Re: [asterisk-users] Problem with call transfer from one server to another server
Server A ( which contain pri line) *chan_dahdi.conf* [channels] group=1 context=outbound usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes faxdetect=both callprogress=no progzone=in pulsedial=yes ;busydetect=yes callreturn=yes echocancel=no echocancelwhenbridged=no rxgain=0.5 txgain=0.5 relaxdtmf=yes callgroup=1 pickupgroup=1 pritimer = t309,6000 immediate=no switchtype=euroisdn context=outgoing signalling=pri_cpe pridialplan=unknown group=1 channel = 1-15,17-31 context=outgoing signalling=pri_cpe pridialplan=unknown group=2 channel = 32-46,48-62 context=outgoing signalling=pri_cpe pridialplan=unknown group=3 channel = 63-77,79-93 context=outgoing signalling=pri_cpe pridialplan=unknown group=4 channel = 94-108,110-124 *sip.conf* [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.53.197/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.53.197 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=60 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid=EXT3 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4004] type=friend context=outbound defaultuser=4004 secret=4004 callerid=EXT4 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani mi...@enterux.in wrote: Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here. Mitul On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com wrote: Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() Regards Akhilesh -- _ -- 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 -- _ -- 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] Problem with call transfer from one server to another server
Server B(child server) *chan_dahdi.conf* [trunkgroups] [channels] group=1 context=outbound usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes faxdetect=both callprogress=no progzone=in pulsedial=yes ;busydetect=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.5 txgain=0.5 callgroup=1 pickupgroup=1 pritimer = t309,6000 immediate=no switchtype=euroisdn context=outgoing signalling=pri_cpe pridialplan=unknown group=1 channel = 1-15,17-31 context=outgoing signalling=pri_cpe pridialplan=unknown group=2 channel = 32-46,48-62 context=outgoing signalling=pri_cpe pridialplan=unknown group=3 channel = 63-77,79-93 context=outgoing signalling=pri_cpe pridialplan=unknown group=4 channel = 94-108,110-124 *Sip.conf* [general] pear=type context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.14.112/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.14.112 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes Disallow=all Allow=all ;Allow=ulaw ;Allow=gsm Canreinvite=no ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani mi...@enterux.in wrote: Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here. Mitul On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com wrote: Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() Regards Akhilesh -- _ -- 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 -- _ -- 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] error cant write to function ODBC_DEVICES
Hi all
asterisk 1.8.23
I have odbc all setup to mysql but cant figure out why the dialplan wont
write to the odbc function
fubc_odbc.conf
[DEVICES]
dsn=device-conn;dsn in res_odbc not odbc.ini
readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM
call INNER JOIN devices ON call.id = devices.callId WHERE deviceNumber = '${
SQL_ESC(${ARG1})}'
writesql=insert into voted (callId,callNum,city,deviceId,SerialNum,
serverResponse) values (${VAL1},${VAL2},${VAL3},${VAL4},${VAL5},${VAL6}
extension.conf
the relevant line
same = n,set(ODBC_DEVICES()=${callid},${call},1,${deviceid},${num},${
serverupdate})
when sending the values from the cli using odbc write it works ok
reading from the dialplan works ok
i tried sending plain values without variables
but from the dialplan gives me a error cant write to function ODBC
_DEVICES
happy to hear any ideas
--
_
-- 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] IVR integration with third party application Help wanted
Hi list, I hope this isn't in error but if it is I apologize. I have a small project request on hand where the clients want their customers to be able to dial in to conduct business over the phone in a completely automated manner. From my limited understanding this looks a lot like a call center where one has to build some sort of proxy that understands their business logic and that can report stuff back to asterisk which then reports it back to the customer. I have little or no understanding of AGI or related architecture, I just know how to setup asterisk as a call manager. if anyone would be willing to help me out to understand what needs doing i'd be very grateful. Thanks for listening, and hope to hear from you soon! -- _ -- 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] Call parking issue with Cisco SPA phone
I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext = 700 parkpos = 701-702 context = parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700@10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701@10.0.1.103 Key4: fnc=blf+sd+cp;sub=702@10.0.1.103 Parking using these works smoothly. I answer the incoming call, press Key2 to park the call. Call is parked, Key3 turns red showing there is a parked call. If I want to unpark the call, I hit Key3 and the call is unparked. My problem happens when Key3 and Key4 are idle (no parked calls): I answer the incoming call and without first parking the car, I hit one of the idle keys (Key3 or Key4), the phone sends a REFER message, and the incoming call hangs up. I'm trying to find out why the call hangs up and how to prevent that? Thanks, Matt -- _ -- 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] IVR integration with third party application Help wanted
