Re: [asterisk-users] Anyone doing speech to text?
hi you can try this link http://zaf.github.io/asterisk-googletts/ 2015-08-26 19:15 GMT+01:00 Tech Support aster...@voipbusiness.us: All; I have a customer who is looking for a good speech to text solution, either open source or reasonably priced commercial product, I’m open to suggestions. Thanks; John V -- _ -- 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] Asterisk Inbound calls, multiple SIP accounts, calledID
what about
exten = s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
regards
2015-04-08 5:45 GMT+00:00 Dmitriy Serov serov@gmail.com:
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete register lines.
b) add option callbackextension=Company1 to Company1 friend section..
And in others with their names too.
or you can change /s to /Company1 in register line.
2. beautiful display of this information
a) add option setvar=fromCompany=Company1 to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
Maybe this will help?
Dmitiy.
08.04.2015 2:48, Andrew Galdes пишет:
Hi Dmitriy and others and thanks for your help so far.
The option match_auth_username=yes seems to have had no effect. From
my reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on. For example, the receptionist answers calls
for 8 different companies and would like the phone to display the company
name that she should announce to the caller.
Here is a more complete output of an incoming call. I've changed the SIP
numbers to Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
Verbosity is at least 12
asterisk*CLI
asterisk*CLI
asterisk*CLI
== Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*,
*thedid=NodePhonesip:compa...@sip.internode.on.net
sip%3acompa...@sip.internode.on.net*) in new stack
-- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
http://sip.internode.on.net*) in new stack
-- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2*) in new stack
-- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*,
*pseudodid=** sip:Company2*) in new stack
-- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,33,1:6*) in new stack
-- Goto (incoming,s,6)
-- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,88,1:7*) in new stack
-- Goto (incoming,s,7)
-- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,36,1:8*) in new stack
-- Goto (incoming,s,8)
-- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*,
*1?internal,36,1:9*) in new stack
-- Goto (internal,36,1)
-- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*,
*CALLERID(name)=SIP/**Company1**-0797*) in new stack
-- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*,
*SIP/36,20*) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/36
-- SIP/36-0798 is ringing
== Spawn extension (internal, 36, 2) exited non-zero on
'SIP/Company1-0797'
asterisk*CLI exit
And here is the sip.conf:
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net
host=sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes
[Company2]
...
[Company3]
...
[Company4]
...
And here is some of the extensions.conf file:
[incoming]
; Get the DID number from the TO header.
exten = s,1,Set(thedid=${SIP_HEADER(TO)})
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)
-Andrew
Re: [asterisk-users] call between snom 300 and aastra 6731i
-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/300-0192, RC=19) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/300-0192, 19,1) in new stack -- Goto (macro-dialout-trunk,19,1) -- Executing [19@macro-dialout-trunk:1] Goto(SIP/300-0192, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] NoOp(SIP/300-0192, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 19 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:2] Set(SIP/300-0192, CALLERID(number)=300) in new stack -- Executing [0176XX@from-internal:7] Macro(SIP/300-0192, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/300-0192, ) in new stack -- Executing [s@macro-outisbusy:2] GotoIf(SIP/300-0192, 0?emergency,1) in new stack -- Executing [s@macro-outisbusy:3] GotoIf(SIP/300-0192, 0?intracompany,1) in new stack -- Executing [s@macro-outisbusy:4] Playback(SIP/300-0192, all-circuits-busy-nowpls-try-call-later, noanswer) in new stack [2015-03-27 18:35:19] WARNING[350][C-00f3]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format [2015-03-27 18:35:19] WARNING[350][C-00f3]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory [2015-03-27 18:35:19] WARNING[350][C-00f3]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/300-0192 for all-circuits-busy-nowpls-try-call-later, noanswer [2015-03-27 18:35:19] WARNING[350][C-00f3]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format [2015-03-27 18:35:19] WARNING[350][C-00f3]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory [2015-03-27 18:35:19] WARNING[350][C-00f3]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/300-0192 for all-circuits-busy-nowpls-try-call-later, noanswer -- Executing [s@macro-outisbusy:5] Congestion(SIP/300-0192, 20) in new stack [2015-03-27 18:35:19] WARNING[350][C-00f3]: channel.c:4862 ast_prod: Prodding channel 'SIP/300-0192' failed == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/300-0192' in macro 'outisbusy' == Spawn extension (from-internal, 0176XX, 7) exited non-zero on 'SIP/300-0192' -- Executing [h@from-internal:1] Hangup(SIP/300-0192, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-0192' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/300-0192 [2015-03-27 18:35:28] WARNING[18275]: chan_sip.c:23527 handle_response_register: Got 423 Interval too brief for service fdmar...@sip.serveurcom.com, minimum is 480 seconds thanks nd regards 2015-03-27 17:08 GMT+00:00 Gareth Blades mailinglist+aster...@dns99.co.uk: You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com : hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ok inbound and outbound the calls between x-lite and snom300 ok inbound and outbound the issue just between snom and aastra i can call from aastra to snom without issue but when itry to call from snom300 to aastra6731i i get bad request all the time i test with 3 snom300 i get the same result please any body have the snom and aastra can help me in order to fixe this issue thanks and 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 -- _ -- 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 between snom 300 and aastra 6731i
please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com: hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ok inbound and outbound the calls between x-lite and snom300 ok inbound and outbound the issue just between snom and aastra i can call from aastra to snom without issue but when itry to call from snom300 to aastra6731i i get bad request all the time i test with 3 snom300 i get the same result please any body have the snom and aastra can help me in order to fixe this issue thanks and 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
[asterisk-users] call between snom 300 and aastra 6731i
hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ok inbound and outbound the calls between x-lite and snom300 ok inbound and outbound the issue just between snom and aastra i can call from aastra to snom without issue but when itry to call from snom300 to aastra6731i i get bad request all the time i test with 3 snom300 i get the same result please any body have the snom and aastra can help me in order to fixe this issue thanks and 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] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan mjor...@digium.com: On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without issue the problem just when i configure the trunk in my server and i use extension all the ip-phone and x-lite and server asterisk in the same network 192.168.1.x == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149XX -- SIP/FD-00b9 is making progress passing it to SIP/306-00b8 0x2afec424c430 -- Probation passed - setting RTP source address to 192.168.1.212:57592 0xc5922b0 -- Probation passed - setting RTP source address to 217.195.xx.xxx:29674 -- Got SIP response 556 No address found back from 217.195.XX.XXX:5060 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:23] NoOp(SIP/306-00b8, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34) in new stack -- Executing [s@macro-dialout-trunk:24] GotoIf(SIP/306-00b8, 0?continue,1:s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/306-00b8, RC=34) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/306-00b8, 34,1) in new stack -- Goto (macro-dialout-trunk,34,1) -- Executing [34@macro-dialout-trunk:1] Goto(SIP/306-00b8, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] NoOp(SIP/306-00b8, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:2] Set(SIP/306-00b8, CALLERID(number)=306) in new stack -- Executing [0149XX@from-internal:7] Macro(SIP/306-00b8, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/306-00b8, ) in new stack -- Executing [s@macro-outisbusy:2] GotoIf(SIP/306-00b8, 0?emergency,1) in new stack -- Executing [s@macro-outisbusy:3] GotoIf(SIP/306-00b8, 0?intracompany,1) in new stack -- Executing [s@macro-outisbusy:4] Playback(SIP/306-00b8, all-circuits-busy-nowpls-try-call-later, noanswer) in new stack [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/306-00b8 for all-circuits-busy-nowpls-try-call-later, noanswer [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/306-00b8 for all-circuits-busy-nowpls-try-call-later, noanswer -- Executing [s@macro-outisbusy:5] Congestion(SIP/306-00b8, 20) in new stack [2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862 ast_prod: Prodding channel 'SIP/306-00b8' failed == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/306-00b8' in macro 'outisbusy' == Spawn extension (from-internal, 0149XX, 7) exited non-zero on 'SIP/306-00b8' -- Executing [h@from-internal:1] Hangup(SIP/306-00b8, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-00b8' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/306-00b8 The verbose output states why your call is congested: -- Got SIP response 556 No address found back from 217.195.XX.XXX:5060 The far end came back with a 556 response to the outbound INVITE request. It doesn't think that whatever you dialled exists. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW
[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without issue the problem just when i configure the trunk in my server and i use extension all the ip-phone and x-lite and server asterisk in the same network 192.168.1.x == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149XX -- SIP/FD-00b9 is making progress passing it to SIP/306-00b8 0x2afec424c430 -- Probation passed - setting RTP source address to 192.168.1.212:57592 0xc5922b0 -- Probation passed - setting RTP source address to 217.195.xx.xxx:29674 -- Got SIP response 556 No address found back from 217.195.XX.XXX:5060 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:23] NoOp(SIP/306-00b8, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34) in new stack -- Executing [s@macro-dialout-trunk:24] GotoIf(SIP/306-00b8, 0?continue,1:s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/306-00b8, RC=34) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/306-00b8, 34,1) in new stack -- Goto (macro-dialout-trunk,34,1) -- Executing [34@macro-dialout-trunk:1] Goto(SIP/306-00b8, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] NoOp(SIP/306-00b8, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:2] Set(SIP/306-00b8, CALLERID(number)=306) in new stack -- Executing [0149XX@from-internal:7] Macro(SIP/306-00b8, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/306-00b8, ) in new stack -- Executing [s@macro-outisbusy:2] GotoIf(SIP/306-00b8, 0?emergency,1) in new stack -- Executing [s@macro-outisbusy:3] GotoIf(SIP/306-00b8, 0?intracompany,1) in new stack -- Executing [s@macro-outisbusy:4] Playback(SIP/306-00b8, all-circuits-busy-nowpls-try-call-later, noanswer) in new stack [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/306-00b8 for all-circuits-busy-nowpls-try-call-later, noanswer [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/306-00b8 for all-circuits-busy-nowpls-try-call-later, noanswer -- Executing [s@macro-outisbusy:5] Congestion(SIP/306-00b8, 20) in new stack [2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862 ast_prod: Prodding channel 'SIP/306-00b8' failed == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/306-00b8' in macro 'outisbusy' == Spawn extension (from-internal, 0149XX, 7) exited non-zero on 'SIP/306-00b8' -- Executing [h@from-internal:1] Hangup(SIP/306-00b8, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-00b8' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/306-00b8 -- _ -- 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] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
thank you for your response but i think that the issue is related to the
RTP because i can call all numbers with the same format
when i call any number except 0033149xx i get the same adress from
provider only with this number cnfigurerd in ip-phone in our network i get
this error
best regards
number works without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033661223291
-- SIP/FD-011f is making progress passing it to SIP/306-011e
0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:12728 ip adress of my x-lite
0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486ip adress of provider
SIP/FD-011f answered SIP/306-011e
0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486 the same ip adress and the same port
number with error
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
- Called SIP/FD/0033149xx
SIP/FD-011d is making progress passing it to SIP/306-011c
0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:47452ip adress of my x-lite
0xc7452e0 -- Probation passed - setting RTP source address to
217.195.31.146:23392ip adress of provider
Got SIP response 556 No address found back from 217.195.31.129:5060
not the same ip and port
2015-03-25 13:47 GMT+00:00 A J Stiles asterisk_l...@earthshod.co.uk:
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks and regards
Make sure you are sending the number in the correct format, when you Dial()
via your trunk. Some providers want you to omit the leading zero from the
STD
code. Others want you to include it. Others still want you to include the
IDD code (and then definitely leave out the 0, just like you were phoning
home
from abroad).
