search for: tootaiaudio

Displaying 9 results from an estimated 9 matches for "tootaiaudio".

2018 Nov 27
2
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: ${HANGUPCAUSE})...
2018 Nov 27
2
PJSIP add header on forwarded call
...queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${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...
2015 Feb 17
2
Res_fax - FAXOPT(faxdetect)
...n new stack -- Executing [s at macro-Fax:1] Dial("SIP/TOOTAi-00008262", "IAX2/300,,") in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-00008262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-00001d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-00001d5b]: chan_sip.c:9868 process_sdp: Insufficient information fo...
2015 Feb 18
2
Res_fax - FAXOPT(faxdetect)
...[s at macro-Fax:1] Dial("SIP/TOOTAi-00008262", > "IAX2/300,,") in new stack > -- Called IAX2/300 > -- Call accepted by 127.0.0.1 (format alaw) > -- Format for call is (alaw) > -- IAX2/300-7211 is ringing > -- IAX2/300-7211 answered SIP/TOOTAiAudio-00008262 > == Using UDPTL TOS bits 184 > == Using UDPTL CoS mark 5 > [2015-02-17 16:52:51] NOTICE[3467][C-00001d5b]: chan_sip.c:10645 > process_sdp: T.38 re-INVITE detected but no fax extension > [2015-02-17 16:52:56] WARNING[3467][C-00001d5b]: chan_sip.c:9868 > process_sdp:...
2020 Jan 16
1
Asterisk16 - PJSIP - Error 401 on outbound registration
...at sip.myprovider.net> Call-ID: e906c156-a23f-4cff-b099-43c61a4447c5 CSeq: 47982 REGISTER Contact: <sip:123456 at zzz.xyz.174.138:5060> Expires: 3600 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Max-Forwards: 70 User-Agent: TOOTAiAudio Content-Length:  0 Please notice the E..V.T at .?...X....2N$.r...B.. in front of REGISTER, could this create the problem ? > As for what you can control, first, you might try reducing the > expiration from 3600 to 999, or maybe something in the 30-60 range is > better for you. If that...
2015 Feb 18
0
Res_fax - FAXOPT(faxdetect)
...[s at macro-Fax:1] Dial("SIP/TOOTAi-00008262", > "IAX2/300,,") in new stack > -- Called IAX2/300 > -- Call accepted by 127.0.0.1 (format alaw) > -- Format for call is (alaw) > -- IAX2/300-7211 is ringing > -- IAX2/300-7211 answered SIP/TOOTAiAudio-00008262 > == Using UDPTL TOS bits 184 > == Using UDPTL CoS mark 5 > [2015-02-17 16:52:51] NOTICE[3467][C-00001d5b]: chan_sip.c:10645 > process_sdp: T.38 re-INVITE detected but no fax extension > [2015-02-17 16:52:56] WARNING[3467][C-00001d5b]: chan_sip.c:9868 > process_sdp:...
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read...
2020 Jan 15
4
Asterisk16 - PJSIP - Error 401 on outbound registration
Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the
2018 Jul 27
3
SHELL() function Asterisk 13 - can only accept one paramter in string?
Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script: