@RealRuler2112 wrote:

I have two buildings, about 10 minutes apart, that share the same phone system and SIP trunk. One is large with about 70 phones and the other very small with only 2 phones.

Our telco set us up with a SIP trial when I was getting the phones ready to use. There was one DID with this, which was used to register with the SIP server. I realized during installation that since the phones at both locations share the same SIP trunk to the outside world, the addresses coming into 911 would all be the same - that of our main building. Thought of a solution rather fast though - simply re-use the DID from our SIP trial (ending in 1060) and have the phones at the remote site send this as their caller ID. All the rest of the phones at the main site would send our primary phone number ending in 4411. (I’m going to use these 4-digit numbers throughout the rest of this message for brevity & privacy; I have the full 10-digit DID in the system itself.)

Unfortunately, things have not worked as smoothly as planned, which you probably already know given that I’m posting in the forums about it.

If I make test calls to my cell phone from our main site, the caller ID comes through correctly with 4411. If I call my cell from the remote site, it comes through with the 1060 number. Everything is working perfectly! I place a test call to 911 … … … and they get 1060 no matter which site I call from.

The engineer at the SIP provider assures me that they send out whatever caller ID is sent by the PBX to 911 as long as that number is on our account, which both are. I have set 4411 in both the default route and the outbound route for 911, at the extension level as the caller ID, and as the emergency caller ID in the extension. It works for non-emergency calls, but doesn’t matter for emergency calls though - they still only receive 1060 as the caller ID no matter what I do. The telco engineer found the SIP session of my latest test call to 911 and sure enough, it shows 1060 in the FROM field, even though it’s set to be 4411 in all these locations.

I’m really at a loss of what to do next. All I really need is to get every phone in the system to outpulse 4411 to 911 except for 2 that would outpulse 1060, exactly the same as they do for calls to a cell phone. It’s kind of important to get this functional as if there’s an emergency at the remote site, we don’t want them coming to the main site first.

Posts: 1

Participants: 1

Read full topic

@mejor wrote:

Hey Everyone!
I am working on making when a specific extension (200) calls another (100), a 3rd (1000) is auto dialed and added to the bridge between 100 and 200. Once either 100 or 200 hang up I am needing to drop the entire 3 way call. I have tested ChanSpy app but not working either. This 3rd extension is just a paging system to broadcast the call to a room. This is what I have so far.

[custom-auto-bridge]
exten => 100,1,GotoIf($[${CALLERID(num)} != 200]?nobridge)
same => n,Dial(SIP/1000,10)
same => n,GotoIf($[${DIALSTATUS} != ANSWER]?nobridge)
same => n,Bridge(${CHANNELID})
same => n(nobridge),Noop()

Thoughts anyone? I really appreciate the help. I am sure this is simple and it is late and I am not seeing it. hahaha

Posts: 1

Participants: 1

Read full topic

@sumic wrote:

hello everyone,i got some problem with using webrtc.
i use sipML5 client ,i can login the accout ,but can not make a call.
the asterisk cli does not receive call request.
here is the debug info,

asterisk debug info
CLI>== Contact 8888/sip:8888@171.214.204.118:57704;rinstance=a90782ad2973ab75 has been deleted
  == Endpoint 8888 is now Reachable
    -- Contact 8888/sips:8888@171.214.204.118:52314;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 25.856 msec