My home phone number is (01332) XX. To call it, you might have to
Dial()
any of the following (assuming OUTSIDE is defined elsewhere):
Dial(${OUTSIDE}/01332XX, 60); with leading 0
Dial(${OUTSIDE}/1332XX, 60) ; without leading 0
Dial(${OUTSIDE}/441332XX, 60) ; with IDD code
If you don't know what format your telco are expecting and have to
determine
by experiment, it probably would be easiest to set up an extension which
just
makes a call to one fixed number -- your own mobile is as good as anything
else.
To remove the leading 0 from ${EXTEN} , you can use ${EXTEN:1} which omits
one
digit from the beginning.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] outbound calls
hi the issue still the same i have 2 trunks whe i configure the first in x-lite and the second in my server or my ip-phone snom320 directly from x-lite i can call my trunk without issue but when i try ti call from snom320 to x-lite or from my server asterisk using extension in x-lite the call all time is failed any help please thanks and regards 2015-03-20 19:28 GMT+00:00 Trey Hilyard kct...@gmail.com: So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and 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 -- _ -- 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] outbound calls
thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and regards. 2015-03-20 18:39 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com: thank you i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and regards. 2015-03-20 17:15 GMT+00:00 Trey Hilyard kct...@gmail.com: I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: Got SIP response 556 No address found back from 217.195.xx.xxx:5060 Are you sure that 0033149xx is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but normally a 556 indicates that your provider didn't have routing for either the R-URI or they didn't recognize that is was coming from you. You might compare the SIP INVITE coming from Asterisk to the one from Z-Lite and see where the differences are. On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xx@from-internal:1] Macro(SIP/101-0103, user-callerid,LIMIT,EXTERNAL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/101-0103, TOUCH_MONITOR=1426869820.301) in new stack -- Executing [s@macro-user-callerid:2] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:3] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:4] ExecIf(SIP/101-0103, 1?Set(REALCALLERIDNUM=101)) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/101-0103, AMPUSERCIDNAME=101) in new stack -- Executing [s@macro-user-callerid:8] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:9] Set(SIP/101-0103, AMPUSERCID=101) in new stack -- Executing [s@macro-user-callerid:10] Set(SIP/101-0103, __DIAL_OPTIONS=tr) in new stack -- Executing [s@macro-user-callerid:11] Set(SIP/101-0103, CALLERID(all)=101 101) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:13] ExecIf(SIP/101-0103, 1?Set(GROUP(concurrency_limit)=101)) in new stack -- Executing [s@macro-user-callerid:14] ExecIf(SIP/101-0103, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:15] GotoIf(SIP/101-0103, 1?continue) in new stack -- Goto (macro-user-callerid,s,28) -- Executing [s@macro-user-callerid:28] Set(SIP/101-0103, CALLERID(number)=101) in new stack -- Executing [s@macro-user-callerid:29] Set(SIP/101-0103, CALLERID(name)=101) in new stack -- Executing [s@macro-user-callerid:30] Set(SIP/101-0103, CDR(cnum)=101) in new stack -- Executing [s@macro-user-callerid:31] Set(SIP/101-0103, CDR(cnam)=101) in new stack -- Executing [s@macro-user-callerid:32] Set(SIP/101-0103, CHANNEL(language)=en) in new stack -- Executing
Re: [asterisk-users] outbound calls
i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and 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
[asterisk-users] outbound calls
hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xx@from-internal:1] Macro(SIP/101-0103, user-callerid,LIMIT,EXTERNAL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/101-0103, TOUCH_MONITOR=1426869820.301) in new stack -- Executing [s@macro-user-callerid:2] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:3] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:4] ExecIf(SIP/101-0103, 1?Set(REALCALLERIDNUM=101)) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/101-0103, AMPUSERCIDNAME=101) in new stack -- Executing [s@macro-user-callerid:8] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:9] Set(SIP/101-0103, AMPUSERCID=101) in new stack -- Executing [s@macro-user-callerid:10] Set(SIP/101-0103, __DIAL_OPTIONS=tr) in new stack -- Executing [s@macro-user-callerid:11] Set(SIP/101-0103, CALLERID(all)=101 101) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:13] ExecIf(SIP/101-0103, 1?Set(GROUP(concurrency_limit)=101)) in new stack -- Executing [s@macro-user-callerid:14] ExecIf(SIP/101-0103, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:15] GotoIf(SIP/101-0103, 1?continue) in new stack -- Goto (macro-user-callerid,s,28) -- Executing [s@macro-user-callerid:28] Set(SIP/101-0103, CALLERID(number)=101) in new stack -- Executing [s@macro-user-callerid:29] Set(SIP/101-0103, CALLERID(name)=101) in new stack -- Executing [s@macro-user-callerid:30] Set(SIP/101-0103, CDR(cnum)=101) in new stack -- Executing [s@macro-user-callerid:31] Set(SIP/101-0103, CDR(cnam)=101) in new stack -- Executing [s@macro-user-callerid:32] Set(SIP/101-0103, CHANNEL(language)=en) in new stack -- Executing [0149xx@from-internal:2] Set(SIP/101-0103, MOHCLASS=default) in new stack -- Executing [0149xx@from-internal:3] Set(SIP/101-0103, _NODEST=) in new stack -- Executing [0149xx@from-internal:4] Gosub(SIP/101-0103, sub-record-check,s,1(out,0149xx,)) in new stack -- Executing [s@sub-record-check:1] Set(SIP/101-0103, REC_POLICY_MODE_SAVE=) in new stack -- Executing [s@sub-record-check:2] GotoIf(SIP/101-0103, 1?check) in new stack -- Goto (sub-record-check,s,7) -- Executing [s@sub-record-check:7] Set(SIP/101-0103, __MON_FMT=wav) in new stack -- Executing [s@sub-record-check:8] GotoIf(SIP/101-0103, 1?next) in new stack -- Goto (sub-record-check,s,11) -- Executing [s@sub-record-check:11] ExecIf(SIP/101-0103, 0?Return()) in new stack -- Executing [s@sub-record-check:12] ExecIf(SIP/101-0103, 0?Set(__REC_POLICY_MODE=)) in new stack -- Executing [s@sub-record-check:13] GotoIf(SIP/101-0103, 0?out,1) in new stack -- Executing [s@sub-record-check:14] Set(SIP/101-0103, __REC_STATUS=INITIALIZED) in new stack -- Executing [s@sub-record-check:15] Set(SIP/101-0103, NOW=1426869820) in new stack -- Executing [s@sub-record-check:16] Set(SIP/101-0103, __DAY=20) in new stack -- Executing [s@sub-record-check:17] Set(SIP/101-0103, __MONTH=03) in new stack -- Executing [s@sub-record-check:18] Set(SIP/101-0103, __YEAR=2015) in new stack -- Executing [s@sub-record-check:19] Set(SIP/101-0103, __TIMESTR=20150320-164340) in new stack -- Executing [s@sub-record-check:20] Set(SIP/101-0103, __FROMEXTEN=101) in new stack -- Executing [s@sub-record-check:21] Set(SIP/101-0103, __CALLFILENAME=out-0149xx-101-20150320-164340-1426869820.301) in new stack -- Executing [s@sub-record-check:22] Goto(SIP/101-0103, out,1) in new stack -- Goto (sub-record-check,out,1) -- Executing [out@sub-record-check:1] ExecIf(SIP/101-0103, 1?Set(__REC_POLICY_MODE=always)) in new stack -- Executing [out@sub-record-check:2] GosubIf(SIP/101-0103, 1?record,1(exten,0149xx,101)) in new stack -- Executing [record@sub-record-check:1] Set(SIP/101-0103, AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack -- Executing [record@sub-record-check:2] MixMonitor(SIP/101-0103, 2015/03/20/out-0149xx-101-20150320-164340-1426869820.301.wav,,) in new stack -- Executing [record@sub-record-check:3] Set(SIP/101-0103,
Re: [asterisk-users] chanspy for group extension
thank you so much Carlos ;the issue has been solved
Best Regards.
2015-03-12 18:40 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
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Re: [asterisk-users] chanspy for group extension
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
but when i do 007100 for exemple i spy another agnet 102 or 103
any help please
thanks and regards
2015-03-12 10:30 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez cur...@telecomabmex.com:
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use chanspy like below
100=199 with Authenticate(1234)
200=299 with Authenticate(5678)
300=399 with Authenticate(8910)
Use a macro and pass the pin as a parameter:
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
[app-chanspy]
exten = _0071XX,Macro(chanspy,1234)
exten = _0072XX,Macro(chanspy,5678)
exten = _0073XX,Macro(chanspy,8910)
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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Re: [asterisk-users] chanspy for group extension
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
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Re: [asterisk-users] chanspy for group extension
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez cur...@telecomabmex.com:
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use chanspy like below
100=199 with Authenticate(1234)
200=299 with Authenticate(5678)
300=399 with Authenticate(8910)
Use a macro and pass the pin as a parameter:
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
[app-chanspy]
exten = _0071XX,Macro(chanspy,1234)
exten = _0072XX,Macro(chanspy,5678)
exten = _0073XX,Macro(chanspy,8910)
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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[asterisk-users] chanspy for group extension
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use chanspy like below
100=199 with Authenticate(1234)
200=299 with Authenticate(5678)
300=399 with Authenticate(8910)
any help please
Thanks and regards
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[asterisk-users] set musiconhold only for caller
hello list, i have created a queue with and i have a question related to musiconhold f there is any way to set the musiconhold just for caller not for agent logged in the queue thanks and 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
[asterisk-users] issue with inbound route
hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud i use elastix Executing [s@from-trunk:1] NoOp(SIP/358-106-00c0, No DID or CID Match) in new stack -- Executing [s@from-trunk:2] Answer(SIP/358-106-00c0, ) in new stack -- Executing [s@from-trunk:3] Wait(SIP/358-106-00c0, 2) in new stack 0x2add5020a390 -- Probation passed - setting RTP source address to 217.xxx.xx.xxx:207xx -- Executing [s@from-trunk:4] Playback(SIP/358-106-00c0, ss-noservice) in new stack -- SIP/358-106-00c0 Playing 'ss-noservice.gsm' (language 'en') -- Executing [s@from-trunk:5] SayAlpha(SIP/358-106-00c0, ) in new stack -- Executing [s@from-trunk:6] Hangup(SIP/358-106-00c0, ) in new stack == Spawn extension (from-trunk, s, 6) exited non-zero on 'SIP/358-106-00c0' -- Executing [h@from-trunk:1] Macro(SIP/358-106-00c0, hangupcall,) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/358-106-00c0, 1?endmixmoncheck) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp(SIP/358-106-00c0, End of MIXMON check) in new stack -- Executing [s@macro-hangupcall:10] GotoIf(SIP/358-106-00c0, 1?nomeetmemon) in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp(SIP/358-106-00c0, End of MEETME check) in new stack -- Executing [s@macro-hangupcall:29] GotoIf(SIP/358-106-00c0, 1?noautomon) in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp(SIP/358-106-00c0, TOUCH_MONITOR_OUTPUT=) in new stack -- Executing [s@macro-hangupcall:35] GotoIf(SIP/358-106-00c0, 1?noautomon2) in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp(SIP/358-106-00c0, MONITOR_FILENAME=) in new stack -- Executing [s@macro-hangupcall:42] GotoIf(SIP/358-106-00c0, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf(SIP/358-106-00c0, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf(SIP/358-106-00c0, 1?theend) in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI(SIP/358-106-00c0, hangup.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- SIP/358-106-00c0AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup(SIP/358-106-00c0, ) in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/358-106-00c0' in macro 'hangupcall' == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/358-106-00c0' thanks and 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
[asterisk-users] asterisk and elastix
Hello list, i have installed elastix 2.4.0 with call center model and i have created an Outgoing Calls https://192.168.1.251/index.php?menu=outgoing_calls my question i want to know the name of the tbale where the csv file is uploaded in order to do some works. NB: i found the cdr table in asteriskcdrdb database but the is no information related to my csv file any help please thanks and 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
[asterisk-users] how to stop asterisk using a call
hello list, i have a question i don't know if there is any possibility to stop asterisk using a call for exp: when i call a number 0522xx i want to excute a script or any idea to stop asterisk automatically i use asterisk 1.4.43 NB: with mysql using a database i can insert into table using php without issue. but now with SSH how can i do thanks and 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] how to stop asterisk using a call
thanks a lot it works correctly 2014-04-07 12:08 GMT+00:00 Andres and...@telesip.net: On 4/7/14, 4:53 AM, Salaheddine Elharit wrote: hello list, i have a question i don't know if there is any possibility to stop asterisk using a call for exp: when i call a number 0522xx i want to excute a script or any idea to stop asterisk automatically Sure, try something like: [custom-stop] exten = 052212345,1,System(sudo /usr/sbin/service asterisk stop) (you need to give the asterisk owner permission to execute 'service' comand via sudo) i use asterisk 1.4.43 NB: with mysql using a database i can insert into table using php without issue. but now with SSH how can i do thanks and regards. -- Technical Supporthttp://www.cellroute.net -- _ -- 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] h extension isn't processed after call file finishes.
hello, try to use failed instead of h exten = failed,1, best regards. 2014-02-18 9:09 GMT+00:00 Ishfaq Malik i...@pack-net.co.uk: What version of asterisk are you using? Ish On 17 February 2014 20:49, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension when it's finished. The second leg is being initiated by a .call file like: Channel: local/1505xxx@context Application: sendfax Data: /tmp/voice11-voice11-1392668806.182025.tiff,zfds WaitTime: 90 MaxRetries: 2 Account: vFax CallerID: Fax 505xxx The h extension calls an agi scrip that logs a bunch of information about the fax attempt. Works just fine when I receive a fax. But there is no sign of it in the logs for the sending leg of the fax. Is there something I need to do in order to get the h extension to get called? Mike. -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] auto-answer call
hi when i try to this with page() exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) exten = 506,n,page(SIP/105) CLIAccepting call from '0661xx' to '506' on channel 1/13, span 1 -- Executing [506@default:1] SIPAddHeader(DAHDI/13-1, Call-Info:__; answer-after=0) in new stack -- Executing [506@default:2] Page(DAHDI/13-1, SIP/105) in new stack -- Called 105 -- DAHDI/13-1 Playing 'beep' (language 'en') -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- Created MeetMe conference 1023 for conference '1894843837d' -- SIP/105-00c7 is ringing -- Span 1: Channel 1/13 got hangup, cause -1 -- Hungup 'DAHDI/pseudo-358137724' == Spawn extension (default, 506, 2) exited non-zero on 'DAHDI/13-1' -- Hungup 'DAHDI/13-1' and the call hungup when i use the Dial the sip/105 still ringing thanks and regards 2014-02-05 Larry Moore lmo...@omninet.net.au: On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, Here is a list of headers used for various vendors, I can't remember which one is for Polycom. SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Alert-Info: Info=Alert-Autoanswer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); SIPAddHeader(Answer-Mode: Auto); Larry. -- _ -- 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] auto-answer call
thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- 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
[asterisk-users] auto-answer call
hello list, i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) exten = 506,n,Dial(SIP/105) when i call the 506 the SIP/105 still ringing, i have snom 320 and i have set the Auto Answer Indication: on i test with Dial and page() but the issue still the same any help please -- _ -- 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] callfiles.call
hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06 MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten = s,1,Ringing() exten = s,n,Playback(hello-world) exten = s,n,Dial(SIP/105) exten = s,n,Hangup() it works with one number how can i do in order to create a callfiles with a lot of numbers i try to create a callfiles.call like that Channel: SIP/watara/0661xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 Channel: SIP/watara/0669xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 but he call only the last number, any help please thanks and 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
[asterisk-users] go to context from server 1 to server 2
hello list i have create i trunk Sip between 2 servers in the same network when i call a number (inbound calls) in the first server i can forward this number to my sip 222 in the second server exten = 0522xx,1,Dial(SIP/222@trunk_created,30) my question if there is any possibility to GOTO a context in the second server after like below exten = 0522xx,1,Dial(SIP/222@ trunk_created,30) same = 0522xx,n,GoTo (context in the second server) thanks and 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] send the calls from to servers
hello thanks for your response i try to switch the provider in the same server without issue but my problem now i have 2 servers in the same network and with the same configuration iw want to use the group 2 of the server 1 and group 2 of server 2 for the same calls. and if group 2 of server 1 is down i can continue to use group 2 of server 2 thanks and regards [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 immediate=yes echocancel=no dtmfmode=auto group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 ;pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=5 immediate=no channel = 32-46,48-52 2013/12/19 Eric Wieling ewiel...@nyigc.com The basic idea is dial using your main outbound dahdi group, then check the value of HANGUPCAUSE, then if appropriate dial out using your secondary dahdi group. This is a standard thing. Check the mailing list archives and voip-info.org See also the [stdexten] section of extensions.conf.sample -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Thursday, December 19, 2013 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send the calls from to servers i ask about outbound calls not inbound round-robin best regards 2013/12/19 Eric Wieling ewiel...@nyigc.com Inbound call hunting is handled by your carrier, not Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Thursday, December 19, 2013 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] send the calls from to servers I have this scenario In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one diguim card with 2 ports: in the first port connection for the provider X : the second port of diguim card the connection of the provider Y In the second server (the same configuration) 192.168.5.200 asterisk installed 1.4.43 and one diguim card with 2 ports : the first port is empty the second port the connection of the provider Y My question how can I do in order to send the calls of the second providers from the port 2 server 1 and port 2 server 2 ()if one of them is down I continue to send the calls from the other Thanks and 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 -- _ -- 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] send the calls from to servers
i attached file my dialplan 2013/12/20 Salaheddine Elharit salah.elharit...@gmail.com in attached file my dialplan thanks and regards 2013/12/20 Eric Wieling ewiel...@nyigc.com You must write dialplan code to do what you want. Assuming you are not using a GUI with Asterisk, post your dialplan used for outgoing calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Friday, December 20, 2013 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send the calls from to servers hello thanks for your response i try to switch the provider in the same server without issue but my problem now i have 2 servers in the same network and with the same configuration iw want to use the group 2 of the server 1 and group 2 of server 2 for the same calls. and if group 2 of server 1 is down i can continue to use group 2 of server 2 thanks and regards [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 immediate=yes echocancel=no dtmfmode=auto group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 ;pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=5 immediate=no channel = 32-46,48-52 2013/12/19 Eric Wieling ewiel...@nyigc.com The basic idea is dial using your main outbound dahdi group, then check the value of HANGUPCAUSE, then if appropriate dial out using your secondary dahdi group. This is a standard thing. Check the mailing list archives and voip-info.org See also the [stdexten] section of extensions.conf.sample -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Thursday, December 19, 2013 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send the calls from to servers i ask about outbound calls not inbound round-robin best regards 2013/12/19 Eric Wieling ewiel...@nyigc.com Inbound call hunting is handled by your carrier, not Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Thursday, December 19, 2013 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] send the calls from to servers I have this scenario In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one diguim card with 2 ports: in the first port connection for the provider X : the second port of diguim card the connection of the provider Y In the second server (the same configuration) 192.168.5.200 asterisk installed 1.4.43 and one diguim card with 2 ports : the first port is empty the second port the connection of the provider Y My question how can I do in order to send the calls of the second providers from the port 2 server 1 and port 2 server 2 ()if one of them is down I continue to send the calls from the other Thanks and 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 -- _ -- 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
[asterisk-users] send the calls from to servers
I have this scenario In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one diguim card with 2 ports: in the first port connection for the provider X : the second port of diguim card the connection of the provider Y In the second server (the same configuration) 192.168.5.200 asterisk installed 1.4.43 and one diguim card with 2 ports : the first port is empty the second port the connection of the provider Y My question how can I do in order to send the calls of the second providers from the port 2 server 1 and port 2 server 2 ()if one of them is down I continue to send the calls from the other Thanks and 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] send the calls from to servers
i ask about outbound calls not inbound round-robin best regards 2013/12/19 Eric Wieling ewiel...@nyigc.com Inbound call hunting is handled by your carrier, not Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Thursday, December 19, 2013 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] send the calls from to servers I have this scenario In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one diguim card with 2 ports: in the first port connection for the provider X : the second port of diguim card the connection of the provider Y In the second server (the same configuration) 192.168.5.200 asterisk installed 1.4.43 and one diguim card with 2 ports : the first port is empty the second port the connection of the provider Y My question how can I do in order to send the calls of the second providers from the port 2 server 1 and port 2 server 2 ()if one of them is down I continue to send the calls from the other Thanks and 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 -- _ -- 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] issue with speech in IVR
hello johan,
i use Authenticate and i get what i want thank you so much for your help :)
exten = 600,1,Ringing(2)
exten = 600,n,Answer
exten = 600,n,Authenticate(1234)
exten = 600,n,Goto(home,s,1)
2013/12/5 Steve Edwards asterisk@sedwards.com
On Thu, 5 Dec 2013, Salaheddine Elharit wrote:
i have one question related to the IVR below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
how can i ask the customer to enter a password before to go to (home,s,1)
and where i must to store a password for example password 1234
if the customer enter 1234 he can Goto(home,s,1) and if the password is
wrong i playback an error message
That's 3 questions :)
You need to provide more details.
Is the password fixed or stored in a database? Is it the same as their
voicemail password?
There are examples for all these scenarios. Goggle about, read ATFOT,
visit voip-info.org or use the Asterisk 'help' commands.
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}error
Why are you fiddling with global variables? Isn't
'/var/lib/asterisk/sounds/' your 'default' sounds path?
Please don't top post.
Please trim irrelevent cruft from previous posts.