On Sun, 20 Oct 2013, Notify Me wrote: I have a small project request on hand where the clients want their customers to be able to dial in to conduct business over the phone in a completely automated manner. From my limited understanding this looks a lot like a call center where one has to build some sort of proxy that understands their business logic and that can report stuff back to asterisk which then reports it back to the customer. We need a lot more detail like what does 'conduct business over the phone in a completely automated manner' mean? Are customers calling in and ordering ink cartridges? To me, 'build some sort of proxy that understands their business logic' does not sound like a 'small project.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] What linux distro most popular for Asterisk
Slackware here. On Thu, Oct 17, 2013 at 8:57 PM, Tiago Geada tiago.ge...@gmail.com wrote: debian wheezy compiled asterisk from source On 18 October 2013 00:27, Andrew Furey andrew.fu...@gmail.com wrote: [Apologies, top-posting, Gmail, yadda yadda] As with a lot of software, I suspect the best answer is whichever distro YOU are most comfortable with. You're the one who has to support it, after all... Just my 2c. Andrew On Thursday, 17 October 2013, Rusty Newton wrote: On Tue, Oct 15, 2013 at 11:58 PM, Michelle Dupuis mdup...@ocg.ca wrote: Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts I don't have any numbers, but I watch the issue tracker a lot and I see pretty much CentOS, Debian and Ubuntu. Which seems to match what everyone else is saying on this thread. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett -- _ -- 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 -- _ -- 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] Call parking issue with Cisco SPA phone
I'll answer my own question: Setting Keep Referee When REFER Failed to Yes on the Cisco phone seems to do the trick. From: mistral9...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 20 Oct 2013 11:56:17 -0400 Subject: [asterisk-users] Call parking issue with Cisco SPA phone I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext = 700 parkpos = 701-702 context = parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700@10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701@10.0.1.103 Key4: fnc=blf+sd+cp;sub=702@10.0.1.103 Parking using these works smoothly. I answer the incoming call, press Key2 to park the call. Call is parked, Key3 turns red showing there is a parked call. If I want to unpark the call, I hit Key3 and the call is unparked. My problem happens when Key3 and Key4 are idle (no parked calls): I answer the incoming call and without first parking the car, I hit one of the idle keys (Key3 or Key4), the phone sends a REFER message, and the incoming call hangs up. I'm trying to find out why the call hangs up and how to prevent that? Thanks, Matt -- _ -- 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] IVR integration with third party application Help wanted
Hi and thanks for the response, much appreciated! From what I'm being told, its some sort of pension (financial) organization, customers are supposed to be able to manage their accounts over the phone. That's all I know so far. On Oct 20, 2013 5:57 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 20 Oct 2013, Notify Me wrote: I have a small project request on hand where the clients want their customers to be able to dial in to conduct business over the phone in a completely automated manner. From my limited understanding this looks a lot like a call center where one has to build some sort of proxy that understands their business logic and that can report stuff back to asterisk which then reports it back to the customer. We need a lot more detail like what does 'conduct business over the phone in a completely automated manner' mean? Are customers calling in and ordering ink cartridges? To me, 'build some sort of proxy that understands their business logic' does not sound like a 'small project.' -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] IVR integration with third party application Help wanted
On Sun, 20 Oct 2013, Notify Me wrote:
From what I'm being told, its some sort of pension (financial)
organization, customers are supposed to be able to manage their accounts
over the phone.
That still leaves a lot unsaid. How far does 'manage their accounts'
extend? I can see 'look up my balance' or 'how much money have you lost
for me in the last month' but 'sell this stock and buy that stock' seems
to be outside the 'liability' tolerance of the financial institutions I've
dealt with.
I'd advise writing this application as an AGI to keep all the ugly details
and database access out of the dialplan.
Depending on the depth of the application, I could see 1 AGI for
authentication and then a separate AGI ('I do one thing at a time, I do
it very well, and then I move on*') for each function.
*) Major Charles Winchester, MASH, 1977.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- 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