chrome debug info

SIPml-api.js?svn=252:1 s_websocket_server_url=wss://webphone.qinweigroup.net:8089/ws
SIPml-api.js?svn=252:1 s_sip_outboundproxy_url=(null)
SIPml-api.js?svn=252:1 b_rtcweb_breaker_enabled=no
SIPml-api.js?svn=252:1 b_click2call_enabled=no
SIPml-api.js?svn=252:1 b_early_ims=yes
SIPml-api.js?svn=252:1 b_enable_media_stream_cache=no
SIPml-api.js?svn=252:1 o_bandwidth={}
SIPml-api.js?svn=252:1 o_video_size={}
SIPml-api.js?svn=252:1 SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:webphone.qinweigroup.net>', impi='8888', impu='<sip:8888@webphone.qinweigroup.net:6871>'
SIPml-api.js?svn=252:1 Connecting to 'wss://webphone.qinweigroup.net:8089/ws'
SIPml-api.js?svn=252:1 ==stack event = starting
SIPml-api.js?svn=252:1 __tsip_transport_ws_onopen
SIPml-api.js?svn=252:1 ==stack event = started
SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js?svn=252:1 SEND: REGISTER sip:webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net:6871>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31976 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 ==session event = sent_request
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv
From: <sip:8888@webphone.qinweigroup.net>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net>;tag=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31976 REGISTER
Content-Length: 0
WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1567862686/367e52b8394fbbba3cbe11246d8153c6",opaque="2c95fe7d750ebfcc",stale=FALSE,algorithm=md5
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=252:1 SEND: REGISTER sip:webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net:6871>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31977 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8888",realm="asterisk",nonce="1567862686/367e52b8394fbbba3cbe11246d8153c6",uri="sip:webphone.qinweigroup.net",response="59a2e35a4ef4be5b29383ca478c1eccc",algorithm=md5,cnonce="6d2b1eec4b3d1f0636a0d0359c2b6b07",opaque="2c95fe7d750ebfcc",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = sent_request
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X
From: <sip:8888@webphone.qinweigroup.net>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net>;tag=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X
Contact: <sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no>;expires=199
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31977 REGISTER
Content-Length: 0
Date: 07 Sep 2019 13:24:46 GMT;07
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPj894c39de-a593-4e96-b2d5-8c11e964ab64;alias
From: <sip:8888@ecs-3a46>;tag=bbd36bad-e620-4909-8514-4e53a8d62f87
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:8888@ecs-3a46:5060;transport=ws>
Call-ID: 009b28b4-14ad-41f7-ab4a-b6cf81e58643
CSeq: 12187 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPj894c39de-a593-4e96-b2d5-8c11e964ab64;alias
From: <sip:8888@ecs-3a46>;tag=bbd36bad-e620-4909-8514-4e53a8d62f87
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Call-ID: 009b28b4-14ad-41f7-ab4a-b6cf81e58643
CSeq: 12187 OPTIONS
Content-Length: 0


SIPml-api.js?svn=252:1 ==session event = connected
SIPml-api.js?svn=252:1 State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api.js?svn=252:1 ICE servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.net:3478"},{"url":"stun:numb.viagenie.ca:3478"}]
SIPml-api.js?svn=252:1 ==stack event = m_permission_requested
SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 onGetUserMediaSuccess
SIPml-api.js?svn=252:1 createOffer
SIPml-api.js?svn=252:1 onNegotiationNeeded
SIPml-api.js?svn=252:1 onCreateSdpSuccess
SIPml-api.js?svn=252:1 ==stack event = m_permission_accepted
SIPml-api.js?svn=252:1 onSignalingstateChange:have-local-offer
SIPml-api.js?svn=252:1 onSetLocalDescriptionSuccess
10SIPml-api.js?svn=252:1 onIceCandidate = gathering
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPj84c61c76-2aee-4086-9fd5-6586c5ba2468;alias
From: <sip:6001@ecs-3a46>;tag=eaac7c40-3341-4d78-aa5f-d85564111c69
To: <sips:6001@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:6001@ecs-3a46:5060;transport=ws>
Call-ID: 21c7e9db-18b8-4381-adf8-d06eade6494f
CSeq: 55275 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPj84c61c76-2aee-4086-9fd5-6586c5ba2468;alias
From: <sip:6001@ecs-3a46>;tag=eaac7c40-3341-4d78-aa5f-d85564111c69
To: <sips:6001@171.214.204.118;rtcweb-breaker=no>
Call-ID: 21c7e9db-18b8-4381-adf8-d06eade6494f
CSeq: 55275 OPTIONS
Content-Length: 0


