[Asterisk-Users] Supervision Issue With Asterisk/Sipura/Talkn
I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn→Sipura→Asterisk/FXO Card. When someone calls in and get’s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the port, rings the phone plays messeges sends the caller to vm or whatever and the caller hangs up, * doesn’t see this disconnect and keeps the port open. I had configured the Sipura to do: Idle Polarity: Caller Conn Polarity: Reverse Callee Conn Polarity: Reverse Caller Conn Polarity: Reverse And Caller Conn Polarity: Forward Callee Conn Polarity: Forward Caller Conn Polarity: Forward Doesn't seem to make a different * still holds the call up and doesn't release it, my context looks like: [incoming-talkn] ; exten = s,1,SetMusicOnHold,random exten = s,2,Zapateller(answer|nocallerid) exten = s,3,NoOp exten = s,4,PrivacyManager exten = s,5,LookupCIDName exten = s,6,LookupBlacklist exten = s,7,Background(pls-wait-connect-call) ;exten = s,8,Dial(Zap/4,15,Ttr) exten = s,9,Answer exten = s,10,Wait(1) exten = s,11,Macro(vmessage,2101) exten = s,105,Macro(vmessage,6000) exten = s,107,Macro(vmessage,6000) Has anyone else had this issue and can offer a solution to my problem? I am using a current download of asterisk as of today 6/4/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] intercept ringing phone
Hi did you get my last email? Thanks. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boater Sent: Wednesday, May 26, 2004 8:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] intercept ringing phone I havent used it with Asterisk, but I think I have seen the feature. On my Definity, the feature is called Call Pickup and you add extensions to the call pickup group. -Original Message- From: Michael George [mailto:[EMAIL PROTECTED] Sent: Wed 5/26/2004 3:33 PM To: [EMAIL PROTECTED] Cc: Subject: [Asterisk-Users] intercept ringing phone I am almost done with my initial configuration of asterisk. One of the things which our BizFone does and I have not been able to emulate with asterisk is the ability to grab a ringing extension. e.g. I am talking to someone in an office with extension 204. I can hear a phone ringing and I know it's mine (extension 201). I want to be able to answer that extension from 204. What application can I use to implement that? -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ^+$Rf)+-^+$RXb+rXb+r+-w-z P (]j+aXb+r+-w-z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Blacklist
I am attempting to write in incoming context for calls.
1. If the caller id is given and it is not black listed it will Playback a
greeting and then right the phone or go to voicemail under busy or
unavailable conditions
2. If no caller id is given, then Privacy Manager will ask for the number.
I am testing 6145551212 to see if the black list will work
3. If a caller id is given, and it is blacklisted (in the blacklist db) I
would like for it to go to Playback/black-list-blocked message
The db shows:
asterisk*CLI database show blacklist
/blacklist/1010987/18887975686number: 1
/blacklist/name/number : 1
/blacklist/unlisted/6145551212: 1
asterisk*CLI
exten = 2129,1,Wait(1)
exten = 2129,2,Zapateller(answer|nocallerid)
exten = 2129,3,NoOp
exten = 2129,4,PrivacyManager
exten = 2129,5,LookupBlacklist
exten = 2129,6,Dial(Zap/4,5,Ttr)
exten = 2129,7,Answer
exten = 2129,8,Wait(1)
exten = 2129,9,Playback(personal/hello)
exten = 2129,10,Playback(personal/i-am-not-in-at-the-moment)
exten = 2129,11,VoiceMail2(u${EXTEN})
exten = 2129,12,Hangup
exten = 2129,102,VoiceMail2(b${EXTEN}) ; Busy Voicemail if extension is
busy
exten = 2129,106,Playback,personal/black-list-blocked
exten = 2129,108,Wait(2)
exten = 2129,110,Hangup
When I dial my test extension of 2129, I get:
asterisk*CLI
-- Starting simple switch on 'Zap/7-1'
-- Disabling Caller*ID on Zap/7-1
-- Executing Wait(Zap/7-1, 1) in new stack
-- Executing Zapateller(Zap/7-1, answer|nocallerid) in new stack
-- Executing NoOp(Zap/7-1, ) in new stack
-- Executing PrivacyManager(Zap/7-1, ) in new stack
== Parsing '/etc/asterisk/privacy.conf': == Parsing
'/etc/asterisk/privacy.conf': Found
-- Playing 'privacy-unident' (language 'en')
-- Playing 'privacy-prompt' (language 'en')
-- Playing 'privacy-thankyou' (language 'en')
-- Changed Caller*ID to Privacy Manager 6145551212
-- Executing LookupBlacklist(Zap/7-1, ) in new stack
-- Executing Dial(Zap/7-1, Zap/4|5|Ttr) in new stack
-- Called 4
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Nobody picked up in 5000 ms
-- Hungup 'Zap/4-1'
-- Executing Answer(Zap/7-1, ) in new stack
-- Executing Wait(Zap/7-1, 1) in new stack
-- Executing Playback(Zap/7-1, personal/hello) in new stack
-- Playing 'personal/hello' (language 'en')
-- Executing Playback(Zap/7-1, personal/i-am-not-in-at-the-moment)
in new stack
-- Playing 'personal/i-am-not-in-at-the-moment' (language 'en')
-- Executing VoiceMail2(Zap/7-1, u2129) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/9' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
It goes to the unavailable voice mail box.
