[asterisk-users] anyone makes get_swagger_ui work?
hello: I install get_swagger_ui by command with Asterisk-15, but can not access the GUI locally. anyone knows more doc for that? thanks! free...@qq.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header on forwarded call
Le 27/11/2018 à 13:18, Joshua C. Colp a écrit : On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote: Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] [TOOTAiAudio] ; ; Call our gateway exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) same = n,Return exten = h,1,NoOp() same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) same = n,NoOp(Dial status : ${DIALSTATUS}) same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)}) same = n,Return [...] Why can't be PJSIP extra headers setted in this case ? As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be executed on the PJSIP channel itself, not the calling channel. You need to use a pre-dial handler and invoke it there. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER Thanks Joshua, that worked. As you see above I want to have the value of headers when call is ended. Problem is that on h extension the channel already gone. Is there a solution to archieve this ? Is there a reason you can't use a normal dialplan variable instead? That's what I do at this time. I thought I could bypass this by retriving the output of headers Otherwise I don't believe PJSIP_HEADER will retrieve such information regardless, it's for querying headers on an incoming INVITE. Ok, thanks for your help. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header on forwarded call
On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote: > Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] > >> > >> [TOOTAiAudio] > >> ; > >> ; Call our gateway > >> > >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) > >> same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) > >> same = n,Return > >> > >> exten = h,1,NoOp() > >> same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) > >> same = n,NoOp(Dial status : ${DIALSTATUS}) > >> same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)}) > >> same = n,Return > [...] > >> > >> Why can't be PJSIP extra headers setted in this case ? > > > > As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be > > executed on the PJSIP channel itself, not the calling channel. You need to > > use a pre-dial handler and invoke it there. > > > > [1] > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER > > > > Thanks Joshua, that worked. As you see above I want to have the value of > headers when call is ended. Problem is that on h extension the channel > already gone. > > Is there a solution to archieve this ? Is there a reason you can't use a normal dialplan variable instead? Otherwise I don't believe PJSIP_HEADER will retrieve such information regardless, it's for querying headers on an incoming INVITE. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header on forwarded call
Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] [TOOTAiAudio] ; ; Call our gateway exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) same = n,Return exten = h,1,NoOp() same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) same = n,NoOp(Dial status : ${DIALSTATUS}) same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)}) same = n,Return [...] Why can't be PJSIP extra headers setted in this case ? As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be executed on the PJSIP channel itself, not the calling channel. You need to use a pre-dial handler and invoke it there. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER Thanks Joshua, that worked. As you see above I want to have the value of headers when call is ended. Problem is that on h extension the channel already gone. Is there a solution to archieve this ? -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PJSIP enforce Transport
On Tue, Nov 27, 2018, at 3:15 AM, Benjamin Marty wrote: > Hello, > > I have an Asterisk 15.6.0 installation with PJSIP SIP Driver and Sorcery > for Realtime. My Goal is to enforce endpoints to UDP, TCP or TLS. For that > I set the 'transport' column in the endpoint to the corresponding transport > in pjsip.conf. But if I e.g. set the transport to my 'transport-tls-nat' > transport I can still register and place calls via UDP and TCP. Is there > any solution for that? There is no ability to really enforce a remote endpoint to use a specific transport type like was done in chan_sip. You can set an explicit one to use, but in some cases this may not get used (SIP responses won't use it). -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header on forwarded call
On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote: > Hi list, > > to manage an external queue agent the only solution I found is to > connect a local account and redirect calls to this account using forward > features from the phone (SNOM). The problem I face is that before > calling the agent I would like to set extra header. Dialplan to call > external agent is this one with (Gosub): > > [TOOTAiAudio] > ; > ; Call our gateway > > exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) > same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) > same = n,Return > > exten = h,1,NoOp() > same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) > same = n,NoOp(Dial status : ${DIALSTATUS}) > same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)}) > same = n,Return > > When a local phone call extension 115 (the one where calls to external > agent are forwarded), everything is working well. But if I call the > account from a queue I get > > > [Nov 27 09:54:08] ERROR[12758][C-005f]: res_pjsip_header_funcs.c:513 > func_write_header: This function requires a PJSIP channel > > Output of queue is > > deblix9*CLI> queue show q301 > q301 has 0 calls (max unlimited) in 'ringall' strategy (6s holdtime, 47s > talktime), W:0, C:25, A:3, SL:100.0%, SL2:100.0% within 60s > Members: > PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has > taken 8 calls (last was 706 secs ago) > No Callers > > where PPermis115 is the local account on a phone who forward calls to > extension 115. > > Why can't be PJSIP extra headers setted in this case ? As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be executed on the PJSIP channel itself, not the calling channel. You need to use a pre-dial handler and invoke it there. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFC about SIP 'To' header after call diversion?
Hi List I'm struggling to find the correct RFC which "exactly" defines how a SIP Invite has to look like after a call has been diverted. Especially what the content of the To: header field has to be. Example call flow: Alice calls Bob who diverts to Carol. Alice => Bob Invite: b...@example.com From: al...@example.com To: b...@example.com Bob => Carol Invite: ca...@example.com From: al...@example.com To: b...@example.com <= is this correct, or should that be carol? Diversion: b...@example.com Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP add header on forwarded call
Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) same = n,Return exten = h,1,NoOp() same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) same = n,NoOp(Dial status : ${DIALSTATUS}) same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)}) same = n,Return When a local phone call extension 115 (the one where calls to external agent are forwarded), everything is working well. But if I call the account from a queue I get [Nov 27 09:54:08] ERROR[12758][C-005f]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel Output of queue is deblix9*CLI> queue show q301 q301 has 0 calls (max unlimited) in 'ringall' strategy (6s holdtime, 47s talktime), W:0, C:25, A:3, SL:100.0%, SL2:100.0% within 60s Members: PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has taken 8 calls (last was 706 secs ago) No Callers where PPermis115 is the local account on a phone who forward calls to extension 115. Why can't be PJSIP extra headers setted in this case ? Thanks for any hint Reagrds -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users