Displaying 20 results from an estimated 53 matches for "elharit200".
2015 Mar 26
2
call between snom 300 and aastra 6731i
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and
2010 Jun 18
6
asterisk issue
Hello,
I have a problem in Asterisk 1.4 each day I need to restart *asterisk
service asterisk* restart in order to unblock the calls
My question how can I do in order to check the issue, and if there is any
tool or log?
Thanks and regards.
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2011 Apr 04
2
call forwarding
Hello list,
i have one question related to call forwarding.
i have 2 number for the inbound and i want to configure asterisk like that.
When the customer call the first number 0522XXXXXX the call will be
forwarding automatically to anther number 0520xxxxxx
Does anybody have a solution to this problem.
Thanks and Regards.
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2011 May 30
3
please help
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten => _0678922645.,1,Set(CALLERID(number)=520460587)
exten => _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten => _067892264*5*,2,Hangup()
i can not call my
2015 Mar 20
2
outbound calls
i noticed that when i active the voicemail in the IP-phone where the number
0033149xxxxxx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording
SIP/101-0000010d
-- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> 0x2b393cfc2610 -- Probation passed
2015 Mar 27
2
call between snom 300 and aastra 6731i
...ially which
codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
> please no body has som with aastra can help me in this issue
>
> 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit
> <salah.elharit200 at gmail.com <mailto:salah.elharit200 at gmail.com>>:
>
> hello list
>
> i need your help please regarding an issue with snom300 and
> aastra6731i using asterisk
>
> 11.13.0 asterisk
>
> snom 300 8.7.3.25
>
> astra 6731i 2.6.0.2019...
2015 Mar 12
0
chanspy for group extension
...spy]
exten => _0071XX,*1,*Macro(chanspy,1234)
exten => _0072XX,*1,*Macro(chanspy,5678)
exten => _0073XX,*1,*Macro(chanspy,8910)
but when i do 007100 for exemple i spy another agnet 102 or 103
any help please
thanks and regards
2015-03-12 10:30 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>:
> thank you so much it work
> you must add 1 like below
>
> [app-chanspy]
> exten => _0071XX,*1,*Macro(chanspy,1234)
> exten => _0072XX,*1,*Macro(chanspy,5678)
> exten => _0073XX,*1,*Macro(chanspy,8910)
>
>
> best regards.
>
> 2015-03-...
2015 Mar 20
0
outbound calls
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.elharit200 at gmail.com> wrote:
> i noticed that when i active the voicemail in the IP-phone where the
> number 0033149xxxxxx is configured i can call this number without issue
>
> Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149xxxxxx == Begin Mi...
2015 Mar 27
0
call between snom 300 and aastra 6731i
please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>:
> hello list
>
> i need your help please regarding an issue with snom300 and aastra6731i
> using asterisk
>
> 11.13.0 asterisk
>
> snom 300 8.7.3.25
>
> astra 6731i 2.6.0.2019
>
> i have configured the trunks like below
>
> 100 in snom3...
2013 May 30
0
asterisk-users Digest, Vol 106, Issue 41
...try
exten = .....,n,System(wget -P /var/log/asterisk/wgets
'http://theUrlYouWantToCall' &)
kind regards,
andre
Am 30.05.2013 19:00, schrieb asterisk-users-request at lists.digium.com:
> Message: 9
> Date: Thu, 30 May 2013 15:06:59 +0000
> From: Salaheddine Elharit <salah.elharit200 at gmail.com>
> Subject: [asterisk-users] how to launch a URl when dialing a number
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <CAHexamt4=eVwLJ7pN90mgqvirN-_mnyH6p2TdHTpq4D89E2Ztg at mail.gmail.co...
2014 Jan 31
2
callfiles.call
hello list,
i have created a callfiles with my asterisk 1.4.43 like:
Channel: SIP/watara/06xxxxxxxx
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
2013 Oct 31
2
issue with dahdi_channels.conf
Hello list
i have an issue with my dahdi_channels.conf
in span 1 when i use it like below i can do my outband calls without issue
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 17-31
context = default
group = 63
but when i add in channel 1-15 like: channel => 1-15,17-31
i receive all
2013 May 09
2
question about CDR
hello list,
i need your help about cdr ,i have installed the module cdr in my asterisk
1.4 .
for the inbound calls when i call my sip exten like below :
exten => 506,1,Dial(SIP/223, 10)
exten => 506,n,Dial(SIP/276, 10)
in CDR i have just one line with SIP /276 the last line but there is
no historic
for the first SIP 223
recid Record ID | calldate |clid |src
2015 Mar 12
2
chanspy for group extension
thank you so much it work
you must add 1 like below
[app-chanspy]
exten => _0071XX,*1,*Macro(chanspy,1234)
exten => _0072XX,*1,*Macro(chanspy,5678)
exten => _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>:
> On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
>
>> hello list,
>>
>> i use
2015 Feb 26
1
issue with inbound route
hello liste
i have creat i trunk sip and inboun route for inbound calls the issue whe i
use the DID in inboud route i have a error No DID or CID Match.
but when i leave this DID field blank i can route the call without any issue
how can ido in order to use DID in route inboud "i use elastix"
Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same provider in my asterisk
i can call all numbers just the numbers are configured in thses ip phones.
but when i configured the same trunk in x-lite i can call theses ip-phones
without
2015 Mar 25
0
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
<salah.elharit200 at gmail.com> wrote:
> hello list,
>
> i have asterisk 11.15.0 and i have some trunks sip from my provider
>
> we have some ip phone astra 6731i
>
> each Ip-phone is configured with trunk and we call
>
> no ihave configured another trunk from the same provider in my as...
2013 Nov 27
3
issue with speech in IVR
hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}music1)
exten => s,n,Background(${sounds_path}music2)
exten => s,n,Background(${sounds_path}music3)
exten =>
2015 Mar 12
5
chanspy for group extension
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a ?crit :
> 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
2015 Mar 20
0
outbound calls
...fa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
-- SIP/FD-0000010e answered SIP/101-0000010d
> 0x1d08efa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
thanks and regards.
2015-03-20 18:39 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>:
> thank you
>
> i noticed that when i active the voicemail in the IP-phone where the
> number 0033149xxxxxx is configured i can call this number without issue
>
> Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149xxxx...