search for: from_pstn

I am using a combo of static files and Asterisk Realtime configuration. This section works fine when a static file: --------------------------- [from_pstn] ;Voipgate exten => 4507,1,Goto(from_pstn,s,1) exten => s,1,Macro(dial-ext) exten => s,2,Hangup --------------------------- But, when I drop it in the database and try it in Realtime mode I get this error: --------------------------- -- Accepting AUTHENTICATED call from 80.127.191.55, r...

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus

...just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000> deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0.154:5062 [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer exten => s,2,D...

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

...n 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone = us defaultzone = us [root at jekyll ~]# cat /etc/asterisk/chan_dahdi.conf [general] [channels] ; Span 1 group = 1 context = from_pstn switchtype = qsig signalling = pri_net channel => 1-23 context = default ; Span 2 group = 2 context = from_avaya switchtype = qsig signalling = pri_net channel => 25-47 context = default ; Span 3 group = 7 context = from_pstn switchtype = qsig signalling = pri_cpe channel => 49-71 context...

...agree that I might need to CUT part of the R-URI, but I don't need access to any other header to find the info I need. When the call arrives at the Asterisk right now, this is the exten/context that it is hitting, so it already has the info I need: Executing [9135041291;rn=+19136630000;npdi at from_pstn:1] As far as I can tell, I think that I just need to figure out how to make an extension entry that matches on the "rn=+19136630000\;npdi" and then moves to another context (or same one) with ${EXTEN,0,10}. I just can't get that first extension to match on the RN value. > >...

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the line exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => s,3,BackGround(demo-instruct) ; Play some instructions What setting is causing the 13-15sec. delay? #Jose...

Hello, Straight to the point. rxgain=20 causes dialplan extensions not to work from a nortel pbx, while rxgain=15 works fine. In both cases a standard analog phone can dial an extension without problem. from zapata.conf signalling=fxs_ks context => from_pstn amaflags => documentation echocancel=yes rxgain=15 channel => 1 Anyone have any ideas? This is more to bounce around than find a solution for, I am alright with an rxgain of 15. I am just curious as to why this is happening.

Hi, all I'd like to configure Asterisk to receiving call from PSTN. After PSTN phone call in, Asterisk will prompt user to enter a number, then Asterisk will transfer the call to a SIP phone by this number. Please help me check the following extensions, is that OK? thanks! [from_pstn] exten => _.,1,Answer() exten => _.,2,GoTo(Xfer,s,1) [Xfer] exten => s,1,Background(privacy-prompt) exten => _.,1,Dial(SIP/${EXTEN}@$(OutboundProxy), 30) Any suggestion? I am really not sure that whether $(EXTEN) can refer to the numbers that entered after prompt? Thanks!...

...442010000 AGI Tx >> agi_calleridname: unknown AGI Tx >> agi_callingpres: 3 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 481 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from_pstn AGI Tx >> agi_extension: AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: As we can see here, the DNID and CALLERID are swapped (idk, maybe it is intended behavior). However, that is not a problem. The problem is - it shows 481, but...

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier