Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phpagi action based on outbound call user response
I am not sure about PHP AGI, but in general via AGI you can monitor the state of the call and so you can know when the call is over. l. 2013/4/17 Rahul R rahul...@gmail.com Hello List, In PHPAGI, I'm using the Astrisk Manager function send_request() to originate an outbound call. I want to execute the remaining PHP code after the call gets executed (depending on user input). But presently the call originates in a different context and asterisk executes the remaining code in parallel. Is there a way in which I can pause the code execution until the call is completed. Note: I wish to return to the context from which the call was originated and continue execution. Any help is greatly appreciated. -- Thanks Regards Rahul http://about.me/rahulr92 +919567607741 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
Hi, So , how to connect asterisk to whatapps ??Please advice .. Thank Regard/chui king man 寄件人︰ Lenz Emilitri lenz.lo...@gmail.com 收件人︰ isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年04月19日 (週五) 4:34 PM 主題︰ Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External call control for Asterisk
Not sure if that's what you are looking for, but I would think about having the dialplan call a web service (maybe using CURL) and passing account and current number. The system would reply with the number to actually dial, or none if blocked, and the maximum possible call length. Then it's all Asterisk (or turtles all the way down). 2013/4/10 Simon Green simon.c.gr...@gmail.com Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not really sure where to start. What I want to do is this: a PBX service ala FreePBX, but where call control is passed via SIP to an external service which will tell Asterisk: a) * Whether the call is allowed b) * Where to connect the call, if necessary (i.e. forced redirection to a C-party) c) * To disconnect the call at some time in future based on charging considerations (i.e. online charging) There is also the option of not using Asterisk at all, and simply using the other service directly, but Asterisk is much better suited to handling end-user devices. The external service does control logic only. Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
I'd start from https://github.com/venomous0x/WhatsAPI/blob/master/README.mdthat offerts PHP and Java APIS, both not hard to integrate with Asterisk. 2013/4/19 kingman chui chuiking...@yahoo.com.hk Hi, So , how to connect asterisk to whatapps ??Please advice .. Thank Regard/chui king man *寄件人︰* Lenz Emilitri lenz.lo...@gmail.com *收件人︰* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *傳送日期︰* 2013年04月19日 (週五) 4:34 PM *主題︰* Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] To enhance the voice quality of the SIP trunk
Hello; I have a SIP trunk with a service provider, the caller from landline or mobile is hearing us very well, but the agent who is sitting on the handset is not hearing well, the voice at the agent is not crystal (like he is talking from well or far deep place). Although the IP Phones are cisco 7942G and the used codec is g711ulaw (actually it gave better quality than g711alaw). If we increase the voice volume from the Cisco handset, the voice is becoming higher but more distortion, it is missing for the sharpness (clearance), we are hearing it like he is talking from well (from far and deep place). I am trying to fix this .. I requested the provider to check if the problem from his telephony card, but it is not doing any thing. What parameters can help me to fix this? trustrip, insecure, jitter, .. etc? Any of this can help? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC dialplan looping problem
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.netwrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each have their own PIN for the same bridge.
So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC
connector to parse the table.
Asterisk is connected and reads the rows as expected. The problem is that
if a user enters a PIN that is NOT in the table, asterisk goes crazy and
continues to loop forever.
Please have a look and tell me where I went so wrong.
Func_odbc.conf looks like this:
[PIN]
dsn=BRIDGE
mode=multirow
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}'
** **
extensions.conf section:
[infromhost] ;Host dials over SIP trunk exten=,1,Answer
exten=,n,Background(conf-getconfno)
exten=,n,WaitExten(10)
exten=,n,Hangup
exten=_XX,1,Set(GLOBAL(CONF_ID)=${EXTEN})
exten=_XX,n,GoTo(rooms,${EXTEN},1)
;
[rooms]
exten=_XX,1,Set(CONF_ID=${EXTEN})
exten=_XX,n,Background(conf-getpin)
exten=_XX,n,WaitExten(5)
exten=_XX,n,Hangup
exten=_1X,1,Goto(getpin,${EXTEN},1)
exten=_2X,1,Goto(getpin,${EXTEN},1)
exten=_3X,1,Goto(getpin,${EXTEN},1)
exten=_4X,1,Goto(getpin,${EXTEN},1)
exten=_5X,1,Goto(getpin,${EXTEN},1)
exten=_6X,1,Goto(getpin,${EXTEN},1)
exten=_7X,1,Goto(getpin,${EXTEN},1)
exten=_8X,1,Goto(getpin,${EXTEN},1)
exten=_9X,1,Goto(getpin,${EXTEN},1)
exten=i,1,Goto(getpin,${CONF_PIN},1)
;
[getpin]
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,n(loop_start),NoOp()
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)***
*
exten=_XX,n,Goto(loop_start)
;
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
same=n,Hangup()
;
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
**
Thank you!!
Pat Collins...
**
**
Change you [getpin] as below
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,GotoIf($[${ODBCROWS} 1]?cleanup,1)
exten=_XX,,Set(COUNTER=1)
exten=_XX,While($[${COUNTER} = ${ODBCROWS}])
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
exten=_XX,n,Set(COUNTER=$[${COUNTER + 1])
exten=_XX,n,EndWhile()
exten=_XX,n,ODBCFinish()
exten=_XX,n,Goto(cleanup,1)
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
Further your readsql should be like this.
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${ARG1})}'
You should have ${ARG1} instead of ${CONF_ID}
Hope this helps
--Satish Barot
Ahmedabad, India
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] ODBC dialplan looping problem
On Fri, Apr 19, 2013 at 5:59 PM, Satish Barot satish4aster...@gmail.comwrote:
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.netwrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each have their own PIN for the same bridge.
So, I put the list of users, PINs bridges into a MYSQL DB and used an
ODBC connector to parse the table.
Asterisk is connected and reads the rows as expected. The problem is
that if a user enters a PIN that is NOT in the table, asterisk goes crazy
and continues to loop forever.
Please have a look and tell me where I went so wrong.
Func_odbc.conf looks like this:
[PIN]
dsn=BRIDGE
mode=multirow
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}'
** **
extensions.conf section:
[infromhost] ;Host dials over SIP trunk exten=,1,Answer
exten=,n,Background(conf-getconfno)
exten=,n,WaitExten(10)
exten=,n,Hangup
exten=_XX,1,Set(GLOBAL(CONF_ID)=${EXTEN})
exten=_XX,n,GoTo(rooms,${EXTEN},1)
;
[rooms]
exten=_XX,1,Set(CONF_ID=${EXTEN})
exten=_XX,n,Background(conf-getpin)
exten=_XX,n,WaitExten(5)
exten=_XX,n,Hangup
exten=_1X,1,Goto(getpin,${EXTEN},1)
exten=_2X,1,Goto(getpin,${EXTEN},1)
exten=_3X,1,Goto(getpin,${EXTEN},1)
exten=_4X,1,Goto(getpin,${EXTEN},1)
exten=_5X,1,Goto(getpin,${EXTEN},1)
exten=_6X,1,Goto(getpin,${EXTEN},1)
exten=_7X,1,Goto(getpin,${EXTEN},1)
exten=_8X,1,Goto(getpin,${EXTEN},1)
exten=_9X,1,Goto(getpin,${EXTEN},1)
exten=i,1,Goto(getpin,${CONF_PIN},1)
;
[getpin]
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,n(loop_start),NoOp()
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)**
**
exten=_XX,n,Goto(loop_start)
;
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
same=n,Hangup()
;
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
**
Thank you!!
Pat Collins...
**
**
Change you [getpin] as below
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,GotoIf($[${ODBCROWS} 1]?cleanup,1)
exten=_XX,,Set(COUNTER=1)
exten=_XX,While($[${COUNTER} = ${ODBCROWS}])
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
exten=_XX,n,Set(COUNTER=$[${COUNTER + 1])
exten=_XX,n,EndWhile()
exten=_XX,n,ODBCFinish()
exten=_XX,n,Goto(cleanup,1)
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
Further your readsql should be like this.
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${ARG1})}'
You should have ${ARG1} instead of ${CONF_ID}
Hope this helps
--Satish Barot
Ahmedabad, India
:) I know I am not good at copy paste...
Change you [getpin] as below
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,n,GotoIf($[${ODBCROWS} 1]?cleanup,1)
exten=_XX,n,Set(COUNTER=1)
exten=_XX,n,While($[${COUNTER} = ${ODBCROWS}])
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
exten=_XX,n,Set(COUNTER=$[${COUNTER + 1])
exten=_XX,n,EndWhile()
exten=_XX,n,ODBCFinish()
exten=_XX,n,Goto(cleanup,1)
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
[asterisk-users] E911 Voip Trunking
During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip phone displaying caller name while on call
Hello, I've just realized that several phones display both caller name and number while ringing but when on call, caller name is not displayed anymore. Could you recommend a sip phone that still displays caller name during phone call ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External call control for Asterisk
AGI is your friend. check A2billing. On Fri, Apr 19, 2013 at 10:43 AM, Lenz Emilitri lenz.lo...@gmail.comwrote: Not sure if that's what you are looking for, but I would think about having the dialplan call a web service (maybe using CURL) and passing account and current number. The system would reply with the number to actually dial, or none if blocked, and the maximum possible call length. Then it's all Asterisk (or turtles all the way down). 2013/4/10 Simon Green simon.c.gr...@gmail.com Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not really sure where to start. What I want to do is this: a PBX service ala FreePBX, but where call control is passed via SIP to an external service which will tell Asterisk: a) * Whether the call is allowed b) * Where to connect the call, if necessary (i.e. forced redirection to a C-party) c) * To disconnect the call at some time in future based on charging considerations (i.e. online charging) There is also the option of not using Asterisk at all, and simply using the other service directly, but Asterisk is much better suited to handling end-user devices. The external service does control logic only. Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set google voice callerid as Unknown/Unavailable ?
I know you that GV won't respect CALLERID from motif, but is there a way have GV show Unknown on outgoing calls. I don't want to have people think my GV number is really my number. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
E911 does not follow the standard SIP RFC. That would be a good reason that they couldn't/wouldn't do it. Now that I say that I should qualify it and say NG911 (or ESINet) does not follow SIP RFC http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your county is not using standard SIP for E911, it just wouldn't be considered NG911. From: Chris Nighswonger Sent: Fri 4/19/2013 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] E911 Voip Trunking During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
Section 6.5.2 (v4 interface) of NENA's v2 Interim Voip Architecture Standard shows a ladder diagram of their SIP flow which seems to match standard SIP. Maybe I'm oversimplifying it? [1] http://c.ymcdn.com/sites/www.nena.org/resource/collection/2851C951-69FF-40F0-A6B8-36A714CB085D/NENA_08-001-v2_Interim_VoIP_Architecture_i2.pdf On Fri, Apr 19, 2013 at 2:51 PM, Terry Brummell te...@brummell.net wrote: E911 does not follow the standard SIP RFC. That would be a good reason that they couldn't/wouldn't do it. Now that I say that I should qualify it and say NG911 (or ESINet) does not follow SIP RFC http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your county is not using standard SIP for E911, it just wouldn't be considered NG911. -- *From:* Chris Nighswonger *Sent:* Fri 4/19/2013 11:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] E911 Voip Trunking During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic realtime + queues.conf Unresolved
Hi, I want queues.conf to be stored on a MySQL database using dynamic realtime. I am using asterisk 11.2.1 and MySQL 5.1.67, the MySQL database is hosted on another server but I can access the database via ODBC. I have created the following tables: SQL show tables; +-+ | Tables_in_asterisk | +-+ | QueueDialplanParameters | | Queues | | queue_log | | queue_member_table | +-+ SQLRowCount returns 4 4 rows fetched Each table has all of the required fields, some people suggested that queue_name is missing but it's not. I am getting the following errors from the asterisk console: node1*CLI queue show support support has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 60s No Members No Callers [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ''\' AND queue_name = ? ORDER BY interface' at line 1 (226) [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]... [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1527 odbc_obj_connect: Connecting asterisk [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector] [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ''\' AND queue_name = ? ORDER BY interface' at line 1 (226) [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]... [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1527 odbc_obj_connect: Connecting asterisk [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector] [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ''\' AND queue_name = ? ORDER BY interface' at line 1 (226) [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]... [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1527 odbc_obj_connect: Connecting asterisk [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector] [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ''\' AND queue_name = ? ORDER BY interface' at line 1 (226) [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]... [Apr 19 22:37:05] WARNING[18366]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1527 odbc_obj_connect: Connecting asterisk [Apr 19 22:37:05] NOTICE[18366]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector] The following error suggests that my syntax is incorrect, that syntax seems to be part of an SQL query. I do not have any SQL queries anywhere within my configuration. You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ''\' AND queue_name
Re: [asterisk-users] Dynamic realtime + queues.conf Unresolved
On Friday, April 19, 2013 1:45 PM, Tommy Cooper wrote: The following error suggests that my syntax is incorrect, that syntax seems to be part of an SQL query. I do not have any SQL queries anywhere within my configuration. iODBC or unixODBC? I'm sure that the query is being generated on-the-fly by res_config_odbc based on information it is being fed by app_queue; there is no .conf file that you can edit to see or modify the query. If Asterisk is doing something wrong when generating this query, it'll most likely have to be addressed in source somewhere. It would definitely be helpful to see what the *whole* query is that Asterisk is trying to execute. Try to enable trace/logging for your ODBC driver. If unixODBC, for example, add: [ODBC] Trace = Yes TraceFile = /tmp/sql.log ...to your odbcinst.ini file. This will dump detailed diagnostic information to the TraceFile, including the actual queries (in full) that are being executed. Just from what little we already know from the Asterisk logs, though, it *almost* looks like an escaping problem of some kind. What do you have the 'backslash_is_escape' option for your DSN set to in res_odbc.conf? Maybe try setting it to the opposite of whatever it's configured for now. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. On Fri, Apr 19, 2013 at 10:41 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
On Friday, April 19, 2013 5:35 PM, Warren Selby wrote: On Fri, Apr 19, 2013 at 10:41 AM, Chris Nighswonger wrote: During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. Indeed. 911ETC is who we use, and is another example. Even if you could peer directly with your county's PSAP, in the case of 911, I think it is a way better idea to go with one of these specialty SIP-based E911 providers, for the simple reason that even if you only sell VoIP service to people residing within your county, ATAs and VoIP phones are nomadic in nature: people are going to take them with them when traveling/on vacations, or maybe even use a soft phone with their account. This means that they are going to need to have the ability to update their physical E911 location, so that when they are away from home or away from the office, their 911 calls are directed to the correct local PSAP for their current location, and not back to their home county's PSAP. So, sure, you might be able to convince your county PSAP to peer with you directly via SIP, but it's not realistic to then go out and do the same for the other 8,000+ PSAPs in the U.S. (http://transition.fcc.gov/pshs/services/911-services/enhanced911/psapregistry.html) that one of your customers *might* be closest to at any given time, not to mention purchase and maintain the infrastructure, technology, and data needed to accurately geocode a physical address and then map it to a given PSAP. That's what these services are for: they deal with all of that, and all you have to do is send 911 calls to their SIP proxy, and they route it appropriately. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
On Fri, Apr 19, 2013 at 8:59 PM, Nathan Anderson nath...@fsr.com wrote: On Friday, April 19, 2013 5:35 PM, Warren Selby wrote: There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. Indeed. 911ETC is who we use, and is another example. Even if you could peer directly with your county's PSAP, in the case of 911, I think it is a way better idea to go with one of these specialty SIP-based E911 providers, for the simple reason that even if you only sell VoIP service to people residing within your county Actually we are not reselling service and the majority of our phones are stationary. The few mobile soft phones we run would not need 911 service since they also carry cell phones, the soft phones being mainly remote extensions. So it sounds like it is at least worth pursuing. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
