488: not acceptable here

[beatific-angel]

current I am trying to call with your phone solution. I created two account and two registered. yet not solved this problem (one contact register in other computers).

anyway please see this problem too. both account registered. so I trying to call.

but I can see this error 488 error: not acceptable here

this is sip track log

<--- Received SIP request (3113 bytes) from WSS:204.44.96.131:17566 ---> INVITE sip:102030@rus.cz SIP/2.0 Via: SIP/2.0/WSS 192.0.2.154;branch=z9hG4bK7394738 To: <sip:102030@rus.cz> From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a CSeq: 1 INVITE Call-ID: g6b3qfbnspnfutm0ourb Max-Forwards: 70 Contact: sip:v2n48i5t@192.0.2.154;transport=wss;ob Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0) Content-Type: application/sdp Content-Length: 2613

v=0 o=- 2667082863342222819 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS 820f49f4-7e18-4570-a9bd-eecfe385ebca m=audio 45439 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126 c=IN IP4 204.44.96.131 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:4078256983 1 udp 2122262783 2408:8234:9c13:3d0:e4bf:40c2:8c9c:3d53 53982 typ host generation 0 network-id 3 network-cost 10 a=candidate:3868393361 1 udp 2122194687 192.168.1.157 45439 typ host generation 0 network-id 2 network-cost 10 a=candidate:2762220737 1 udp 2122129151 198.18.58.215 41690 typ host generation 0 network-id 1 network-cost 50 a=candidate:3180773287 1 tcp 1518283007 2408:8234:9c13:3d0:e4bf:40c2:8c9c:3d53 9 typ host tcptype active generation 0 network-id 3 network-cost 10 a=candidate:2819687265 1 tcp 1518214911 192.168.1.157 9 typ host tcptype active generation 0 network-id 2 network-cost 10 a=candidate:3928479793 1 tcp 1518149375 198.18.58.215 9 typ host tcptype active generation 0 network-id 1 network-cost 50 a=candidate:1742403877 1 udp 1685987071 204.44.96.131 45439 typ srflx raddr 192.168.1.157 rport 45439 generation 0 network-id 2 network-cost 10 a=candidate:636232309 1 udp 1685921535 204.44.96.131 41690 typ srflx raddr 198.18.58.215 rport 41690 generation 0 network-id 1 network-cost 50 a=ice-ufrag:gzyw a=ice-pwd:f2SfVyTIG7bjEy0e51xgi0yo a=ice-options:trickle a=fingerprint:sha-256 33:B3:E0:AC:A6:53:61:4E:16:41:9C:6C:CD:87:26:4D:D7:83:A7:0B:84:B0:66:B1:6B:AB:08:1E:2E:DF:7B:89 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:820f49f4-7e18-4570-a9bd-eecfe385ebca a50fe8a8-5be9-40d5-8195-73738d491b22 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2126563126 cname:g6J6Y49/5yPjCW3i a=ssrc:2126563126 msid:820f49f4-7e18-4570-a9bd-eecfe385ebca a50fe8a8-5be9-40d5-8195-73738d491b22 a=ssrc:2126563126 mslabel:820f49f4-7e18-4570-a9bd-eecfe385ebca a=ssrc:2126563126 label:a50fe8a8-5be9-40d5-8195-73738d491b22

<--- Transmitting SIP response (476 bytes) to WSS:204.44.96.131:17566 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS 192.0.2.154;rport=17566;received=204.44.96.131;branch=z9hG4bK7394738 Call-ID: g6b3qfbnspnfutm0ourb From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a To: sip:102030@rus**.cz;tag=z9hG4bK7394738 CSeq: 1 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1658371213/1f35983509e52b3b65de72297d43af73",opaque="50f921b24d528899",algorithm=md5,qop="auth" Server: FPBX-16.0.21.3(16.25.0) Content-Length: 0

<--- Received SIP request (290 bytes) from WSS:204.44.96.131:17566 ---> ACK sip:102030@rus.cz SIP/2.0 Via: SIP/2.0/WSS 192.0.2.154;branch=z9hG4bK7394738 To: <sip:102030@rus.cz>;tag=z9hG4bK7394738 From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a Call-ID: g6b3qfbnspnfutm0ourb CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0