Please don't burn all your karma points asking simple questions.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] issue with speech in IVR
hello list
i have one question related to the IVR below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
how can i ask the customer to enter a password before to go to (home,s,1)
and where i must to store a password for example password 1234
if the customer enter 1234 he can Goto(home,s,1) and if the password is
wrong i playback an error message
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
the customer must enter 1234 if yes go to (home,s,1) if no go to error
exten = 600,n,Goto(home,s,1)
[error]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}error
any example would be appreciated
2013/11/29 Mitul Limbani mi...@enterux.in
Sounds cool, I suspected the echo cancel situation, these are usually
issue even for FAX communication on dahdi.
Mitul
On Friday, November 29, 2013, Salaheddine Elharit wrote:
hello
i add the following in chan_dahdi and the issue has been solved thanks a
lot for your help and support now ican stop the speech and go to my context
i really appreciate your help and support
immediate = yes
echocancel = no
dtmfmode = auto
-- Forwarded message --
From: Salaheddine Elharit salah.elharit...@gmail.com
Date: 2013/11/29
Subject: Re: [asterisk-users] issue with speech in IVR
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
hello
i add the following in chan_dahdi and the issue has been solved thanks
a lot for your help and support now ican stop the speech and go to my
context
i really appreciate your help and support
2013/11/29 Mitul Limbani mi...@enterux.in
Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode = auto
Mitul
On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote:
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use
the keypad to control the playback
-Original Message-
From: Salaheddine Elharit salah.elharit...@gmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
08:05:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] issue with speech in IVR
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Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] issue with speech in IVR
hi yes if imake an extension-to-extension call, i can send DTMF, Both ways yes in my case i don't need a Hardware SIP phone or a software SIP phones i have just a number 05xx600 when the customer call this number i stor his number in my database and i call him later if he press 1 for xx 1 press 2 for yyy i sotre his phone number and his choice in my database for me the issue the customer he can nto wait the speech of unless and finished . best regards i use a diguim card with PRI 2013/11/29 A J Stiles asterisk_l...@earthshod.co.uk On 28/11/13 15:36, Salaheddine Elharit wrote: hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context any other idea please best regards . It sounds as thgough the DTMF tones are not being sent in a way that Asterisk is seeing . What type of telephone technology are you using? Hardware SIP phones, software SIP phones, analogue phones via an FXS card, analogue phones via a SIP ATA? What codec are you using? If you make an extension-to-extension call, can you send DTMF tones down the line? Both ways around? Do they decode properly? (You can get a mobile phone app for this.) -- AJS Answers come *after* questions. -- _ -- 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] issue with speech in IVR
hello i add the following in chan_dahdi and the issue has been solved thanks a lot for your help and support now ican stop the speech and go to my context i really appreciate your help and support 2013/11/29 Mitul Limbani mi...@enterux.in Try following in chan_dahdi immediate = yes echocancel = no dtmfmode = auto Mitul On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote: Are you using a mp3 file? I have noticed that using control playback with a mp3 file I cannot use the keypad to control the playback -Original Message- From: Salaheddine Elharit salah.elharit...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013 08:05:16 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] issue with speech in IVR -- _ -- 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 -- _ -- 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] Fwd: issue with speech in IVR
hello i add the following in chan_dahdi and the issue has been solved thanks a lot for your help and support now ican stop the speech and go to my context i really appreciate your help and support immediate = yes echocancel = no dtmfmode = auto -- Forwarded message -- From: Salaheddine Elharit salah.elharit...@gmail.com Date: 2013/11/29 Subject: Re: [asterisk-users] issue with speech in IVR To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com hello i add the following in chan_dahdi and the issue has been solved thanks a lot for your help and support now ican stop the speech and go to my context i really appreciate your help and support 2013/11/29 Mitul Limbani mi...@enterux.in Try following in chan_dahdi immediate = yes echocancel = no dtmfmode = auto Mitul On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote: Are you using a mp3 file? I have noticed that using control playback with a mp3 file I cannot use the keypad to control the playback -Original Message- From: Salaheddine Elharit salah.elharit...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013 08:05:16 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] issue with speech in IVR -- _ -- 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 -- _ -- 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] issue with speech in IVR
hello,
i have add the the code below but the issue still the same i can't go to
the project during the speech
any other solution
best regards
NB:for the version of asterisk i can't move to another version for the
moment
exten = _X,1,NoOp(Digit entered during prompt)
exten = _X,2,Goto(project,s,1)
2013/11/28 Paul Belanger paul.belan...@polybeacon.com
On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
hello list
i have an IVR menu in asterisk 1.4
like below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}music1)
exten = s,n,Background(${sounds_path}music2)
exten = s,n,Background(${sounds_path}music3)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,WaitExten(5)
exten = i,n,goto(home,s,1)
exten = 1,1,Goto(project,s,1)
exten = _X,1,NoOp(Digit entered during prompt)
exten = _X,2,Goto(project,s,1)
[project]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}mymusic)
exten = s,n,WaitExten(5)
exten = s,n,Goto(project,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to
go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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Re: [asterisk-users] issue with speech in IVR
hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context any other idea please best regards . 2013/11/28 A J Stiles asterisk_l...@earthshod.co.uk On Wednesday 27 November 2013, Salaheddine Elharit wrote: hello list i have an IVR menu in asterisk 1.4 [stuff deleted] my problem when the customor call the number 600 and press 1 in order to go to the project menu he must wait all the speech music1 music2 and music 3 if there is any way to go to project menu during the speech thanks and regards This is an actual dialplan application that I wrote. It's a spike -- a proof of concept that is all depth and no breadth. It's known to work in Asterisk 1.8. The sound files ajs_juke01 and ajs_anykey you will need to create for yourself, depending what MP3s you have available (and replace ajs_ with your own prefix). You can interrupt the announcements or the MP3s by pressing keys while playing. ;;; VERY PRIMITIVE JUKE BOX CONTEXT ;;; [vpjb] exten = s,1,Background(ajs_juke01) ; Press 1 for Ocean Colour Scene, 2 for Crowded House exten = s,n,WaitExten(1) exten = s,n,Goto(1) exten = i,1,Hangup() exten = 1,1,Background(ajs_anykey) ; Press any key to stop the music and return to the menu exten = 1,n,MP3Player(/songs/09_policemen+pirates.mp3) exten = 1,n,Goto(vpjb,s,1) exten = 2,1,Background(ajs_anykey) ; Press any key to stop the music and return to the menu exten = 2,n,MP3Player(/songs/15_distant_sun.mp3) exten = 2,n,Goto(vpjb,s,1) exten = _X,1,Hangup() -- AJS Answers come *after* questions. -- _ -- 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] issue with speech in IVR
thanks steve for your response i use dahdi. and in my sip.conf i have dtmfmode=auto idon't know if i must to put relaxdtmf=yes ? in sip.conf or i need to it in another files FYI i have a diguim card with dahdi and asterisk 1.4 thanks and regards 2013/11/28 Steve Murphy m...@parsetree.com On Thu, Nov 28, 2013 at 8:36 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context any other idea please best regards . My guess is that your DTMF tones are not reaching Asterisk. Seen it many times. Study the path whereby the DTMF is generated and recognized and processed by Asterisk. What kind of device are you using? Dahdi? SIP? You can use the rtp set debug to see if the DTMF is coming thru; look at your channel config, there may be something there that might prevent DTMF. Same with the phone settings. Best of Luck, murf -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ murf at parsetree dott com ☎ 307-899-5535 -- _ -- 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] issue with speech in IVR
hello list
i have an IVR menu in asterisk 1.4
like below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}music1)
exten = s,n,Background(${sounds_path}music2)
exten = s,n,Background(${sounds_path}music3)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,WaitExten(5)
exten = i,n,goto(home,s,1)
exten = 1,1,Goto(project,s,1)
[project]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}mymusic)
exten = s,n,WaitExten(5)
exten = s,n,Goto(project,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
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[asterisk-users] issue with dahdi_channels.conf
Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 17-31 context = default group = 63 but when i add in channel 1-15 like: channel = 1-15,17-31 i receive all the time this message chan_dahdi.c:9438 pri_fixup_principle: Can't move call (DAHDI/3-1) from channel 3 to 2. It is already in use. WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Span 1: PRI requested channel 1/2 is not available. in span 2 there is no problem ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 group=0,12 context=from-pstn switchtype = qsig signalling = pri_net channel = 32-46,48-62 context = default group = 63 could you please help me thanks and 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] issue with dahdi_channels.conf
below etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Tue Oct 22 15:03:14 2013 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) span=1,0,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Global data loadzone = us defaultzone = us dahdi-channels.conf === with this configuration there is no problem but when i add 1-15 and i make service asterisk stop, service dahdi stop, service dahdi start, service asterisk start i can't make the calls i must remove 1-15 in order to make the calls ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Oct 22 15:03:14 2013 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 17-31 context = default group = 63 ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 group=0,12 context=from-pstn switchtype = qsig signalling = pri_net channel = 32-46,48-62 context = default group = 63 chan_dahdi.conf === [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 ;pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=52xx immediate=no channel = 32-46,48-52 thanks and regards 2013/10/31 A J Stiles asterisk_l...@earthshod.co.uk On Thursday 31 October 2013, Salaheddine Elharit wrote: Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 17-31 context = default group = 63 but when i add in channel 1-15 like: channel = 1-15,17-31 i receive all the time this message chan_dahdi.c:9438 pri_fixup_principle: Can't move call (DAHDI/3-1) from channel 3 to 2. It is already in use. WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Span 1: PRI requested channel 1/2 is not available. in span 2 there is no problem ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 group=0,12 context=from-pstn switchtype = qsig signalling = pri_net channel = 32-46,48-62 context = default group = 63 could you please help me Not without more information. Can you post the contents of /etc/dahdi/system.conf ? What country are you in? Are the jumpers on your card set correctly for there? Do your telco have any information regarding configuring Asterisk to work with their equipment? (They should have at least heard of Asterisk by now.) -- AJS Answers come *after* questions. -- _ -- 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] issue with dahdi_channels.conf
thanks for your response i will swap the cables and i will update by the result best regards 2013/10/31 Tony Mountifield t...@softins.co.uk In article cahexamsp4nenuntymuzwjgep69v+7rb7ekbyzsalmbm+zyo...@mail.gmail.com, Salaheddine Elharit salah.elharit...@gmail.com wrote: below etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Tue Oct 22 15:03:14 2013 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) span=1,0,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Global data loadzone = us defaultzone = us OK, that looks fine. dahdi-channels.conf === with this configuration there is no problem but when i add 1-15 and i make service asterisk stop, service dahdi stop, service dahdi start, service asterisk start i can't make the calls i must remove 1-15 in order to make the calls It's always possible that the problem is a misconfiguration at the remote end. I had that once, where the PBX to which Asterisk was talking had had its channel numbers misconfigured, resulting in a similar problem to what you have described. What happens if you swap the cables over between the two E1 ports on the card? Does the problem move to the second card (channels 32-46)? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- _ -- 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] issue after install dahdi
