search for: prohib_passed_screen

Displaying 20 results from an estimated 23 matches for "prohib_passed_screen".

2008 May 14
2
Setting CallerID UNKNOWN on an outgoing call
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 --
2004 Dec 28
2
caller-id blocking
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
2005 Mar 10
2
hide callerid via presention bits of ISDN
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
2005 Mar 24
1
Missing CallingPres Application
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
2006 Mar 22
1
How to hide CallerID - SetCallerPres(prohib) not working
Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still displayed. Is there something else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for
2009 Jul 27
1
INVITE Privacy Information
Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at
2009 Jul 22
3
CallerPres SIP headers Analog Phone
hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?
2010 Mar 26
1
SIP/2.0 403 Forbidden
...ID(all)=s2") in new stack > -- Executing [s at macro-outbound-callerid:13] > ExecIf("SIP/75002-b7705298", "0|Set|CALLERID(all)=") in new stack > -- Executing [s at macro-outbound-callerid:14] > ExecIf("SIP/75002-b7705298", "0|SetCallerPres|prohib_passed_screen") in new > stack > -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298", > "0|AGI|fixlocalprefix") in new stack > -- Executing [s at macro-dialout-trunk:13] Set("SIP/75002-b7705298", > "OUTNUM=015921256331") in ne...
2010 Mar 26
1
send a call from A to B use sip trunk prablem
...new > stack > -- Executing [s at macro-outbound-callerid:13] > ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack > -- Executing [s at macro-outbound-callerid:14] > ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen") > in new stack > -- Executing [s at macro-dialout-trunk:12] > ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack > -- Executing [s at macro-dialout-trunk:13] > Set("SIP/192.168.0.151-088e7938", "OUTNUM=159212563...
2010 Nov 03
1
Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other
2013 Feb 16
1
Dial failed due to trunk reporting BUSY - giving up
Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Aug 29
3
How to use * and # as part of number indialcommand
...owed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen : Presentation Prohibited, Passed Screen prohib_failed_screen : Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Have a look at this doc for more info on keypad protocol h...
2010 May 05
0
T38 trunk configuration for relay appears to affect default trunks for voip
...2649] pbx.c: -- Executing [s at macro-outbound-callerid:14] ExecIf("SIP/21-00000058", "0?Set(CALLERID(all)=)") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-outbound-callerid:15] ExecIf("SIP/21-00000058", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/21-00000058", "0?AGI(fixlocalprefix)") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:13] Set("SIP/21-000...
2015 Mar 20
3
outbound calls
...?Set(CALLERID(all)=0176xxxxxx)") in new stack -- Executing [s at macro-outbound-callerid:16] ExecIf("SIP/101-00000103", "0?Set(CALLERID(all)=)") in new stack -- Executing [s at macro-outbound-callerid:17] ExecIf("SIP/101-00000103", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s at macro-outbound-callerid:18] Set("SIP/101-00000103", "CDR(outbound_cnum)=0176215694") in new stack -- Executing [s at macro-outbound-callerid:19] Set("SIP/101-00000103", "CDR(outbound_cnam)=") in new stack --...
2010 Jun 21
1
How to tell if a dropped call is my fault
...DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Last app: Set|CALLERID(all)=<5053251685> [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-outbound-callerid:14] ExecIf("SIP/611-b7b9ae38", "0|SetCallerPres|prohib_passed_screen") in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Macro [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/611-b7b9ae38", "1...
2011 Sep 28
2
PSTN connectivity
Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before
2015 Mar 27
0
call between snom 300 and aastra 6731i
..."1?Set(CALLERID(all)=300)") in new stack -- Executing [s at macro-outbound-callerid:16] ExecIf("SIP/300-00000192", "0?Set(CALLERID(all)=)") in new stack -- Executing [s at macro-outbound-callerid:17] ExecIf("SIP/300-00000192", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s at macro-outbound-callerid:18] Set("SIP/300-00000192", "CDR(outbound_cnum)=300") in new stack -- Executing [s at macro-outbound-callerid:19] Set("SIP/300-00000192", "CDR(outbound_cnam)=") in new stack -- Execut...
2015 Mar 20
0
outbound calls
...176xxxxxx)") in new stack > -- Executing [s at macro-outbound-callerid:16] > ExecIf("SIP/101-00000103", "0?Set(CALLERID(all)=)") in new stack > -- Executing [s at macro-outbound-callerid:17] > ExecIf("SIP/101-00000103", "0?Set(CALLERPRES()=prohib_passed_screen)") in > new stack > -- Executing [s at macro-outbound-callerid:18] Set("SIP/101-00000103", > "CDR(outbound_cnum)=0176215694") in new stack > -- Executing [s at macro-outbound-callerid:19] Set("SIP/101-00000103", > "CDR(outbound_cnam)=&q...
2015 Mar 27
2
call between snom 300 and aastra 6731i
You would need to give more information really. Your sip.conf file listing the entries for the phones especially 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 >
2015 Mar 20
0
outbound calls
...gt;>> -- Executing [s at macro-outbound-callerid:16] >>> ExecIf("SIP/101-00000103", "0?Set(CALLERID(all)=)") in new stack >>> -- Executing [s at macro-outbound-callerid:17] >>> ExecIf("SIP/101-00000103", "0?Set(CALLERPRES()=prohib_passed_screen)") in >>> new stack >>> -- Executing [s at macro-outbound-callerid:18] Set("SIP/101-00000103", >>> "CDR(outbound_cnum)=0176215694") in new stack >>> -- Executing [s at macro-outbound-callerid:19] Set("SIP/101-00000103",...