<--- Received SIP request (3393 bytes) from WSS:204.44.96.131:17566 ---> INVITE sip:102030@rus.cz SIP/2.0 Via: SIP/2.0/WSS 192.0.2.154;branch=z9hG4bK6711416 To: <sip:102030@rus.cz> From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a CSeq: 2 INVITE Call-ID: g6b3qfbnspnfutm0ourb Max-Forwards: 70 Authorization: Digest algorithm=MD5, username="2022207", realm="asterisk", nonce="1658371213/1f35983509e52b3b65de72297d43af73", uri="sip:102030@rus**.cz", response="d82d05398520c81e2157ac09c99429e5", opaque="50f921b24d528899", qop=auth, cnonce="4662ikper2a2", nc=00000001 Contact: sip:v2n48i5t@192.0.2.154;transport=wss;ob Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0) Content-Type: application/sdp Content-Length: 2613

v=0 o=- 2667082863342222819 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS 820f49f4-7e18-4570-a9bd-eecfe385ebca m=audio 45439 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126 c=IN IP4 204.44.96.131 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:4078256983 1 udp 2122262783 2408:8234:9c13:3d0:e4bf:40c2:8c9c:3d53 53982 typ host generation 0 network-id 3 network-cost 10 a=candidate:3868393361 1 udp 2122194687 192.168.1.157 45439 typ host generation 0 network-id 2 network-cost 10 a=candidate:2762220737 1 udp 2122129151 198.18.58.215 41690 typ host generation 0 network-id 1 network-cost 50 a=candidate:3180773287 1 tcp 1518283007 2408:8234:9c13:3d0:e4bf:40c2:8c9c:3d53 9 typ host tcptype active generation 0 network-id 3 network-cost 10 a=candidate:2819687265 1 tcp 1518214911 192.168.1.157 9 typ host tcptype active generation 0 network-id 2 network-cost 10 a=candidate:3928479793 1 tcp 1518149375 198.18.58.215 9 typ host tcptype active generation 0 network-id 1 network-cost 50 a=candidate:1742403877 1 udp 1685987071 204.44.96.131 45439 typ srflx raddr 192.168.1.157 rport 45439 generation 0 network-id 2 network-cost 10 a=candidate:636232309 1 udp 1685921535 204.44.96.131 41690 typ srflx raddr 198.18.58.215 rport 41690 generation 0 network-id 1 network-cost 50 a=ice-ufrag:gzyw a=ice-pwd:f2SfVyTIG7bjEy0e51xgi0yo a=ice-options:trickle a=fingerprint:sha-256 33:B3:E0:AC:A6:53:61:4E:16:41:9C:6C:CD:87:26:4D:D7:83:A7:0B:84:B0:66:B1:6B:AB:08:1E:2E:DF:7B:89 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:820f49f4-7e18-4570-a9bd-eecfe385ebca a50fe8a8-5be9-40d5-8195-73738d491b22 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2126563126 cname:g6J6Y49/5yPjCW3i a=ssrc:2126563126 msid:820f49f4-7e18-4570-a9bd-eecfe385ebca a50fe8a8-5be9-40d5-8195-73738d491b22 a=ssrc:2126563126 mslabel:820f49f4-7e18-4570-a9bd-eecfe385ebca a=ssrc:2126563126 label:a50fe8a8-5be9-40d5-8195-73738d491b22

<--- Transmitting SIP response (305 bytes) to WSS:204.44.96.131:17566 ---> SIP/2.0 100 Trying Via: SIP/2.0/WSS 192.0.2.154;rport=17566;received=204.44.96.131;branch=z9hG4bK6711416 Call-ID: g6b3qfbnspnfutm0ourb From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a To: sip:102030@rus**.cz CSeq: 2 INVITE Server: FPBX-16.0.21.3(16.25.0) Content-Length: 0

[2022-07-21 02:40:14] ERROR[13295]: res_pjsip_session.c:937 handle_incoming_sdp: 2022207: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) <--- Transmitting SIP response (359 bytes) to WSS:204.44.96.131:17566 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WSS 192.0.2.154;rport=17566;received=204.44.96.131;branch=z9hG4bK6711416 Call-ID: g6b3qfbnspnfutm0ourb From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a To: sip:102030@rus**.cz;tag=52756495-fe6e-460a-9ff5-1b36a702a31c CSeq: 2 INVITE Server: FPBX-16.0.21.3(16.25.0) Content-Length: 0

<--- Received SIP request (312 bytes) from WSS:204.44.96.131:17566 ---> ACK sip:102030@rus.cz SIP/2.0 Via: SIP/2.0/WSS 192.0.2.154;branch=z9hG4bK6711416 To: <sip:102030@rus.cz>;tag=52756495-fe6e-460a-9ff5-1b36a702a31c From: "2022207" sip:2022207@rus**.cz;tag=3gfrdlvq0a Call-ID: g6b3qfbnspnfutm0ourb CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

<--- Transmitting SIP request (492 bytes) to WSS:204.44.96.131:17566 ---> OPTIONS sip:v2n48i5t@204.44.96.131:17566;transport=ws SIP/2.0 Via: SIP/2.0/WSS 31.42.177.246:8089;rport;branch=z9hG4bKPj9d444316-e774-4389-b6a9-5f8e7484b300;alias From: sip:2022207@freepbx.sangoma.local;tag=9d12c91b-6517-40c2-8d28-951ec7a603f7 To: sip:v2n48i5t@204.44.96.131 Contact: sip:2022207@freepbx.sangoma.local:5060;transport=ws Call-ID: 92dc9595-f8a6-45d1-9d86-e8248f07c12f CSeq: 50664 OPTIONS Max-Forwards: 70 User-Agent: FPBX-16.0.21.3(16.25.0) Content-Length: 0

<--- Received SIP response (548 bytes) from WSS:204.44.96.131:17566 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS 31.42.177.246:8089;rport;branch=z9hG4bKPj9d444316-e774-4389-b6a9-5f8e7484b300;alias From: sip:2022207@freepbx.sangoma.local;tag=9d12c91b-6517-40c2-8d28-951ec7a603f7 To: sip:v2n48i5t@204.44.96.131;tag=3t7bjv3li4 CSeq: 50664 OPTIONS Call-ID: 92dc9595-f8a6-45d1-9d86-e8248f07c12f Supported: outbound User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0) Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Accept: application/sdp,application/dtmf-relay Content-Length: 0

[beatific-angel]

It depends how its used, if you are calling someone but that buddy has not been created yet, then you can populate the caller id or display name while the buddy is being created.

current I am trying the caller Id i mentioned in past. if I access cell phone, it show outbound caller id of trunk. I think my think is possible. so if I send caller ID when try call to cell phone, for example:

if I set caller id in this vaiable, and then if receive the callerid from cell phone, I think the caller id can show. I think by using customized dialplan, it can done. but not sure.

so what i want is I want to know location of receive the caller id of webphone In receiving call section.

[InnovateAsterisk]

The caller id is set in the endpoint config of pjsip, not here.

This is a typical "From" header From: "Conrad De Wet" <sip:yAS43lAg8L@ws-eu-west-1.innovateasterisk.com>;tag=g1n0at248n

What you need to do is set the caller id on the trunk or via some dial-plan action.

[beatific-angel]

Ah, I see, no there isn't such a function, only sending DTMF https://developer.mozilla.org/en-US/docs/Web/API/RTCDTMFSender

to receive the DTMF code in webphone, what I have to do? in the url, I can not find receive dtmf function....

[InnovateAsterisk]

It’s not possible to receive DTMF tones via the api in the peer connection, so it’s also not possible in browser phone.

[beatific-angel]

I dont understand your message. in your code, you can send dtmf code. if can dtmf code, then why can not receive dtmf code?

[InnovateAsterisk]

I can only use API methods and properties of the underlying base code provided by Chrome. There is only a send method. I would assume that the tone is converted into audible audio packets as it’s sent (or on the asterisks side), and that’s why we can hear the tones, but these is no raised event when it arrives in the receiving side (the browser phone side)

[InnovateAsterisk]

More in DTMF here: https://www.rfc-editor.org/rfc/rfc4733.html

[beatific-angel]

in this webphone, we can change voice pitch -- (voice change)?

[beatific-angel]

voice pitch is possible or not? I think that is possible.

[InnovateAsterisk]

There is something called a "AudioContext" https://developer.mozilla.org/en-US/docs/Web/API/AudioContext

part of the larger API - Web Audio API: https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API

This would be something that you would have to undertake on your own, but essentially what you would have to do is take streams (either before you send them, or after you receive them), and capture them with the audio context, process them with whatever effects you like, and then capture the output streams.

If you are unfamiliar with this - there is quite a learning curve, and not much online material. This may get you started: https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API/Using_Web_Audio_API

[beatific-angel]

as my checking, you are using audiocontext.

with this, I can make voice pitch?

[InnovateAsterisk]

Correct, I use it in a few places. This one you have highlighted here is the "sound meter" (top right of UI) - when you are in a call, your voice (and the other persons voice) makes a little green bar dance up and down - visually indicating to the user that audio samples are bing processed.

[InnovateAsterisk]

with this, I can make voice pitch?

No, this has no output

[beatific-angel]

Dear please check this link. https://groups.google.com/g/sip_js/c/-aUrhGtApvU/m/HhQb1qxcCgAJ

in here I can see receiving dtmf part. how is your think?

[InnovateAsterisk]

Their answer is not correct, and doesn’t not answer the problem of “capture the event with my session objet, in the receiving site”, the solution shows how to capture the event on the same client, not on the other side of the call.

[beatific-angel]

I see... and then other question do you have much experience in asterisk dialplan? I have some asterisk problem.

[MrLonely14]

@Beatific-Angel the title of this issue is not related to DTMF is better to move this discussion to the discussion section and close this issue because it has already solved