Hello i check the dahdi-channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 17-31 context = default group = 63 but when i add in channel 1-15 like: channel = 1-15,17-31 i receive all the time this message chan_dahdi.c:9438 pri_fixup_principle: Can't move call (DAHDI/3-1) from channel 3 to 2. It is already in use. WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Span 1: PRI requested channel 1/2 is not available. could you please help me thanks and regards 2013/10/24 Salaheddine Elharit salah.elharit...@gmail.com ok thanks for your comment i really appreciate it best regards 2013/10/23 Russ Meyerriecks rmeyerrie...@digium.com On Wed, Oct 23, 2013 at 11:27 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hi the issue has been solved after change the span from span =1,1,0,ccs,hdb3 to span =1,0,0,ccs,hdb3 thanks for everyone Salaheddine, Just a comment here: I'm not sure who your spans are connected to but, it is highly unlikely that this changed is what fixed your problem. I think it's more likely that the process of reloading something else actually fixed it. What you are doing here is telling span 1 to provide (or ignore) timing to the other end. If it's the case that you're connected to a public e1 pri provider, this probably isn't the correct configuration and will likely cause further problems like slips and alarms. If it's connected to something internal to your business, (like a channel bank), then it's fine. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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] issue after install dahdi
ok thanks for your comment i really appreciate it best regards 2013/10/23 Russ Meyerriecks rmeyerrie...@digium.com On Wed, Oct 23, 2013 at 11:27 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hi the issue has been solved after change the span from span =1,1,0,ccs,hdb3 to span =1,0,0,ccs,hdb3 thanks for everyone Salaheddine, Just a comment here: I'm not sure who your spans are connected to but, it is highly unlikely that this changed is what fixed your problem. I think it's more likely that the process of reloading something else actually fixed it. What you are doing here is telling span 1 to provide (or ignore) timing to the other end. If it's the case that you're connected to a public e1 pri provider, this probably isn't the correct configuration and will likely cause further problems like slips and alarms. If it's connected to something internal to your business, (like a channel bank), then it's fine. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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] issue after install dahdi
hi the issue has been solved after change the span from span =1,1,0,ccs,hdb3 to span =1,0,0,ccs,hdb3 thanks for everyone 2013/10/22 Salaheddine Elharit salah.elharit...@gmail.com 2013/10/22, A J Stiles asterisk_l...@earthshod.co.uk: On Tuesday 22 October 2013, Salaheddine Elharit wrote: hello yes this is a fresh install [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 the issue h=just with group 1 can not call via G1 with group 2 theris no problem group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=520xx immediate=no channel = 32-46,48-52 thanks and regards If group 2 works the way you want it to, then it must be configured correctly; meaning you just need to configure group 1 to match group 2. So, *make a backup copy* of your chan_dahdi.conf first, in case this goes horribly wrong and you can't even remember where you started from, and try: group=1 ;switchtype=euroisdn switchtype=qsig ;signalling=pri_cpe signalling=pri_net callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 Then power the server off and on, to make sure DAHDI and Asterisk restart from scratch. If that works, congratulations, you've fixed it. However, I don't think it will. switchtype=euroisdn and signalling=pri_cpe are the correct settings for plugging into an ISDN-30 exchange line. pri_net makes the card behave as though it was the exchange end (like FXS on steroids). Can you post the contents of /etc/dahdi/system.conf ? -- AJS Answers come *after* questions. -- _ -- 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 hello thanks for your response i have try to do this but the issue still the same NB: the group 1 is for the first provider and the secend is for the secend provider if the issue still the same can i call my provider becouse for the inbound call is ok bat the issue is for the outban calls below etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 17 12:37:31 2013 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Global data loadzone= fr defaultzone = fr thanks and 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] issue after install dahdi
hello yes this is a fresh install [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 the issue h=just with group 1 can not call via G1 with group 2 theris no problem group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=520xx immediate=no channel = 32-46,48-52 thanks and regards 2013/10/21 John Novack jnov...@stromberg-carlson.org A VERY OLD and beyond EOF version. If you MUST, due to some driver issue, use Asterisk 1.4, then please use 1.4.44 Otherwise I suggest you move to something more current, either version 1.8.current or beyond. Also, CLI says 1.4.43, your message says 1.4.32 ??? Some examination of chan_dahdi and your dialplan would help someone give you some assistance. Is this a fresh install, or one that has been working for years? What Digium card? John Novack Salaheddine Elharit wrote: i need your help regarding some issue related to the outband calls i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports when i try to call my phone number all time i receive message busy number this error just with g1. with g2 there is no problem i can call without issue can anyone see the CLI and tell me what is the problem thanks and regards == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12 currently running on SRVRADI O (pid = 4147) Verbosity is at least 3 -- Executing [0661049303@agents:1] Set(SIP/223-0021, CALLERID(number) =520460587) in new stack -- Executing [0661049303@agents:2] Dial(SIP/223-0021, DAHDI/g1/066104 9303|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/0661049303 -- Moving call (DAHDI/3-1) from channel 3 to 2. [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438 pri_fixup_principle: Can't mo ve call (DAHDI/3-1) from channel 3 to 2. It is already in use. [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Spa n 1: PRI requested channel 1/2 is not available. -- Hungup 'DAHDI/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [0661049303@agents:3] Hangup(SIP/223-0021, ) in new sta ck == Spawn extension (agents, 0661049303, 3) exited non-zero on 'SIP/223-002 1' -- Executing [h@agents:1] GotoIf(SIP/223-0021, 0?3:2) in new stack -- Goto (agents,h,2) -- Executing [h@agents:2] AHEventsProxy(SIP/223-0021, MSG_TYPE_TERMIN ATE_CALL1382377407) in new stack AHEventsProxy: Channel [SIP/223-0021]. Data [MSG_TYPE_TERMINATE_CALL138 2377407] -- chan is SIP/223-0021 AHEventsProxy: Send To CtiServer: socket:[89]. message:[41,1382377407stcrpb x^~] -- Executing [h@agents:3] Hangup(SIP/223-0021, ) in new stack == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-0021' -- SIP/224-0020 is ringing SRVRADIO*CLI Disconnected from Asterisk server Executing last minute cleanups -- Dog is my Co-pilot -- _ -- 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] issue after install dahdi
2013/10/22, A J Stiles asterisk_l...@earthshod.co.uk: On Tuesday 22 October 2013, Salaheddine Elharit wrote: hello yes this is a fresh install [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 the issue h=just with group 1 can not call via G1 with group 2 theris no problem group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=520xx immediate=no channel = 32-46,48-52 thanks and regards If group 2 works the way you want it to, then it must be configured correctly; meaning you just need to configure group 1 to match group 2. So, *make a backup copy* of your chan_dahdi.conf first, in case this goes horribly wrong and you can't even remember where you started from, and try: group=1 ;switchtype=euroisdn switchtype=qsig ;signalling=pri_cpe signalling=pri_net callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 Then power the server off and on, to make sure DAHDI and Asterisk restart from scratch. If that works, congratulations, you've fixed it. However, I don't think it will. switchtype=euroisdn and signalling=pri_cpe are the correct settings for plugging into an ISDN-30 exchange line. pri_net makes the card behave as though it was the exchange end (like FXS on steroids). Can you post the contents of /etc/dahdi/system.conf ? -- AJS Answers come *after* questions. -- _ -- 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 hello thanks for your response i have try to do this but the issue still the same NB: the group 1 is for the first provider and the secend is for the secend provider if the issue still the same can i call my provider becouse for the inbound call is ok bat the issue is for the outban calls below etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 17 12:37:31 2013 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Global data loadzone= fr defaultzone = fr thanks and 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
[asterisk-users] issue after install dahdi
i need your help regarding some issue related to the outband calls i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports when i try to call my phone number all time i receive message busy number this error just with g1. with g2 there is no problem i can call without issue can anyone see the CLI and tell me what is the problem thanks and regards == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12 currently running on SRVRADI O (pid = 4147) Verbosity is at least 3 -- Executing [0661049303@agents:1] Set(SIP/223-0021, CALLERID(number) =520460587) in new stack -- Executing [0661049303@agents:2] Dial(SIP/223-0021, DAHDI/g1/066104 9303|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/0661049303 -- Moving call (DAHDI/3-1) from channel 3 to 2. [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438 pri_fixup_principle: Can't mo ve call (DAHDI/3-1) from channel 3 to 2. It is already in use. [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Spa n 1: PRI requested channel 1/2 is not available. -- Hungup 'DAHDI/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [0661049303@agents:3] Hangup(SIP/223-0021, ) in new sta ck == Spawn extension (agents, 0661049303, 3) exited non-zero on 'SIP/223-002 1' -- Executing [h@agents:1] GotoIf(SIP/223-0021, 0?3:2) in new stack -- Goto (agents,h,2) -- Executing [h@agents:2] AHEventsProxy(SIP/223-0021, MSG_TYPE_TERMIN ATE_CALL1382377407) in new stack AHEventsProxy: Channel [SIP/223-0021]. Data [MSG_TYPE_TERMINATE_CALL138 2377407] -- chan is SIP/223-0021 AHEventsProxy: Send To CtiServer: socket:[89]. message:[41,1382377407stcrpb x^~] -- Executing [h@agents:3] Hangup(SIP/223-0021, ) in new stack == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-0021' -- SIP/224-0020 is ringing SRVRADIO*CLI Disconnected from Asterisk server Executing last minute cleanups -- _ -- 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] (no subject)
thanks for your response
with the code below i can't get the extenssions 223
exten = 529,1,Answer()
exten =
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
i can get my number only with uniqueid
test_num-0661xx_name-_529_UID-1376564701.1204.wav
any help please
thanks and regards
2013/8/13 Positively Optimistic positivelyoptimis...@gmail.com
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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[asterisk-users] (no subject)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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Re: [asterisk-users] asterisk and IVR
i have Create a h extension and all works without issue .thank you so
much for your help and support i really appreciate it.
2013/7/31 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
So it looks as though it's breaking out of the extension logic altogether,
if
the call gets answered. In that case, you'll have to do it the
old-fashioned
way: Create a h extension (which fires when a call is hung up) *in the
same context as your 534 extension* (you can have a h extension in each
context, if needs be), and do all your fancy end-of-call stuff there.
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,Goto(home,s,1)
exten = h,1,NoOp(Hangup received. Dial status is ${DIALSTATUS})
Note that if there are other extensions in the context, h will be called
when
they get hung up -- you might need some logic in there to deal with this
(or
cheat by just having one extension besides h in this context, and use a
fully-
specified Goto() to jump into it.)
--
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Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1'
-- Hungup 'Zap/26-1'
srvradio*CLI
the CLI for whe the call is no answer :
Accepting call from '0661xx' to '534' on channel 0/23, span 1
-- Executing [534@default:1] Dial(Zap/23-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e8b4b0 is ringing
-- Nobody picked up in 1 ms
-- Executing [534@default:2] NoOp(Zap/23-1, Dial status is
NOANSWER) in new stack
-- Executing [534@default:3] GotoIf(Zap/23-1, 0?answered) in new
stack
-- Executing [534@default:4] Goto(Zap/23-1, home|s|1) in new stack
-- Goto (home,s,1)
-- Executing [s@home:1] SetGlobalVar(Zap/23-1,
sounds_path=/var/lib/asterisk/sounds/) in new stack
== Setting global variable 'sounds_path' to '/var/lib/asterisk/sounds/'
-- Executing [s@home:2] BackGround(Zap/23-1,
/var/lib/asterisk/sounds/welcome) in new stack
-- Zap/23-1 Playing '/var/lib/asterisk/sounds/welcome' (language 'en')
-- Channel 0/23, span 1 got hangup request, cause 16
== Spawn extension (home, s, 2) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
2013/7/26 A J Stiles asterisk_l...@earthshod.co.uk
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with
the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
any help please
Do you get the dial status displayed? Then the NoOp() immediately before
the
GotoIf is executing. It's just possible I messed up the syntax of the
GotoIf() since I can't actually test that right now -- I do have an
Asterisk
box with a dialplan stuffed with GotoIf() statements; but right at the
moment,
I can't get to that machine.
Please paste your CLI output below, for the cases where (1) the call is
answered and (2) the Dial() command times out.
--
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Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
2013/7/31 Joshua Colp jc...@digium.com
A J Stiles wrote:
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1'
-- Hungup 'Zap/26-1'
srvradio*CLI
As dialplan execution stops if the outgoing call is answered and then
bridged the approach of using a Goto afterwards for ANSWER as well will not
work. You *must* use the h extension that was previously mentioned to cover
this case.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] asterisk and IVR
hi
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call from '06' to '534' on channel 0/12, span 1
-- Executing [534@default:1] Dial(Zap/12-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08361358 is ringing
-- Got SIP response 480 Temporarily Unavailable back from
192.168.5.131
-- SIP/228-08361358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/12-1' status is 'CONGESTION'
2) for no answer i get status is 'NOANSWER'
Accepting call from '06' to '534' on channel 0/4, span 1
-- Executing [534@default:1] Dial(Zap/4-1, SIP/228| 10) in new stack
-- Called 228
-- SIP/228-08362880 is ringing
-- Nobody picked up in 1 ms
== Auto fallthrough, channel 'Zap/4-1' status is 'NOANSWER'
3) for answered i don't get the status is 'answered'
Accepting call from '06' to '534' on channel 0/15, span 1
-- Executing [534@default:1] Dial(Zap/15-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08363bb8 is ringing
-- SIP/228-08363bb8 answered Zap/15-1
when i have the result is 'CONGESTION' or 'NOANSWER'i can go to the next
(home,s,1)
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = CONGESTION])
exten = 534,n,Goto(home,s,1)
how to do in order to go to the next if the result is answered
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
thanks and regards
2013/7/25 Salaheddine Elharit salah.elharit...@gmail.com
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like
below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
Immediately after the Dial() statement, add a line like
exten = s,nNoOp(Dial status is ${DIALSTATUS})
That will show you the actual contents of ${DIALSTATUS} in the CLI (in
case
it is not what you are expecting). Call your extension a few times, and
see
exactly what you get when the line is answered, unanswered, engaged and
maybe
if the phone is unplugged.
Instead of having a separate extension named after every possible value of
${DIALSTATUS} it might be easier to use a GotoIf() statement to jump away
in
one case (most sensibly, if the call was answered), and fall through to
the
default otherwise (engaged and phone not connected are similar
enough to
no answer for that probably to be what you want, barring special values
--
feel free to use more GotoIf() statements if required).
Something like:
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = s,n,NoOp(execution continues here if no answer)
...
exten = s,n,Hangup()
exten = s,n(answered),NoOp(we jump here if call was answered)
...
exten = s,n,Hangup()
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Re: [asterisk-users] asterisk and IVR
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
any help please
2013/7/26 A J Stiles asterisk_l...@earthshod.co.uk
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call from '06' to '534' on channel 0/12, span 1
-- Executing [534@default:1] Dial(Zap/12-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08361358 is ringing
-- Got SIP response 480 Temporarily Unavailable back from
192.168.5.131
-- SIP/228-08361358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/12-1' status is 'CONGESTION'
2) for no answer i get status is 'NOANSWER'
Accepting call from '06' to '534' on channel 0/4, span 1
-- Executing [534@default:1] Dial(Zap/4-1, SIP/228| 10) in new
stack -- Called 228
-- SIP/228-08362880 is ringing
-- Nobody picked up in 1 ms
== Auto fallthrough, channel 'Zap/4-1' status is 'NOANSWER'
3) for answered i don't get the status is 'answered'
Accepting call from '06' to '534' on channel 0/15, span 1
-- Executing [534@default:1] Dial(Zap/15-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08363bb8 is ringing
-- SIP/228-08363bb8 answered Zap/15-1
when i have the result is 'CONGESTION' or 'NOANSWER'i can go to the next
(home,s,1)
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = CONGESTION])
exten = 534,n,Goto(home,s,1)
how to do in order to go to the next if the result is answered
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
You're nearly there; you need to have a label answered in your dialplan.
This is done by inserting the name, in round brackets, after the priority
and
before the following comma. After a Goto() would be an excellent place to
put
it. Try this:
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
...
Note that if you answer the phone, as far as Asterisk is concerned, the
Dial()
statement is still being executed; so it won't fall through to the next
priority until the phone is hung up.
--
AJS
Answers come *after* questions.
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[asterisk-users] asterisk and IVR
Hello list,
i need your help about the IVR please
i have asterisk 1.4 installed and i configure an IVR like below
exten = 529,1,Ringing()
exten = 529,n,Wait(4)
exten = 529,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}welcome)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,goto(home,s,1)
exten = t,1,Goto(home,s,1)
exten = 1,1,Goto(call,s,1)
[call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 30)
exten = s,n,NoOp(User chose support option)
exten = s,n,MYSQL(Connect connid localhost database login password)
exten = s,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ menu\ SET\
callerid='${CALLERID(num)}'\, calldate=now()\, ext=no response\)
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,hangup
when i call the number 529 i can get the home and when i press 1 i get the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with the
ext = response or no response
thanks and regards
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Re: [asterisk-users] asterisk and IVR
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
any help please
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
i have asterisk 1.4 installed and i configure an IVR like below
. stuff deleted .
when i call the number 529 i can get the home and when i press 1 i get
the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with
the
ext = response or no response
You need to do this from an extension called h (which gets run when a
call
is hung up), in the same context where the call was placed. You can look
at
the variables ${DIALSTATUS} and ${HANGUPCAUSE} to see how the call went.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like
below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
Immediately after the Dial() statement, add a line like
exten = s,nNoOp(Dial status is ${DIALSTATUS})
That will show you the actual contents of ${DIALSTATUS} in the CLI (in
case
it is not what you are expecting). Call your extension a few times, and
see
exactly what you get when the line is answered, unanswered, engaged and
maybe
if the phone is unplugged.
Instead of having a separate extension named after every possible value of
${DIALSTATUS} it might be easier to use a GotoIf() statement to jump away
in
one case (most sensibly, if the call was answered), and fall through to
the
default otherwise (engaged and phone not connected are similar enough
to
no answer for that probably to be what you want, barring special values
--
feel free to use more GotoIf() statements if required).
Something like:
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = s,n,NoOp(execution continues here if no answer)
...
exten = s,n,Hangup()
exten = s,n(answered),NoOp(we jump here if call was answered)
...
exten = s,n,Hangup()
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Re: [asterisk-users] block certain numbers
hello
if you have just some numbers to block you can use the below code in your
dial plan
exten = 5xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 5xx,n,GotoIf($[${CALLERID(num)}=0661xx ]?3:4)
exten = 5xx,n,hangup
exten = 5xx,n,Dial(SIP/223, 30)
2013/6/17 A J Stiles asterisk_l...@earthshod.co.uk
On Monday 17 June 2013, binary dreamer wrote:
Hi.
i would like to manually create a list of numbers to block.
these numbers are from spammers (advertizers).
is there an easy way to send these particular numbers to busy or even
drop
the call?
Yes! Dead easy.
Use an external script, written in your favourite language, to look up the
number in some sort of database and return failure (exit 1) if it finds
it
there, or success (exit 0) if not. Call this with System() in dialplan.
If
the System() call succeeds (meaning the number was not found in the
database), Asterisk will move onto the next priority; if it fails
(meaning
the number was in the database) then it will move on by an extra 100.
Alternatively, you can read the value of ${SYSTEMSTATUS} to get the exit
code.
--
AJS
Answers come *after* questions.
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[asterisk-users] meetme configuration
hello list , i want to use meetme with asterisk1.4 i check in this forum and i found this code : exten = 508,1,MeetMe(1000,ipdM) when i use this code in my server i can say my name and i press 1 in order to enter in the conference ; but i want to asks the customer to press an number and password in order to join this conference could you please give me an example thanks and 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
[asterisk-users] sendmail when no response
hello list,
i need your help please regarding send mail i use astreisk 1.4;
i try to send mail when no response like below
exten = 5xx,1,Dial(SIP/223, 10)
exten = 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed'
myadresseem...@gmail.com)
when i launch the CLI i found :
You have new mail in /var/spool/mail/root
i check the root and i found :
Return-Path: root
Received: (from root@localhost)
by localhost.localdomain (8.13.1/8.13.1/Submit) id r55B3Deh023821;
Wed, 5 Jun 2013 11:03:13 GMT
Date: Wed, 5 Jun 2013 11:03:13 GMT
From: root root
Message-Id: 201306051103.r55B3Deh023821@localhost.localdomain
To: failed, myadresseem...@gmail.com
Subject: Call
test Email
--r55B3Dei023821.1370430193/localhost.localdomain--
could you please tell me how to do in order to send email to my address
gmail for example
thanks and regards
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Re: [asterisk-users] how to launch a URl when dialing a number
thanks justin i try to do this but the issue still the same.this link is
stored in my server 192.168.5.109 .but what i want to receive this link
when i call this number in my pc
ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored
thanks and regards
2013/5/30 Justin Killen jkil...@allamericanasphalt.com
**
If you just want the url to be opened (perhaps to update a counter via a
web service or cgi script), you can do this:
** **
system(“wget http://”)
or
system(“fetch http://...”)
** **
** **
** **
-Justin
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*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
Elharit
*Sent:* Thursday, May 30, 2013 8:07 AM
*To:* **Asterisk Users Mailing List - Non-Commercial Discussion**
*Subject:* [asterisk-users] how to launch a URl when dialing a number
** **
Hello
** **
i want to luanch an URL in my PC when i call a number like below
** **
exten = 066104,1,Set(CALLERID(number)=52xxx)
exten = 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten
=
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = 066104,n,http://192.168.5.109/interface2/interface2.php (
here i want to launch this url in my pc )
exten = 066104,n,Hangup()
** **
** **
thanks and regards
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Re: [asterisk-users] how to launch a URl when dialing a number
hello ,
thanks alex for your help and support the scenario is correct.
i will try to follow your suggestion and i will update you asap
thank you again for your explication i really appreciate it
2013/5/31 Alex Villacís Lasso a_villa...@palosanto.com
El 31/05/13 09:21, Salaheddine Elharit escribió:
thanks justin i try to do this but the issue still the same.this link is
stored in my server 192.168.5.109 .but what i want to receive this link
when i call this number in my pc
ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored
thanks and regards
2013/5/30 Justin Killen jkil...@allamericanasphalt.com
If you just want the url to be opened (perhaps to update a counter via
a web service or cgi script), you can do this:
system(“wget http://” http://%3F)
or
system(“fetch http://...” http://...%3F)
-Justin
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
Elharit
*Sent:* Thursday, May 30, 2013 8:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] how to launch a URl when dialing a number
Hello
i want to luanch an URL in my PC when i call a number like below
exten = 066104,1,Set(CALLERID(number)=52xxx)
exten
= 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten
=
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = 066104,n,http://192.168.5.109/interface2/interface2.php (
here i want to launch this url in my pc )
exten = 066104,n,Hangup()
From this discussion, I am guessing the following scenario. Please correct
me if I am wrong.
