Re: [asterisk-users] Anyone doing speech to text?

[Salaheddine Elharit]

hi
 you can try this link

http://zaf.github.io/asterisk-googletts/


2015-08-26 19:15 GMT+01:00 Tech Support :

> 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
>
>
>
>
>
> --
> _
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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Salaheddine Elharit]

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 :

>  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=""NodePhone"> >"*") in new stack
>> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> >*") in new stack
>> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> -- 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}"

Re: [asterisk-users] call between snom 300 and aastra 6731i

[Salaheddine Elharit]

s@sub-flp-5:2] GotoIf("SIP/300-0192", "0?match") in
new stack
-- Executing [s@sub-flp-5:3] Return("SIP/300-0192", "") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/300-0192",
"OUTNUM=00XX17621") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/300-0192",
"custom=SIP/FD") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/300-0192",
"0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/300-0192",
"0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/300-0192",
"dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1]
MacroExit("SIP/300-0192", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/300-0192",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/300-0192",
"1?Set(CONNECTEDLINE(num,i)=00XX17621)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/300-0192",
"1?Set(CONNECTEDLINE(name,i)=CID:300)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/300-0192",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/300-0192",
"SIP/FD/00XX17621,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/FD/00XX17621
-- Got SIP response 480 "No address found" back from 217.195.XX.XXX:5060
-- SIP/FD-0193 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/300-0192", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 19")
in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/300-0192",
"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/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-now&pls-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-now&pls-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-now&pls-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 

Re: [asterisk-users] call between snom 300 and aastra 6731i

[Salaheddine Elharit]

please no body has som with aastra can help me in this issue

2015-03-26 11:02 GMT+00:00 Salaheddine Elharit :

> 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.
>
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[asterisk-users] call between snom 300 and aastra 6731i

[Salaheddine Elharit]

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.
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Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

[Salaheddine Elharit]

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 :

> ** 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 .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

[Salaheddine Elharit]

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 :

> On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
>  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-now&pls-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-now&pls-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-now&pls-try-call-later, noanswer
> > -- Executing [s@macro-outisbusy:5] Congestion("SIP/306-00b8",
> 

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

[Salaheddine Elharit]

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-now&pls-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-now&pls-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-now&pls-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
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Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

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 :

> 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 --
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>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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> _
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Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

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 :

> 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 :
>
>> 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&q

Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

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.
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[asterisk-users] outbound calls

[Salaheddine Elharit]

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/1

Re: [asterisk-users] chanspy for group extension

[Salaheddine Elharit]

thank you so much Carlos ;the issue has been solved

Best Regards.

2015-03-12 18:40 GMT+00:00 Salaheddine Elharit :

> 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 :
>
>> 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
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] chanspy for group extension

[Salaheddine Elharit]

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 :

> 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
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] chanspy for group extension

[Salaheddine Elharit]

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 :

> 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 :
>
>> 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

[Salaheddine Elharit]

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 :

> 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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> _
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[asterisk-users] chanspy for group extension

[Salaheddine Elharit]

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

[Salaheddine Elharit]

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.
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[asterisk-users] issue with inbound route

[Salaheddine Elharit]

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
--  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
-- AGI 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
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[asterisk-users] asterisk and elastix

[Salaheddine Elharit]

Hello list,



i have installed elastix 2.4.0 with call center model and i have
created an 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.
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Re: [asterisk-users] how to stop asterisk using a call

[Salaheddine Elharit]

thanks a lot it works correctly


2014-04-07 12:08 GMT+00:00 Andres :

>  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
>
>
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[asterisk-users] how to stop asterisk using a call

[Salaheddine Elharit]

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.
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Re: [asterisk-users] h extension isn't processed after call file finishes.

[Salaheddine Elharit]

hello,

try to use failed instead of h

exten => failed,1,

best regards.