SIPml-api.js?svn=252:1 onIceCandidate = complete
SIPml-api.js?svn=252:1 ICE GATHERING COMPLETED!
SIPml-api.js?svn=252:1 onIceGatheringCompleted
SIPml-api.js?svn=252:1 SEND: INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 INVITE
Content-Type: application/sdp
Content-Length: 2692
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 5248616704460199000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
m=audio 52985 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 171.214.204.118
a=rtcp:52987 IN IP4 171.214.204.118
a=candidate:1302483413 1 udp 2122260223 192.168.2.70 52985 typ host generation 0 network-id 1
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 52986 typ host generation 0 network-id 2
a=candidate:1302483413 2 udp 2122260222 192.168.2.70 52987 typ host generation 0 network-id 1
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 52988 typ host generation 0 network-id 2
a=candidate:52538661 1 tcp 1518280447 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:52538661 2 tcp 1518280446 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:3149015041 1 udp 1686052607 171.214.204.118 52985 typ srflx raddr 192.168.2.70 rport 52985 generation 0 network-id 1
a=candidate:3149015041 2 udp 1686052606 171.214.204.118 52987 typ srflx raddr 192.168.2.70 rport 52987 generation 0 network-id 1
a=ice-ufrag:njEb
a=ice-pwd:sgS07lkCk+L7Mpt/4o7DkNn3
a=ice-options:trickle
a=fingerprint:sha-256 F1:C6:62:D6:00:E7:93:AD:E2:A5:F6:07:6E:04:D7:15:9F:37:E0:D8:39:10:EE:0D:9D:48:66:14:2E:6D:15:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
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:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1023339144 cname:vMOnepzLpY47mIGp
a=ssrc:1023339144 msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=ssrc:1023339144 mslabel:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
a=ssrc:1023339144 label:d2ae0b13-0281-4a11-be28-1bbd8828786e

SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 INVITE
Content-Length: 0
WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1567862736/d140d4f5a784df87f6f657c642098d3c",opaque="6bebb1a730b11b48",stale=FALSE,algorithm=md5
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 SEND: ACK sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 ACK
Content-Length: 0
Max-Forwards: 70


SIPml-api.js?svn=252:1 State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api.js?svn=252:1 SEND: INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Type: application/sdp
Content-Length: 2692
Max-Forwards: 70
Authorization: Digest username="8888",realm="asterisk",nonce="1567862736/d140d4f5a784df87f6f657c642098d3c",uri="sip:*69@webphone.qinweigroup.net",response="20270395d0a5bed5a1757e50eea4377b",algorithm=md5,cnonce="b3e6cb0ee6bb427a6edc186655476f81",opaque="6bebb1a730b11b48",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 5248616704460199000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
m=audio 52985 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 171.214.204.118
a=rtcp:52987 IN IP4 171.214.204.118
a=candidate:1302483413 1 udp 2122260223 192.168.2.70 52985 typ host generation 0 network-id 1
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 52986 typ host generation 0 network-id 2
a=candidate:1302483413 2 udp 2122260222 192.168.2.70 52987 typ host generation 0 network-id 1
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 52988 typ host generation 0 network-id 2
a=candidate:52538661 1 tcp 1518280447 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:52538661 2 tcp 1518280446 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:3149015041 1 udp 1686052607 171.214.204.118 52985 typ srflx raddr 192.168.2.70 rport 52985 generation 0 network-id 1
a=candidate:3149015041 2 udp 1686052606 171.214.204.118 52987 typ srflx raddr 192.168.2.70 rport 52987 generation 0 network-id 1
a=ice-ufrag:njEb
a=ice-pwd:sgS07lkCk+L7Mpt/4o7DkNn3
a=ice-options:trickle
a=fingerprint:sha-256 F1:C6:62:D6:00:E7:93:AD:E2:A5:F6:07:6E:04:D7:15:9F:37:E0:D8:39:10:EE:0D:9D:48:66:14:2E:6D:15:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
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:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1023339144 cname:vMOnepzLpY47mIGp
a=ssrc:1023339144 msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=ssrc:1023339144 mslabel:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
a=ssrc:1023339144 label:d2ae0b13-0281-4a11-be28-1bbd8828786e

SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Length: 0
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=bc984f23-7939-43c2-91fb-edd7f4633fb8
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Length: 0
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 SEND: ACK sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=bc984f23-7939-43c2-91fb-edd7f4633fb8
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 ACK
Content-Length: 0
Max-Forwards: 70