According to the documentation and my understanding:
LookupBlacklist: Looks up the Caller*ID number on the active channel in the
Asterisk database (family 'blacklist'). If the number is found, and if there
exists a priority n + 101, where 'n' is the priority of the current
instance, then the channel will be setup to continue at that priority level.
Otherwise, it returns 0. Does nothing if no Caller*ID was received on the
channel.
Example: database put blacklist name/number 1
Could someone tell me what I am doing wrong that it won't go to Priority 106
and Playback black-list-blocked.
Would someone share their context that is using blacklist to show me how
they are doing it?
Thanks.
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[Asterisk-Users] Adding another X100P Card
History: 1. Added X100P to my system 2. Added Sipura 3. Added TDM400P (2 port) Worked fine so far 4. Now I want to add an additional X100P My question is...is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answer from a specific Number
I was trying to only have Asterisk only answer with extension when it came from a specific Caller-id number, it works from all numbers with my example below: include = parkedcalls exten = s,1,Answer exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(vm-extension) Modified to: include = parkedcalls exten = s,1,Answer/6145551212 exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(vm-extension) I thought be adding the /6145551212 after the Answer above would do what I wanted but it doesn't. Could someone advise me of the right example? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured
History: 1. Added X100P to my system 2. Added TDM400P (2 port) Worked fine so far 3. Now I want to add an additional X100P Is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Sipura 2000 Transmit Issues? No Sound being passed to caller
I have a similar set up, I don't have a separate sip phone, but I have the same exact problem with the line 1. I don't know if my config files aren't right, but I can't transfer between exts yet, but my issues is with line one on an incoming call from an X100P as well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Higgins Sent: Sunday, January 25, 2004 3:54 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Example of TDM20B
I am trying to find an example of how to set up my FXS Station Card in my Asterisk. I have (1) XP100P I have (1) tdm20B (2 Port FXS) Could someone tell me if this is correct? /etc/zaptel.conf fxsks=1 fxoks=2 fxoks=3 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] ; language=en ; ;X100P Port 1 context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 ; ; FXS Port 1 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes ; ;FXS Port 2 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes /etc/asterisk/extensions.conf [local] exten = 2203,1,SetMusicOnHold,loud exten = 2203,2,Dial(Zap/2,15,Ttr) exten = 2203,102,Voicemail(2203) exten = 2203,Hangup exten = 2204,1,SetMusicOnHold,loud exten = 2204,2,Dial(Zap/2,15,Ttr) exten = 2204,102,Voicemail(2203) exten = 2204,Hangup Should this be enough for me to get dial tone on my FXS Cards? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Sipura 2000 Transmit Issues? No Sound beingpassed to caller
I added the music on hold feature. I answer on line 1, flash for a sec and come back and transmission both way is fine, just can't answer initially. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: Sunday, January 25, 2004 11:55 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound beingpassed to caller same here, when i recive an incoming call from x100p to line 1 on sipura, i can hear them but people can't hear me im using 1.0.24 on my firmware Miguel On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/X100 - Sipura Configuration
Title: Asterisk/X100 - Sipura Configuration I have a Sipura behind my Asterisk Box. I have a X100 card in the box. Calls are coming in ok. When I try to configure my extensions for dialing out, I can dial 9X and then get a fast busy. I followed an example from moxilla on the Spirua and Asterisk before but I am questioning the dialing plan in the Sipura causing my problems. Has anyone got the above configuration working? I would like to be able to (for now) I will worry about voicepulse, fwd, iax, later. Just dial: 9XXX 9XX 91XX And have all traffic go out over my X100 from either line of my Sipura. Thanks in advance. Steve
[Asterisk-Users] Asterisk Sipura 2000
Title: Asterisk Sipura 2000 I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works fine. I just got my Sipura in the last day or so and I just built my first Asterisk. If anyone has any experience with Sipura, I would appreciate some help. I can send my config files of my Asterisk and Sipura if someone could help me. Thanks. Steve
[Asterisk-Users] Your config files
Is there an easy way to get all of your sound files, do you have ftp or just http? Thanks. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