2014-02-18 9:09 GMT+00:00 Ishfaq Malik :

> What version of asterisk are you using?
>
> Ish
>
>
> On 17 February 2014 20:49, Mike Diehl  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 --
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>
>
>
> --
>
> 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
>
>
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Re: [asterisk-users] auto-answer call

[Salaheddine Elharit]

hi

when i try to this with page()

exten => 506,1,SIPAddHeader("Call-Info:__\; answer-after=0")
exten => 506,n,page(SIP/105)

CLI>Accepting 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
--  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 :

> 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 > <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.
>
>
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Re: [asterisk-users] auto-answer call

[Salaheddine Elharit]

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 :

> 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
>
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[asterisk-users] auto-answer call

[Salaheddine Elharit]

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
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[asterisk-users] callfiles.call

[Salaheddine Elharit]

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
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[asterisk-users] go to context from server 1 to server 2

[Salaheddine Elharit]

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
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Re: [asterisk-users] send the calls from to servers

[Salaheddine Elharit]

i attached file my dialplan


2013/12/20 Salaheddine Elharit 

> in attached file my dialplan
>
> thanks and regards
>
>
>
>
> 2013/12/20 Eric Wieling 
>
>> 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 
>>
>>
>>
>> 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 
>>
>>
>> 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
>> ht

Re: [asterisk-users] send the calls from to servers

[Salaheddine Elharit]

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 

>
> 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 
>
>
> 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--
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> Thurs:
>http://www.asterisk.org/hello
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>
>
>
>
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Re: [asterisk-users] send the calls from to servers

[Salaheddine Elharit]

i ask about outbound calls not inbound round-robin

best regards


2013/12/19 Eric Wieling 

> 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
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] send the calls from to servers

[Salaheddine Elharit]

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
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

> 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

[Salaheddine Elharit]

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 

> 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 
>> 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 
>>
>>> Try following in chan_dahdi
>>>
>>> immediate = yes
>>> echocancel = no
>>> dtmfmode = auto
>>>
>>> Mitul
>>> On Nov 29, 2013 1:42 PM,  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 
>>>> 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
>>>>  
>>>> 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
>>>>
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>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>
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>>>> _
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>>>>
>>>
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>>>
>>
>>
>>
>
> --
> Regards,
> Mitul Limbani,
> 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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[asterisk-users] Fwd: issue with speech in IVR

[Salaheddine Elharit]

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 
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 

> Try following in chan_dahdi
>
> immediate = yes
> echocancel = no
> dtmfmode = auto
>
> Mitul
> On Nov 29, 2013 1:42 PM,  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 
>> 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
>>  
>> 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

> Try following in chan_dahdi
>
> immediate = yes
> echocancel = no
> dtmfmode = auto
>
> Mitul
> On Nov 29, 2013 1:42 PM,  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 
>> 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
>>  
>> Subject: Re: [asterisk-users] issue with speech in IVR
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
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>
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

>  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.
>
>
> --
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

>
>
>
> 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
>
>
>
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

> 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 --
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Re: [asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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 

> 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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[asterisk-users] issue with speech in IVR

[Salaheddine Elharit]

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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Re: [asterisk-users] issue with dahdi_channels.conf

[Salaheddine Elharit]

thanks for your response i will swap the cables and i will update by the
result

best regards


2013/10/31 Tony Mountifield 

> In article <
> cahexamsp4nenuntymuzwjgep69v+7rb7ekbyzsalmbm+zyo...@mail.gmail.com>,
> Salaheddine Elharit  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
>
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Re: [asterisk-users] issue with dahdi_channels.conf

[Salaheddine Elharit]

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 

> 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.
>
> --
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[asterisk-users] issue with dahdi_channels.conf

[Salaheddine Elharit]

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
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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

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 

> ok thanks for your comment i really appreciate it
>
>
> best regards
>
>
> 2013/10/23 Russ Meyerriecks 
>
>> 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

ok thanks for your comment i really appreciate it


best regards


2013/10/23 Russ Meyerriecks 

> 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 --
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>http://www.asterisk.org/hello
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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

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 

> 2013/10/22, A J Stiles :
> > 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
>
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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

2013/10/22, A J Stiles :
> 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

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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

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 

>  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
>
>
> --
> _
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[asterisk-users] issue after install dahdi

[Salaheddine Elharit]

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
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Re: [asterisk-users] (no subject)

[Salaheddine Elharit]