SIPml-api.js?svn=252:1 State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
SIPml-api.js?svn=252:1 === INVITE Dialog terminated ===
SIPml-api.js?svn=252:1 PeerConnection::stop()
2SIPml-api.js?svn=252:1 ==session event = i_ao_request
SIPml-api.js?svn=252:1 ==session event = terminated
SIPml-api.js?svn=252:1 The FSM is in the final state
tsk_utils_log_warn @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_transac.fsm_act @ SIPml-api.js?svn=252:1
(anonymous) @ SIPml-api.js?svn=252:1
setTimeout (async)
tsip_transac_layer.cancel_by_dialog @ SIPml-api.js?svn=252:1
tsip_dialog.deinit @ SIPml-api.js?svn=252:1
__tsip_dialog_invite_onterm @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_dialog.fsm_act @ SIPml-api.js?svn=252:1
__tsip_dialog_invite_event_callback @ SIPml-api.js?svn=252:1
tsip_dialog.callback @ SIPml-api.js?svn=252:1
__tsip_transac_ict_Proceeding_2_Completed_X_300_to_699 @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_transac.fsm_act @ SIPml-api.js?svn=252:1
__tsip_transac_ict_event_callback @ SIPml-api.js?svn=252:1
tsip_transac.callback @ SIPml-api.js?svn=252:1
tsip_transac_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPjab895d3c-9a73-4471-aa40-bdf5ed7205f0;alias
From: <sip:8888@ecs-3a46>;tag=bbb3efa4-0e14-4e3b-9d26-f284108a1094
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:8888@ecs-3a46:5060;transport=ws>
Call-ID: 30a53df1-ff64-416b-bb20-04796e7d840f
CSeq: 24739 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPjab895d3c-9a73-4471-aa40-bdf5ed7205f0;alias
From: <sip:8888@ecs-3a46>;tag=bbb3efa4-0e14-4e3b-9d26-f284108a1094
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Call-ID: 30a53df1-ff64-416b-bb20-04796e7d840f
CSeq: 24739 OPTIONS
Content-Length: 0


```
could someone can help me?
i realy need you help .thx.

Posts: 1

Participants: 1

Read full topic

@windswept321 wrote:

Hi,
I used to have version 5.0.0.21 for the old Mitel dual mode SIP 5224 and 5235 endpoints, which is an important stepping stone for flashing to later revisions.
Does anyone know where I might find a copy of it, please? I have tried the unofficial Mitel forums but they don’t seem to be very active now.
Thank you

Posts: 1

Participants: 1

Read full topic

@assexix wrote:

Guys, I’m in trouble with the IVR Announcement.

Freepbx is not recognizing any ads.

how do i solve this?

Posts: 12

Participants: 4

Read full topic

@aussietony wrote:

Hi, I have a simple freepbx setup which currently has just 2 Chan_sip exensions, 101 and 102
I have setup 2 softphones with these two extensions, the system works great.
I have been trying to setup Call Flow Control, when setup I can call *280 it toggles on and off.
I set the Green/off to go to an Announcement and the to Red/on to extension 102.
I simply ignores the CFC and calls the extension
Am I missing something?

Posts: 2

Participants: 2

Read full topic

@dan_ce wrote:

I’d like to send IAX calls through the same call flow control that my PSTN inbound calls go through. At the moment if I call home from work, if I call via the PSTN then if my wife isn’t in, I’ll get a message telling me so. If I call via IAX, the call starts ringing the extensions without being passed through the call flow control which knows whether or not the house is empty. @billsimon had a great idea for how to modify the custom context to route calls through a time condition, but ideally, I’d like to route IAX calls through a flow toggle (and if poss the SAME flow toggle that I’m already using for another application?)

Ideally I don’t want two call flow controls one for the IAX route and one for the PSTN inbound route - is this avoidable?

Thank you!!

Posts: 5

Participants: 4

Read full topic

@alreilly wrote:

Greetings everyone - I am sure y’all have the same issue I have with the RoboCallers calling in. What I am looking for is that if there is a Blacklist function such as if you want to block an NPA/NXX (ex: 800555XXXX) but allow (ex: 8005551234) as there might a legitimate number that may need to get through. I know of the Blacklist Feature in FreePBX, but you have to enter each number and that could be a very long process. Does anyone have any other ideas or will FreePBX be coming out with a feature like this in the future? Thanks !

Posts: 3

Participants: 3

Read full topic

@sumic wrote:

hi all,
i try to use WEBRTC make a call from chrome use sipml5,
but the CLI show message “certificate verify failed”,how can i troubleshoot?
i have already have the certificate file by domain :webphoen.qinweigroup.net
thank you ~~
best regards!

[2019-09-09 17:40:54] ERROR[28748]: res_pjsip.c:4261 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 6001
<--- Received SIP request (2894 bytes) from WSS:110.184.145.26:5540 --->
INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9;rport
From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>
Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 7204 INVITE
Content-Type: application/sdp
Content-Length: 2286
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 8214160573771452000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80
m=audio 7306 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 110.184.145.26
a=rtcp:7307 IN IP4 110.184.145.26
a=candidate:1002299927 1 udp 2122260223 192.168.6.80 41131 typ host generation 0 network-id 1
a=candidate:1002299927 2 udp 2122260222 192.168.6.80 58302 typ host generation 0 network-id 1
a=candidate:1967005415 1 tcp 1518280447 192.168.6.80 9 typ host tcptype active generation 0 network-id 1
a=candidate:1967005415 2 tcp 1518280446 192.168.6.80 9 typ host tcptype active generation 0 network-id 1
a=candidate:3450706883 1 udp 1686052607 110.184.145.26 7306 typ srflx raddr 192.168.6.80 rport 41131 generation 0 network-id 1
a=candidate:3450706883 2 udp 1686052606 110.184.145.26 7307 typ srflx raddr 192.168.6.80 rport 58302 generation 0 network-id 1
a=ice-ufrag:/D80
a=ice-pwd:5okDbtNyoE6yUkMzLXBiqlkN
a=ice-options:trickle
a=fingerprint:sha-256 A9:32:22:20:31:BE:33:67:6A:15:A4:E7:51:96:CA:56:B6:8B:8B:7C:92:5A:95:1B:FE:26:CB:47:94:AC:57:ED
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c
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:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2150700876 cname:vScPyFZzs73nSNMx
a=ssrc:2150700876 msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c
a=ssrc:2150700876 mslabel:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80
a=ssrc:2150700876 label:d4197ebc-3f1d-4476-bd11-b6b57191d46c

<--- Transmitting SIP response (570 bytes) to WSS:110.184.145.26:5540 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9
CSeq: 7204 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",opaque="7b46d5942c5a6099",algorithm=md5,qop="auth"
Server: FPBX-15.0.16(16.5.0)
Content-Length:  0


<--- Received SIP request (403 bytes) from WSS:110.184.145.26:5540 --->
ACK sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9;rport
From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 7204 ACK
Content-Length: 0
Max-Forwards: 70


<--- Received SIP request (3188 bytes) from WSS:110.184.145.26:5540 --->
INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N;rport
From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>
Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 7205 INVITE
Content-Type: application/sdp
Content-Length: 2286
Max-Forwards: 70
Authorization: Digest username="6001",realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",uri="sip:*69@webphone.qinweigroup.net",response="cb9e1bef0f3f6eb3dd7758a2aa34522c",algorithm=md5,cnonce="5147d96682bd6403b00648ffa4d0e929",opaque="7b46d5942c5a6099",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 8214160573771452000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80
m=audio 7306 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 110.184.145.26
a=rtcp:7307 IN IP4 110.184.145.26
a=candidate:1002299927 1 udp 2122260223 192.168.6.80 41131 typ host generation 0 network-id 1
a=candidate:1002299927 2 udp 2122260222 192.168.6.80 58302 typ host generation 0 network-id 1
a=candidate:1967005415 1 tcp 1518280447 192.168.6.80 9 typ host tcptype active generation 0 network-id 1
a=candidate:1967005415 2 tcp 1518280446 192.168.6.80 9 typ host tcptype active generation 0 network-id 1
a=candidate:3450706883 1 udp 1686052607 110.184.145.26 7306 typ srflx raddr 192.168.6.80 rport 41131 generation 0 network-id 1
a=candidate:3450706883 2 udp 1686052606 110.184.145.26 7307 typ srflx raddr 192.168.6.80 rport 58302 generation 0 network-id 1
a=ice-ufrag:/D80
a=ice-pwd:5okDbtNyoE6yUkMzLXBiqlkN
a=ice-options:trickle
a=fingerprint:sha-256 A9:32:22:20:31:BE:33:67:6A:15:A4:E7:51:96:CA:56:B6:8B:8B:7C:92:5A:95:1B:FE:26:CB:47:94:AC:57:ED
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c
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:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2150700876 cname:vScPyFZzs73nSNMx
a=ssrc:2150700876 msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c
a=ssrc:2150700876 mslabel:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80
a=ssrc:2150700876 label:d4197ebc-3f1d-4476-bd11-b6b57191d46c