- There are (at least) three roles in your scenario: the Asterisk server,
the PHP webserver (which may or may not be the same machine as the Asterisk
server), and the client PC.
- Apparently your client PC runs a softphone (but the exact nature of the
telephony client is not important).
- A call is connected from the phone to your Asterisk, is directed to your
context, and dials some trunk (Zap/g1 in your snippet).
- You then want, somehow, to make the Asterisk server reach out to your
client PC (which runs a GUI and has a web browser) and force it to open an
arbitrary web page on the PHP webserver, presumably a callcenter data
collecting form.
The problematic issue is the last part. Especially the implication of
remotely opening a web page on some random PC.
If the above scenario is in fact what you were planning to do, maybe you
need to rethink your design. In the default case, there is no way to make a
remote PC open an arbitrary URL on its GUI. Think about the security
implications. You should instead have the web interface already open, and
program a Click2Call capability that contacts the Asterisk server and uses
AMI to execute an Originate action with your context as your target. Then
the web page would load your target URL in order to handle the call. Or, if
the calls come from an external source, you should program some kind of
monitor that alerts the web interface that the call was handled by the
context.
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[asterisk-users] how to launch a URl when dialing a number
Hello
i want to luanch an URL in my PC when i call a number like below
exten = 066104,1,Set(CALLERID(number)=52xxx)
exten = 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten
=
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = 066104,n,http://192.168.5.109/interface2/interface2.php ( here
i want to launch this url in my pc )
exten = 066104,n,Hangup()
thanks and regards
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Re: [asterisk-users] dahdi driver not getting install
hi You can download a tarball of the release here: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz 2013/5/11 Andrew Colin and...@vsave.co.za I thought he said rhel 6.3 Sent from my iPhone On 11 May 2013, at 2:48 PM, Asghar Mohammad asghar...@gmail.com wrote: he is using debian. debian have yum? On Sat, May 11, 2013 at 2:44 PM, Andrew Colin and...@vsave.co.za wrote: Do a yum install kernel-devel kernel-headers Reboot and it will work Sent from my iPhone On 11 May 2013, at 12:20 PM, Alec Davis siva...@paradise.net.nz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harish Mandowara Sent: Saturday, 11 May 2013 8:15 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi driver not getting install Dear, I have redhat enterprise linux 6.3. snip `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux/driver s/dahdi/firmware' You do not appear to have the sources for the 2.6.32-279.el6.x86_64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux' make: *** [all] Error 2 I'm a debian user after an inplace upgrade of Debian 6.0 to Debian 7.0, but had exactly that last night. From googling I reckon you need to install kernel-headers-2.6.32-279.el6.x86_64.rpm Alec Davis -- _ -- 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 -- _ -- 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] question about CDR
thanks asghar for your help and support and thanks ishfaq 2013/5/9 Asghar Mohammad asghar...@gmail.com hi, asterisk insert cdr when call is hangup and last dial statment, i dont understatnd why you are using 2 dial statment on same extenstion? if you you want dial to both extensions you can use 506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want to do failover the check Dial status and gotoif dialstatus = NO ANSWER or what ever you need. On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and 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 -- _ -- 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] question about CDR
hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and 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] question about CDR
thanks i verify but i don't understanding if can someone give me an example best regards 2013/5/9 Ishfaq Malik i...@pack-net.co.uk On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203| 506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 Hi You need to look into Channel Event Logging https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932 Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] hwo to stok variable wiith menu
hello list
i would your help please regarding this issue
with the below code i can store the call date and the callerid ,now i want
to store also the sip phone called 223
could you please see the code and tell me how can i add the sip phone in
my table 'Menu'
exten = 506,1,Ringing()
exten = 506,n,Dial(SIP/223, 30)
exten = 506,n,Goto(support,s,1)
[support]
exten = s,1,NoOp(User chose support option)
exten = s,n,MYSQL(Connect connid localhost aheevaccs aheevaccs aheevaccs)
exten = s,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ menu\ SET\
callerid='${CALLERID(num)}'\, calldate=now())
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
thanks and regards
2011/12/1 salaheddine elharit salah.elharit...@gmail.com
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn...@wi.rr.com
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
best regards
The example table that I gave originally was before I knew what you were
looking to do. I assumed, incorrectly that you simply wanted to track how
many times an option was selected in the menu.
I would recommend that you create a table specifically for this
application.
That table may look like this. Please name the table and columns
appropriately for your application.
create table option_three (
calldatedatetime,
calleridvarchar(40)
)
Then the sql would look something like this...
exten = 3,n,MYSQL(Query resultid ${connid} insert into option_three (
calldate, callerid ) values ( now(), '${CALLERID(num)}'))
Dale
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Re: [asterisk-users] WARNING[28151] from CLI
Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, yves -- _ -- 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] WARNING[28151] from CLI
thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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] WARNING[28151] from CLI
ok thanks for support and help 2013/3/27 Yves A. yves...@gmx.de you have already listed the two config files for using zaptel. on first sight, they look ok to me (did not use zaptel for years now) maybe you should definitely comment out any span that is not in use... or do the opposite. i´ve seen this warning several times, but i cant remember it had anything to do with spans being configured but not used. it always had something to do with timing or even defective cards or cabling or even wrong settings on providers´ site. what changes were made to the system so that these warnings occur? or have they been visible from the very start? do they affect telefony (e.g. loss of calls, one side audio only etc.)? how much load (concurrent calls) is on the asterisk, does the warning occur periodically or only a few times? these are all questions you should ask yourself to help you find the answer yourself... it can be very frustrating sometimes, but for me, thats all i can tell about. regards, yves Am 27.03.2013 13:06, schrieb Salaheddine Elharit: thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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 -- _ -- 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] WARNING[28151] from CLI
Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! -- _ -- 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] question about zapata.conf
ok thanks for your help and support i really appreciated 2013/3/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command /etc/asterisk/zapata.conf is a configuration ifle of Asterisk's chan_zap.so alone. So changes to it would generally require no more than restart of Asterisk. The simpler of them would be applied with a simple reload (or 'reload chan_zap.so' as you mention). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] question about zapata.conf
hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and 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] Need help about round-robin
thanks a lot i will test and i will update you as soon as i have any
problem
2013/3/22 Asghar Mohammad asghar...@gmail.com
your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out
using sip or any other tech. then use context defined in group 1 for
provider 1 and context defined in group 2 for provider 2.
2.
if your providers come in using sip just give him deferent ips, provider 1
send to wimax ip and provider to FH.
or explain if you are using other scenario.
On Fri, Mar 22, 2013 at 7:14 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works
without problem
now my question i have 2 providers i use g1 for the first and g2 for
the second
if i understand i must use r1 instead of g1 for the first provider and
r2 instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH
thanks and regards
2013/3/21 Asghar Mohammad asghar...@gmail.com
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten =
_0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i
use the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the
leading slash.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice:
+1-760-468-3867 PST
Newline Fax:
+1-760-731-3000
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Re: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and 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 -- _ -- 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] question about zapata.conf
thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and 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 -- _ -- 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] question about zapata.conf
ok thank you so much for your help and support 2013/3/25 Yves A. yves...@gmx.de hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and 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 -- _ -- 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 -- _ -- 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] Need help about round-robin
ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH
thanks and regards
2013/3/21 Asghar Mohammad asghar...@gmail.com
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the leading
slash.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST
Newline Fax:
+1-760-731-3000
--
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Re: [asterisk-users] Need help about round-robin
Hello bharat,
ok thank you so much for your help and support now i understand :)
2013/3/22 Bharat Lalcheta bharatlalch...@gmail.com
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works
without problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH
thanks and regards
2013/3/21 Asghar Mohammad asghar...@gmail.com
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i
use the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the
leading slash.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST
Newline Fax:
+1-760-731-3000
--
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Re: [asterisk-users] Need help about round-robin
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works
without problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH
thanks and regards
2013/3/21 Asghar Mohammad asghar...@gmail.com
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i
use the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the
leading slash.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST
Newline Fax:
+1-760-731-3000
--
__**__**
_
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[asterisk-users] Need help about round-robin
hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is difference between etc\zapataa.conf and etc\asterisk\zapata.conf i make this configuration just in etc\asterisk\zapata.conf i don't know if i must do this configuration also in etc\zapata.conf etc\asterisk\zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=X(my callerID) immediate=no channel = 156-170 channel = 172-176 channel = 32-46 channel = 48-62 etc\zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 #span=5,5,0,ccs,hdb3 # termtype: te #bchan=125-139,141-155 #dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us thank you so much -- _ -- 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] Need help about round-robin
i mean the burden-sharing between Wimax and FH 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com What do you mean by roundrobin here On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is difference between etc\zapataa.conf and etc\asterisk\zapata.conf i make this configuration just in etc\asterisk\zapata.conf i don't know if i must do this configuration also in etc\zapata.conf etc\asterisk\zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=X(my callerID) immediate=no channel = 156-170 channel = 172-176 channel = 32-46 channel = 48-62 etc\zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 #span=5,5,0,ccs,hdb3 # termtype: te #bchan=125-139,141-155 #dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us thank you so much -- _ -- 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] Need help about round-robin
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the leading
slash.
