similar to: Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers

here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: > here is what I have: > > exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) > > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) > > exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) > > not having success; > >

Motty Yes From your dial plan accept 9 + 7 digits then concat your dialed number together with your areacode. This s a brief example. exten => _9XXXXXXX,1,Set(l_HomeAreaCode=555) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This line should combine your area code and the last 7 digits of your dialed phone number exten =>

this code worked for me, here is what I did and worked for me: exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) Thanks for you help! On 04/27/2015 02:56 PM, Matt Riddell wrote: > >> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com >> <mailto:motty.cruz at gmail.com>> wrote:

On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz <motty.cruz at gmail.com> wrote: > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

06.09.2016 16:42, George Joseph ?????: > > > On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Several months server working on asterisk 13.7 and pjproject 2.5 > (installed separately). Once a day the server crashes or hangs and > is familiar sores that written

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to

Hi I want to use Prefix app in extensions but get this error: WARNING[9255] pbx.c: No application 'Prefix' for extension ... I am just want to do somethig like this: exten => _9XXXXXXX,1,ANSWER() exten => _9XXXXXXX,2,Wait(1) exten => _9XXXXXXX,3,Prefix(511) exten => _5119XXXXXXX,4,DeadAGI(a2billtest.php|1) exten => _5119XXXXXXX,5,Hangup() Please someone tell me how to

Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>

For outgoing calls made on our PRI circuit we are setting the Caller ID using the format Exten => _9XXXXXXX,1,SetCallerID(1601XXXXXXX) The monitor shows that the CallerID is being set to the specified number, but yet when the call is received on the user end the ID is always the base number of our DID. For example we have 8600-8650 as DID's but the callerid is always 8600 regardless of

Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head against the desk here... _9XXXXXXX lets me make an outbound call, but _9X. only lets me dial 9 plus