  == Setting global variable 'SIPDOMAIN' to 'webphone.qinweigroup.net'
<--- Transmitting SIP response (374 bytes) to WSS:110.184.145.26:5540 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>
CSeq: 7205 INVITE
Server: FPBX-15.0.16(16.5.0)
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [*69@from-internal:1] Goto("PJSIP/6001-0000002d", "app-calltrace-perform,s,1") in new stack
    -- Goto (app-calltrace-perform,s,1)
    -- Executing [s@app-calltrace-perform:1] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [s@app-calltrace-perform:2] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(name,i)=呼叫追踪") in new stack
    -- Executing [s@app-calltrace-perform:3] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(num,i)=s") in new stack
    -- Executing [s@app-calltrace-perform:4] Answer("PJSIP/6001-0000002d", "") in new stack
<--- Transmitting SIP response (1514 bytes) to WSS:110.184.145.26:5540 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84
CSeq: 7205 INVITE
Server: FPBX-15.0.16(16.5.0)
Contact: <sips:192.168.1.158:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "呼叫追踪" <sip:s@webphone.qinweigroup.net>
Content-Type: application/sdp
Content-Length:   791

v=0
o=- 2007666272 4 IN IP4 192.168.1.158
s=Asterisk
c=IN IP4 192.168.1.158
t=0 0
a=group:BUNDLE 0
m=audio 12072 UDP/TLS/RTP/SAVPF 111 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 53:49:22:7D:B4:4E:69:5F:62:F6:18:50:65:84:D0:77:EC:6E:8D:DE:AB:00:A7:76:2D:25:89:A7:77:2D:E6:4F
a=ice-ufrag:23d30b5b2dbde9b75fe38a4b654da6d3
a=ice-pwd:0039026f23c975ca23d5a814280ca8af
a=candidate:He0ec0f98 1 UDP 2130706431 fe80::f816:3eff:fe20:b27f 12072 typ host
a=candidate:Hc0a8019e 1 UDP 2130706431 192.168.1.158 12072 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1560523773 cname:45b90ca4-4336-40eb-a066-08032a3e4f91
a=mid:0

<--- Received SIP request (899 bytes) from WSS:110.184.145.26:5540 --->
ACK sips:192.168.1.158:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKb5No2TLObME2ZeTA6mZu;rport
From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS
To: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84
Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 7205 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",uri="sips:192.168.1.158:8089;transport=ws",response="f29731cd3b7930c85f64e989a7f4ed8c",algorithm=md5,cnonce="5147d96682bd6403b00648ffa4d0e929",opaque="7b46d5942c5a6099",qop=auth,nc=00000002
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom


[2019-09-09 17:41:34] ERROR[30514][C-00000023]: res_rtp_asterisk.c:2970 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f780408e7f0' due to reason 'certificate verify failed', terminating
[2019-09-09 17:41:34] WARNING[30514][C-00000023]: res_rtp_asterisk.c:7108 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
  == Spawn extension (app-calltrace-perform, s, 4) exited non-zero on 'PJSIP/6001-0000002d'
<--- Transmitting SIP request (507 bytes) to WSS:110.184.145.26:5540 --->
BYE sips:6001@110.184.145.26:5540;transport=ws;rtcweb-breaker=no;click2call=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPjd8baabe6-cc14-43de-9a61-d5efd783537e;alias
From: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84
To: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 14366 BYE
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)
Content-Length:  0


<--- Received SIP response (420 bytes) from WSS:110.184.145.26:5540 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPjd8baabe6-cc14-43de-9a61-d5efd783537e;alias
From: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84
To: "jack"<sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS
Contact: <sips:6001@df7jal23ls0d.invalid;transport=wss>
Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46
CSeq: 14366 BYE
Content-Length: 0

Posts: 1

Participants: 1

Read full topic

@duli wrote:

Hello, Forum:

I`d like to develop a special button in my company´s groupware (web based) to enable / disable call forward (*72 / *73) on the user´s extension. I know the UCP has this ability, but I´d like to provide the same functionality from inside the groupware. Any pointers or tips about where I should begin to look?

Thanks a lot!

Posts: 2

Participants: 1

Read full topic

@trychlik wrote:

Hi,

i have problem with queue freezing on asterisk16 + freepbx14. Most often it happens with bigger amount of incoming calls (100+), but sometimes it happens with 5 calls too. Extensions stops ringing, melodies stops playing… Calls are still open and arent cancelled corectly until reboot.
