote: Hello, I have an ISDN PRI E1. For some reason I am not receiving
Re: [Asterisk-Users] Playback before Answer
It was after an upgrade? This E1 is new. We have 10 E1 R2. This is our first pri. I am starting to suspect that audio suppresion is done at the telco. ia On 8/11/05, Trevor Peirce [EMAIL PROTECTED] wrote: Panitaxx wrote: I have an ISDN PRI E1. I want to send an audio before answering, I am using noanswer option in playback app but all the audio is muted before the answer. I would like to play this audio. I have a T1 and a few months ago my ability to playback audio before answering ceased. Now I just get silence as well. I imagine a code change is responsible but I haven't had the time to go back and figure out exactly when this happened. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN DID
Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback before Answer
Hello, I have an ISDN PRI E1. I want to send an audio before answering, I am using noanswer option in playback app but all the audio is muted before the answer. I would like to play this audio. Regards, ia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
Hi, thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ 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) [18 03 a9 83 8d] 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 ] [1e 02 84 83] Progress Indicator (len= 4) [ 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) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8d] 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 ] -- Accepting call from '915451900' to 's' on channel 0/13, span 1 Asterisk*CLI -- Executing Playback(Zap/13-1, vm-intro|noanswer) in new stack Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: PROGRESS (3) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Playing 'vm-intro' (language 'es') Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8d] 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 ] [1e 02 81 82] Progress Indicator (len= 4) [ 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) ] -- Playing 'vm-goodbye' (language 'es') Asterisk*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing NoOp(Zap/13-1, ) in new stack -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/13-1' Asterisk*CLI On 8/9/05, jj [EMAIL PROTECTED] wrote: What does pri debug span 1 show? On Aug 9, 2005, at 5:02 PM, Panitaxx wrote: Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
I tried that it says Extension 's' in context 'primario' from '915451900' does not exist. Rejecting call on channel 0/14, span 1 thanks, ia On 8/9/05, Damon Estep [EMAIL PROTECTED] wrote: How many digits is your pri provider sending in the setup message? It needs to match your dilaplan, ie if they are sending 4 you need 4 digit extensions or some other monkey business to translate. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Panitaxx Sent: Tuesday, August 09, 2005 4:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN DID Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
yes. overlapdial=yes. On 8/9/05, Matt Fredrickson [EMAIL PROTECTED] wrote: On Tue, Aug 09, 2005 at 05:20:15PM -0500, Panitaxx wrote: thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ 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) [18 03 a9 83 8d] 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 ] [1e 02 84 83] Progress Indicator (len= 4) [ 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) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received I don't see a number specified here. Do you have overlapdial=yes enabled? -- Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
To a telco, specifically Colombia's Telecom On 8/9/05, Paul Belanger [EMAIL PROTECTED] wrote: Where are your calls coming from? Are you connected to the Telco or PBX? PB Panitaxx wrote: Hi, thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ 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) [18 03 a9 83 8d] 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 ] [1e 02 84 83] Progress Indicator (len= 4) [ 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) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8d] 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 ] -- Accepting call from '915451900' to 's' on channel 0/13, span 1 Asterisk*CLI -- Executing Playback(Zap/13-1, vm-intro|noanswer) in new stack Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: PROGRESS (3) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Playing 'vm-intro' (language 'es') Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8d] 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 ] [1e 02 81 82] Progress Indicator (len= 4) [ 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) ] -- Playing 'vm-goodbye' (language 'es') Asterisk*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing NoOp(Zap/13-1, ) in new stack -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/13-1' Asterisk*CLI On 8/9/05, jj [EMAIL PROTECTED] wrote: What does pri debug span 1 show? On Aug 9, 2005, at 5:02 PM, Panitaxx wrote: Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
Re: [Asterisk-Users] ISDN DID
It did not work. thanks anyway On 8/9/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: You want it to be no. Panitaxx wrote: yes. overlapdial=yes. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFCR2 Venezuela with libunicall