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Newline Fax: +1-760-731-3000
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Re: [asterisk-users] issue with inbound calls
thank you for your help the issue has been solved after disabled crc4 in etc/zaptel.conf from my side and from the FAI 2013/2/20 Justin Killen jkil...@allamericanasphalt.com ** When you add a card, it adds channels, so what used to be dahdi channel 1 is now probably channel 49 or 97. Look at /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf to see how you have it configured. I’m not sure what the zaptel equivalents are – my guess would be /etc/zaptel/system.conf and /etc/asterisk/zaptel-channels.conf ** ** -Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine Elharit *Sent:* Wednesday, February 20, 2013 10:33 AM *To:* **Asterisk Users Mailing List - Non-Commercial Discussion** *Subject:* [asterisk-users] issue with inbound calls ** ** hello list, ** ** i add a new diguim card in my server i use asterisk 1.4 with zaptel .conf* *** ** ** after that i can't receive the calls in my server with outbound calls there is no problem ** ** ** ** i have all time this error msg ** ** [Feb 20 18:15:48] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Feb 20 18:15:52] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Feb 20 18:15:56] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! ** ** ** ** any help please thank you ** ** ** ** ** ** [image: Images intégrées 1] -- _ -- 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 image002.gif-- _ -- 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] dahdi-linux dahdi-tools and libpri/libpri-
thank you so much for your response the issue was solved after using http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz best regards 2013/2/15 Russ Meyerriecks rmeyerrie...@digium.com /usr/src/dahdi-linux-2.6.1/drivers/dahdi/xpp/xdefs.h:152: error: conflicting types for âboolâ This issue is resolved by the latest dahdi-linux release 2.6.2-rc1. You can download a tarball of the release here: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz Or you can check out the v2.6.2-rc1 tag from git: git clone git.asterisk.org/dahdi/linux dahdi-linux cd dahdi-linux git checkout v2.6.2-rc1 -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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] round-robin in asterisk 1.4
I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210 with 2 port E1. now i bought another card Diguim TE410 and I want to add it the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP. the desired configuration : connection (WIMAX) and connection (radio beam) from the first ISP.from the second ISP no change (still have the fibre optic) my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic) any help please -- _ -- 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] round-robin in asterisk 1.4
thanks leandro how can i use that line in extensions.conf ? 2013/1/29 Leandro Dardini ldard...@gmail.com The simplest way is to use the Random function and to pickup one number from 1 to 3 and use that line. Leandro I am typing from my mobile phone... Il giorno 29/gen/2013 11:35, Salaheddine Elharit salah.elharit...@gmail.com ha scritto: I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210 with 2 port E1. now i bought another card Diguim TE410 and I want to add it the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP. the desired configuration : connection (WIMAX) and connection (radio beam) from the first ISP.from the second ISP no change (still have the fibre optic) my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic) any help please -- _ -- 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] block one number in incoming calls
Hello list could you please help me about one question. i have asterisk 1.4 installed, i configure the inbound call in my asterisk like below. exten = 520xx,1,Dial(SIP/224, 30). when the customer call my number (520xx) the sip phone 224 works without issue my problem i have a lot of calls coming from this number (0666xx) and i want to block it. if you can give me an example please . thanks and 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] block one number in incoming calls
thanks danny i think i didn’t explain correctly may question i revive a lot of calls from this number _0666XX and i wants to block it to call my number 520xx . 2013/1/14 Danny Nicholas da...@debsinc.com Exten = _0666XX,1,answer() Exten = _0666XX,n,playback(tt-monkeys) Exten = _0666XX,n,hangup() -- _ -- 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] block one number in incoming calls
hi Zohair Raza
thanks for your replay but this script will allow just this 0666XX to
call my number 520xx what i want is block this number to call 520xx
not allow it
thank you
exten = 520xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 520xx,2,GotoIf($[${CALLERID(num)} = 0666XX ]?3:4)
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
2013/1/14 Salaheddine Elharit salah.elharit...@gmail.com
thanks danny
i think i didn’t explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to block
it to call my number 520xx .
2013/1/14 Danny Nicholas da...@debsinc.com
Exten = _0666XX,1,answer()
Exten = _0666XX,n,playback(tt-monkeys)
Exten = _0666XX,n,hangup()
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Re: [asterisk-users] block one number in incoming calls
thanks a lot danny it works perfectly :) thanks a lot all
have a nice day
2013/1/14 Danny Nicholas da...@debsinc.com
Reverse the 3:4 and you will have the desired effect.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
Elharit
*Sent:* Monday, January 14, 2013 10:51 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] block one number in incoming calls
** **
hi Zohair Raza
** **
thanks for your replay but this script will allow just this 0666XX to
call my number 520xx what i want is block this number to
call 520xx not allow it
** **
thank you
** **
exten = 520xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 520xx,2,GotoIf($[${CALLERID(num)} = 0666XX ]?3:4)***
*
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
** **
2013/1/14 Salaheddine Elharit salah.elharit...@gmail.com
thanks danny
i think i didn’t explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to block
it to call my number 520xx .
** **
** **
** **
2013/1/14 Danny Nicholas da...@debsinc.com
Exten = _0666XX,1,answer()
Exten = _0666XX,n,playback(tt-monkeys)
Exten = _0666XX,n,hangup()
** **
** **
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Re: [asterisk-users] block one number in incoming calls
thanks all for your support and help a really appreciate it 2013/1/14 Carlos Alvarez car...@televolve.com So I'm not the only one who uses the monkeys as our place to send bad calls to. -- Sent from my iPhone On Jan 14, 2013, at 10:02 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 14 January 2013, Salaheddine Elharit wrote: i think i didn’t explain correctly may question i revive a lot of calls from this number _0666XX and i wants to block it to call my number 520xx . Use something like Exten = _520X./0666XX,1,Answer() Exten = _520X./0666XX,n,PlayBack(tt-monkeys) Exten = _520X./0666XX,n,HangUp() Now when a call comes in from 0666XX to _520X. they will get monkey noises. -- AJS Answers come *after* questions. -- _ -- 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] IVR: Dealing with database and returned variables
Hi Bilal
in my case i use an IVR menu using asterisk 1.4 an i can store the number
of the customer in my database and after i can select
the phone number and the date_time of calling i use mysql
you must change database login password with yours and also the name of
table
regards
exten = 500xx,1,Ringing()
exten = 500xx,n,Wait(4)
exten = 500xx,n,Goto(support,s,1)
[support]
exten = s,1,NoOp(User chose support option)
exten = s,n,MYSQL(Connect connid localhost database login password)
exten = s,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ table\ SET\
callerid='${CALLERID(num)}'\, calldate=now())
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,Dial(SIP/224, 30)
2012/3/7 bilal ghayyad bilmar...@yahoo.com
Hi All;
If I need to build IVR using Asterisk (so I will read and write to
database), until now from my reading, I can understand that the best way is
to use AGI to call external script like php which will manipulate every
thing, correct?
Well, the returned values from this script that I can use it to route the
call to the proper queue or Phone, how I can handle these returned values?
Do I have to store it in the database? Well, how I will read it from
database and use it in the extensions.conf?
From the other side, is there any tool to have IVR script (let us say,
studio programing) that can be used in Asterisk? Any advise in this way?
Regards
Bilal
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[asterisk-users] how to get the Record_ID
Hello List coud you please show me how to get the RECORD_ID for all outbond calls, i use asterisk 1.4 with database mysql thanks and 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] hwo to stok variable wiith menu
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn...@wi.rr.com
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
best regards
The example table that I gave originally was before I knew what you were
looking to do. I assumed, incorrectly that you simply wanted to track how
many times an option was selected in the menu.
I would recommend that you create a table specifically for this
application.
That table may look like this. Please name the table and columns
appropriately for your application.
create table option_three (
calldatedatetime,
calleridvarchar(40)
)
Then the sql would look something like this...
exten = 3,n,MYSQL(Query resultid ${connid} insert into option_three (
calldate, callerid ) values ( now(), '${CALLERID(num)}'))
Dale
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Re: [asterisk-users] hwo to stok variable wiith menu
thank you so much for you help,i have flowed your email and installed
thesesadd-ons all
works perfectly i can store the phone_number of the Customer ,now i can do
what i want :)
thanks every one for your support J
2011/11/30 Dale Noll dn...@wi.rr.com
On 11/28/2011 08:24 AM, salaheddine elharit wrote:
thank you for your help
You are welcome.
i would to ask you please, i want to store the phone number of the
customer in the option_name column when he press 3 in context menu
i have created a database aheevacss with user aheevaccs and password
aheevaccs and also i have creatd a table in this database name of table
test with two columns:
option_namevarchar(15)
countint
1-how can i check if the app_mysql module compiled and loaded i use
asterisk 1.4 and if not installed how can ido in order to install and
loaded it
I saw in some other message threads, it looks like you are working out
getting the mysql connectivity working in 1.4. In this version, it is an
'add on' that you have to download separately from the Asterisk source
tree. The instructions given by Warren Selby are correct.
When you do the 'make menuselect', you are presented with a menu with 5
options. Under 'Applications' you need to check app_addon_sql_mysql. Under
'Call Detail Recording' select cdr_addon_mysql. Under 'Resource Modules'
check res_config_mysql. Exit from menuselect and type 'make'. You
probably do not need the res_config_mysql, but it does not hurt anything to
compile it.
Aslo as mentioned in another thread, you do need to have mysql-devel
package installed.
Then run 'make' and 'make install' and 'make samples'. This will build
the modules app_addon_sql_mysql.so, cdr_addon_mysql.so and
res_config_mysql.so and install them in /usr/lib/asterisk/modules. This
does not change any existing modules, just adds the new ones.
Start an Asterisk cli (asterisk -r) and issue the command 'module load
app_addon_sql_mysql'. This should load the module and the MYSQL app will
be available in your dialplan. To verify it is loaded, you can issue the
command 'module show like sql'
You should also check the /etc/asterisk/modules.conf file. There should
be a line that says 'autoload=yes'. If it says no, you will have to add a
line 'load = app_addon_sql_mysql' (do not include the quotes). Note: If
you want to load cdr_addon_mysql, you will have to add a 'load =
cdr_addon_mysql' line as well. This file is read by asterisk at startup,
so after you restart asterisk for the first time after these changes, make
sure the module is loaded with the module show command.
2- can you please veify the menu below and tell me waht is wrong
thanks and regards
[default]
exten = 529,1,Ringing()
exten = 529,2,Wait(4)
exten = 529,3,Goto(accueil,s,1)
[accueil] ; définition d’un contexte pour l’accueil
exten = s,1,SetGlobalVar(sounds_path=/**var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}**welcome)
exten = s,3,goto(accueil,s,1)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,goto(accueil,s,1)
exten = t,1,Goto(accueil,s,1)
[menu]
exten = s,1,Background(${sounds_path}**menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,NoOp(User chose support option)
exten = 3,n,MYSQL(Connect connid localhost aheevaccs aheevaccs aheevaccs)
exten = 3,n,MYSQL(Query resultid ${connid} update test set count =
count + 1 where option_name = 'support')
exten = 3,n,MYSQL(Clear ${resultid})
exten = 3,n,MYSQL(Disconnect ${connid})
exten = 3,n,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
[appel] ; définition d’un contexte pour le menu d’appel
exten = s,1,Background(${sounds_path}**appel)
exten = s,2,WaitExten(10)
exten = 0,1,Goto(menu,s,1)
exten = 223,1,Dial(SIP/${EXTEN},20,tr)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(appel,s,1)
exten = t,1,Goto(appel,s,1)
[message] ; définition d’un contexte pour la messagerie
exten = s,1,VoiceMailMain(${**CALLERIDNUM})
exten = t,1,Hangup()
[support] ; définition d’un contexte pour le support
exten = s,1,GoToIfTime(09:00-17:00|**mon-fri|*|*?s,4)
exten = s,2,Playback(${sounds_path}no-**relation-support)
exten = s,3,Goto(menu,s,1)
exten = s,4,Playback(${sounds_path}**relation-support)
exten = s,5,Queue(default)
exten = t,1,Hangup()
In the [accueil] context, you call Background with the name of the file to
play, then immediately return to the top and play the message again, and
again and again. It will never stop until the caller hangs up. Also, you
are asking the caller to press the '#' key to get past the welcome greeting
before getting to the main menu. I would recommend playing the welcome
followed immediately by the Background() for the menu. The call the
WaitExten() to give