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 

> 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
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] (no subject)

[Salaheddine Elharit]

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

[Salaheddine Elharit]

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 

> 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.)
>
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
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>
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Re: [asterisk-users] asterisk and IVR

[Salaheddine Elharit]

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 

> 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

[Salaheddine Elharit]

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
--  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 

> * 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.
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
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Re: [asterisk-users] asterisk and IVR

[Salaheddine Elharit]

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 

> * 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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Re: [asterisk-users] asterisk and IVR

[Salaheddine Elharit]

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 

> ok thank you i will verify and i will update you
>
> thanks for your help
>
>
> 2013/7/25 A J Stiles 
>
>> 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()
>>
>>
>> --
>> _
>> -- 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
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] asterisk and IVR

[Salaheddine Elharit]

ok thank you i will verify and i will update you

thanks for your help


2013/7/25 A J Stiles 

> 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

[Salaheddine Elharit]

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 

> 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.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] asterisk and IVR

[Salaheddine Elharit]

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] block certain numbers

[Salaheddine Elharit]

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 

> 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

[Salaheddine Elharit]

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
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[asterisk-users] sendmail when no response

[Salaheddine Elharit]

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: 
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 
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

[Salaheddine Elharit]

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 

>  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 
>
>>  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.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] how to launch a URl when dialing a number

[Salaheddine Elharit]

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 

> **
>
> 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 
>   --
>
> *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
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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[asterisk-users] how to launch a URl when dialing a number

[Salaheddine Elharit]

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

[Salaheddine Elharit]

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 

> I thought he said rhel 6.3
>
> Sent from my iPhone
>
> On 11 May 2013, at 2:48 PM, Asghar Mohammad  wrote:
>
> he is using debian. debian have yum?
>
>
> On Sat, May 11, 2013 at 2:44 PM, Andrew Colin  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  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.
>> >
>> > 
>> >
>> >> `/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 --
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>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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>
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Re: [asterisk-users] question about CDR

[Salaheddine Elharit]

thanks asghar for your help and support  and thanks ishfaq


2013/5/9 Asghar Mohammad 

> 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/223&SIP/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 --
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>>http://www.asterisk.org/hello
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] question about CDR

[Salaheddine Elharit]

 thanks i verify but i don't understanding if can someone give me an example

best regards




2013/5/9 Ishfaq Malik 

> 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 
> 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
>
>
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[asterisk-users] question about CDR

[Salaheddine Elharit]

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
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Re: [asterisk-users] hwo to stok variable wiith menu

[Salaheddine Elharit]

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 

> Hi Noll,
>
> all works perfectly thanks a lot for your help and support i really
> appreciate it :)
>
> Best Regards
>
> 2011/12/1 Dale Noll 
>
>>
>> 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
>>
>> --
>> "The truth speaks for itself. I'm just the messenger."
>> Lyta Alexander - Babylon 5
>>
>>
>> --
>> __**__**_
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>>
>
>
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Re: [asterisk-users] WARNING[28151] from CLI

[Salaheddine Elharit]

ok thanks for support and help

2013/3/27 Yves A. 

>  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 
>
>>  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
>>
>&g

Re: [asterisk-users] WARNING[28151] from CLI

[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 

>  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. 
>
>>  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,
>>
>
>
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Re: [asterisk-users] WARNING[28151] from CLI

[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. 

>  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 --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

ok thanks for your help and support i really appreciated

2013/3/26 Tzafrir Cohen 

> On Mon, Mar 25, 2013 at 10:44:47AM +0000, 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
>
> --
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[asterisk-users] WARNING[28151] from CLI

[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!
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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

ok thank you so much for your help and support

2013/3/25 Yves A. 

>  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 
>
>> 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. 
>>
>>
>> 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>>
>> _
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>> Thurs:
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>>
>>
>>
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>>
>
>
>
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>
>
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Re: [asterisk-users] question about zapata.conf

[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 

> 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. 
>
>
> 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 --
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> every Thurs:
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>
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>
>
>
> --
>
> _
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> Thurs:
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>
>
>
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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

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. 

>  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 --
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>http://www.asterisk.org/hello
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>
>
>
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

thanks a lot i will test and i will update you as soon as i have any
problem

2013/3/22 Asghar Mohammad 

> 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 
>>
>>> 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 
>>>>
>>>>> 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 
>>>>>>
>>>>>>> File is ok there is no etc/zapata file.
>>>>>>> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
>>>>>>> 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,
>>>>>>>> --**--**

[asterisk-users] question about zapata.conf

[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
--
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

yes i want to use the burden-sharing between Wimax and FH using a diguim
cards

2013/3/22 Asghar Mohammad 

> 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 
>>
>>> 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 
>>>>
>>>>> File is ok there is no etc/zapata file.
>>>>> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
>>>>> 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
>>>>>>
>>>>>> --
>>>>>> __**__**
>>>>>> _
>>>>>> -- 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<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
>>>>>
>>

Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

Hello bharat,

ok thank you so much for your help and support now i understand :)

2013/3/22 Bharat Lalcheta 

> 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" 
> 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 
>>
>>> 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 
>>>>
>>>>> File is ok there is no etc/zapata file.
>>>>> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
>>>>> 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
>>>>>>
>>>>>> --
>>>>>> __**__**
>>>>>> _
>>>>>> -- 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<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 vi

Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

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 

> 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 
>>
>>> File is ok there is no etc/zapata file.
>>> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
>>> 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
>>>>
>>>> --
>>>> __**__**
>>>> _
>>>> -- 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<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
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>
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

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 

> File is ok there is no etc/zapata file.
> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
> 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
>>
>> --
>> __**__**_
>> -- 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<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
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

i mean the burden-sharing between Wimax and FH

2013/3/21 Bharat Lalcheta 

> What do you mean by roundrobin here
> On Mar 21, 2013 8:27 PM, "Salaheddine Elharit" 
> 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
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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> _
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[asterisk-users] Need help about round-robin

[Salaheddine Elharit]

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
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Re: [asterisk-users] issue with inbound calls

[Salaheddine Elharit]

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 

> **
>
> 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]
>
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Re: [asterisk-users] dahdi-linux dahdi-tools and libpri/libpri-

[Salaheddine Elharit]

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 

> > /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
>
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Re: [asterisk-users] round-robin in asterisk 1.4

[Salaheddine Elharit]

thanks leandro

how can i use that line  in extensions.conf ?

2013/1/29 Leandro Dardini 

> 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
>>
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[asterisk-users] round-robin in asterisk 1.4

[Salaheddine Elharit]

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
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Re: [asterisk-users] block one number in incoming calls

[Salaheddine Elharit]

thanks all for your support and help a really appreciate it

2013/1/14 Carlos Alvarez 

> 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 
> 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.
> >
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Re: [asterisk-users] block one number in incoming calls

[Salaheddine Elharit]

thanks a lot danny it works perfectly :) thanks a lot all


have a nice day

2013/1/14 Danny Nicholas 

> 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 
>
> 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 
>
> 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

[Salaheddine Elharit]

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 

> 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 
>
>> 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

[Salaheddine Elharit]

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 

> Exten => _0666XX,1,answer()
>
> Exten => _0666XX,n,playback(tt-monkeys)
>
> Exten => _0666XX,n,hangup()
>
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[asterisk-users] block one number in incoming calls

[Salaheddine Elharit]

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
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Re: [asterisk-users] IVR: Dealing with database and returned variables

[salaheddine elharit]

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 

> 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

[salaheddine elharit]

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
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Re: [asterisk-users] hwo to stok variable wiith menu

[salaheddine elharit]

Hi Noll,

all works perfectly thanks a lot for your help and support i really
appreciate it :)

Best Regards

2011/12/1 Dale Noll 

>
> 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
>
> --
> "The truth speaks for itself. I'm just the messenger."
> Lyta Alexander - Babylon 5
>
>
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Re: [asterisk-users] hwo to stok variable wiith menu

[salaheddine elharit]

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

2011/11/30 salaheddine elharit 

>  thank you so much for you help,i have flowed your email and installed
> theses add-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 
>
>> 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 ${resul