I have a setup for a 30 incoming channels with telcel. The incoming is R2, they told me the outgoing is MF not R2. If the other channels are fxo, you should change your zaptel.conf so you can use zapata.conf and comment out those channels on unicall.conf. ia On 5/20/05, Andres Maduro [EMAIL PROTECTED] wrote: Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P E1/T1. I have loaded in the followin order the kernel modules: 1. zaptel 2. wcte11xp 3. wcfxs The E1 is configured as this: the first 15 channels are for incoming calls using MFC/R2 Venezuela protocol, the last 15 channels are configured as a normal analog line, you pick up the channel and hear dial tone, you then only need to send dtmf to place a call (how I configure this channels ??), I guess with fxo loop start in the E1 channel ?? The problem I have is that I can't bring up asterisk, when I try, I receive the following error on the logs about being unable to load chan_zap: May 20 12:01:05 ERROR[13693]: Signalling requested is FXO Kewlstart but line is in R2 Signalling signalling May 20 12:01:05 ERROR[13693]: Unable to register channel '1' May 20 12:01:05 WARNING[13693]: chan_zap.so: load_module failed, returning -1 May 20 12:01:05 WARNING[13693]: Loading module chan_zap.so failed! I am including zaptel.conf, zapata.conf and unicall.conf to see if you can help me. zttool shows both cards in an OK status. -- zaptel.conf start -- # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 # Span 2: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=32 fxoks=33 fxsks=34 fxsks=35 # Global data loadzone= us defaultzone = us --- zaptel.conf end --- --- zapata.conf start --- ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf ; (extensiones) signalling = fxo_ls channel = 32-33 signalling = fxs_ls channel = 34-35 --- zapata.conf end --- --- unicall.conf start --- ; ; Unicall telephony channel driver ; ; Sample configuration file ; [channels] ; ; Default language ; ;language=en ; ; Default context ; context=default ; ; Whether or not to use caller ID ; usecallerid=yes ; ; Whether or not to hide outgoing caller ID ; hidecallerid=no ; ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user) ; Mostly use with FXS ports ; ;restrictcid=no ; ; Support Caller*ID on Call Waiting ; callwaitingcallerid=yes ; ; Support three-way calling ; threewaycalling=yes ; ; Support flash-hook call transfer (requires three way calling) ; transfer=yes ; ; Support call forward variable ; cancallforward=yes ; ; Whether or not to support Call Return (*69) ; callreturn=yes ; ; Enable echo cancellation ; Use either yes, no, or a power of two from 32 to 256 if you wish ; to actually set the number of taps of cancellation. ; echocancel=yes ; ; Generally, it is not necessary (and in fact undesirable) to echo cancel ; when the circuit path is entirely TDM. You may, however, reverse this ; behavior by enabling the echo cancel during pure TDM bridging below. ; echocancelwhenbridged=yes ; ; In some cases, the echo canceller doesn't train quickly enough and there ; is echo at the beginning of the call. Enabling echo training will cause ; asterisk to briefly mute the channel, send an impulse, and use the impulse ; response to pre-train the echo canceller so it can start out with a much ; closer idea of the actual echo. Value may be yes, no, or a number of ; milliseconds to delay before training (default = 400) ; ;echotraining=yes
[Asterisk-Users] MGCP transfer and CDR
Hello: I have an asterisk deployment with 15 MGCP extensions and 30 incoming E1 R2 channels. Calls are received by a receptionist queue (which only member is the receptionist phone). The receptionist then transfers (using hook flash) the call to one of the extensions. I want to be able to have two separate records in the cdr, one for the call between the receptionist and the unicall channel and another that starts when the call is transferred. I have tried to use forkcdr in the extensions context, seems to work randomly. Also call records show that the destination is between the receptionist and the extension or between the unicall channel and the extension. Sometimes minutes are billed to the receptionist, sometimes to the call between extension. I want cdr to show dst and src correctly. Any Ideas? thanks, ia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
Push it with enough force, it will come in. On Sun, 20 Feb 2005 11:51:05 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same questionhow do I go from the RJ11 standard analog to the RJ45 on the TDM400P card? I'd appreciate any response. thx chuks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP - Unicall
Hi: I have installed a call center with 15 extensions. I used an E1 with R2 signalling for incoming calls. The calls are received through clarent cpg-101 (aka DLINK 104s). I have a lot of trouble with transfers. Usually an extension receives the incoming calls from the e1 then the call is transferred (using flash key) to another extension. Behavior of transfers seem to be erratic sometimes the transfer goes OK. But sometimes when call is transferred the person in the extension hears the caller from the E1 but the caller does not hears the person in the extension. I suspect that the problem is the mgcp part (either asterisk or the device) because the e1 calls go ok before they are transferred. I search the list and found posts about firmware upgrades. But i cannot find ther firmware upgrade and i am afraid that since i am using the original clarent device flashing with the dlink firmware may render it useless. any ideas? thanks, ia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
