search for: allowed_passed_screen

...s When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel: SIP/outbound.vitelity.net/8005555555 Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable: CALLERID(num-pres)=allowed_passed_screened Async: true I translated those settings to the following for pjsip.conf... [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = yes contact = sip:64.2.142.93 at 5060 [outbound.vitelity.net] type = endpoint context = TestApp transpor...

...rough Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=...

...the sip.conf they sent > me, everything works. > > Action: Originate > ActionID: S8 > Channel: SIP/outbound.vitelity.net/8005555555 > Exten: createcall > Context: TestApp > Priority: 1 > Timeout: 60000 > CallerID: John Doe <1234> > Variable: CALLERID(num-pres)=allowed_passed_screened > Async: true > > > I translated those settings to the following for pjsip.conf... > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > [outbound.vitelity.net] > type = aor > remove_existing = yes > contact = sip:64.2.142.93 at 5060 &...

...; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen ; Set caller ID presentation ; See doc/callingpres.txt for more information [22] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=22 ;callerid=John Doe <1234> ; Full caller ID, to override th...

...s screen=yes. Remote-Party-ID: "Jane Done" <sip:8005551234 at xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no For the AMI Originate, I have been passing variables in an attempt to modify the CALLERID(name-pres). My understanding is that a variable of "CALLERID(name-pres)=allowed_passed_screen" should result in the RPID screen setting being yes. I have tried many different values for this variable, but the RPID line is always "screen=no". What am I missing to force the screen=yes to be passed as part of the Remote-Party-ID? Dan -------------- next part -------------- A...

...ation data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002 Exten: createcall Context: MyContext Priority: 1 Timeout: 60000 CallerID: CustomerName <########## > Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 Async: true Is there a setting that's required on the PJSIP endpoint to allow overwriting the INVITE packet's Contact header? Is there something else I am missing to perform this? Have a great day! Dan -------------- next part -------------- An HTML...

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

...;limitonpeers=yes [100] type=peer context=demo callerid=Back Office <100> username=100 secret=(Private) host=dynamic nat=no qualify=yes canreinvite=no dtmfmode=rfc2833 call-limit=5 mailbox=100 at default disallow=all allow=ulaw allow=alaw ;allow=g723.1 allow=g729 ;callingpres=allowed_passed_screen notifyringing=no callgroup=1 pickupgroup=1 Asterisk CLI: Extension Changed 6100[demo] new state Ringing for Notify User 105 Extension Changed 6100[demo] new state Ringing for Notify User 104 Extension Changed 6100[demo] new state Ringing for Notify User 102 Extension Changed 6100[demo] new state R...

...eturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no switchtype=5ess context=main signalling=pri_cpe group=1 channel => 1-23 channel => 25-47 exten => 1234,1,Wait,1 exten => 1234,2,Answer exten => 1234,3,SetCallerPres(allowed_passed_screen) exten => 1234,4,SetCIDNum(8881234567) exten => 1234,5,Dial(Zap/g1/18887654321,,,) exten => 1234,6,Hangup Thanks, Shaun Tierney

...c27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten => _X.,2,SetCallerPres(allowed_passed_screen) exten => _X.,3,Dial(mISDN/g:E1/${EXTEN},40) exten => _X.-BUSY,4,Busy(1) But whenever a sip client calls to an exten that is busy through e1 I get busy tones for 10s before I get disconnected. But I want to have it only for 1s. Does anyone know how to fix that? regards, Christophorus

...ormats=ulaw&timeout=30&callerId=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200807/a56aec37/attachment.html>

...using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel: SIP/outbound.vitelity.net/8005555555<http://outbound.vitelity.net/8005555555> Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable: CALLERID(num-pres)=allowed_passed_screened Async: true I translated those settings to the following for pjsip.conf... [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net<http://outbound.vitelity.net>] type = aor remove_existing = yes contact = sip:64.2.142.93 at 5060 You might want to set a qualif...

...his with AMI successfully. > > Action: Originate > > ActionID: S40 > > Channel: PJSIP/1003 at 1003 > > Exten: createcall > > Context: IS > > Priority: 1 > > Timeout: 60000 > > CallerID: Dan Cropp <291> > > Variable: > CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 > > Async: true > > > > Dan > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: &g...

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

...his with AMI successfully. > > Action: Originate > > ActionID: S40 > > Channel: PJSIP/1003 at 1003 > > Exten: createcall > > Context: IS > > Priority: 1 > > Timeout: 60000 > > CallerID: Dan Cropp <291> > > Variable: > CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 > > Async: true > > > > Dan > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum a...

...ormats=ulaw&timeout=30&callerId=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200807/b2ed1087/attachment.html>

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed

...> immediate=no > > switchtype=5ess > > context=main > > signalling=pri_cpe > > group=1 > > channel => 1-23 > > channel => 25-47 > > > > exten => 1234,1,Wait,1 > > exten => 1234,2,Answer > > exten => 1234,3,SetCallerPres(allowed_passed_screen) > > exten => 1234,4,SetCIDNum(8881234567) exten => > > 1234,5,Dial(Zap/g1/18887654321,,,) > > exten => 1234,6,Hangup > > > > Try something like this... > exten => _1NXXXXXXXXX,1,SetCallerID(8881234567|a) > exten => _1NXXXXXXXXX,2,SetCIDName(MyN...