it happens 1-2 per week

its running on proxmox, 32cpu, 48G ram
i tried to change virtual network card, but it doesnt helped

suspicious log:

[2019-08-05 20:02:37] WARNING[17642] chan_sip.c: Autodestruct on dialog '[1c25fe404573620a3865974933377964 @ (ip_deleted)
(1c25fe404573620a3865974933377964 @ (ip_deleted):5060' with owner SIP/XXXXX_ustredna-00006112 in place (Method: BYE). Rescheduling destruction for 10000 ms

any idea?
thanks

Posts: 1

Participants: 1

Read full topic

@avayax wrote:

Fail2ban blocked a local IP address yesterday, which belongs to an ordinary workstation because of failed “attempts against apache-auth”.

Httpd acces logs show this.
Do those logs show that something is trying to steal my http provisioning credentials and has someone else seen this GET /mnt/mtd/AVAST-HNS-SCAN-PROBE HTTP/1.1 before?

10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - webadmin [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - webadmin [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - webadmin [08/Sep/2019:21:40:34 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - webadmin [08/Sep/2019:21:40:34 -0400] "GET / HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /etc/passwd HTTP/1.1" 401 477 "-" "-"
10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /language/Swedish${IFS}&&ping$IFS-c1$IFS-s41${IFS}10.1.10.119>/dev/null&&tar${IFS}/string.js HTTP/1.1" 404 384 "-" "-"
10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /language/Swedish${IFS}&&echo${IFS}AVAST-HNS-SCAN-PROBE>AVAST-HNS-SCAN-PROBE&&tar${IFS}/string.js HTTP/1.1" 404 389 "-" "-"
10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /mnt/mtd/AVAST-HNS-SCAN-PROBE HTTP/1.1" 404 302 "-" "-"
10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /etc/passwd HTTP/1.1" 404 284 "-" "-"

Posts: 1

Participants: 1

Read full topic

@PBXKos wrote:

Hello,

Can anyone help me I have unselected admin access than accidentally submitted.

Now I don’t have access to make changes ? Is there possible to return the access ?

Posts: 2

Participants: 2

Read full topic

@epp289 wrote:

Good morning may i ask you please
i have set a password in the Outbound Route in order the extensions that are being used by the clients (in rooms to let) not to be able to call outside without letting us know. But the problem is that if i want to transfer an incoming call to a custom extension that calls a mobile phone (in the queue or manually) the caller hears “please enter the password followed the # key”. What to do to fix this?

Posts: 3

Participants: 3

Read full topic

@VOIPDummy wrote:

I am trying to pick up the MWI from my VOIP provider (Anveo) and have it appear on my ATA (Grandstream), but a couple of weird things are happening. I put a line in sip_general_custom that says:
mwi => 1234567:password@sip.anveo.com:5010/100
And I changed the mailbox for the SIP extension (301) to:
1234567@SIP_Remote

First odd thing is that the MWI on the extension doesn’t come on. A “show subscriptions” lists the extension and shows the mailbox as 1234567 (but no context). For the VOIP provider, it lists the subscription mailbox as ‘none’.

If anyone is using the mwi=> setting, could you tell me how it works for you?
What does your “show subs” list for the extension and the ‘source’?
Should the VOIP provider mailbox really be ‘none’?

Second, and really odd thing is that my subscriptions keep multiplying. After booting, “show subs” lists TWO mwi subscriptions to the VOIP provider. If I make any change through the GUI and do a Apply Config, I get two MORE mwi subscriptions to the VOIP provider. Each time I do Apply Config I get one or two MORE mwi subscriptions. Right now, I have 8 separate subscriptions to the VIOP service, each with a different ID number. Wireshark confirms that there really are this many subscriptions active.

Can anyone tell me where all these subscriptions are coming from?
And how to stop them from multiplying?

Any help is appreciated. More info on request.
Thanks.
Peter.

Posts: 1

Participants: 1

Read full topic

@davids1 wrote:

Hi,
My original post was here BLF attended transfers with transfer callbacks but it closed with inactivity as I was pulled back into another project at that time.

I have my custom dial plan built and tested by dialling *2 for an attended transfer, however when I move to using a BLF attended transfer, it doesnt use the same context as a dtmf *2 attended transfer. Can anyone tell me how or where I can get the BLF transfer to pick up my custom dial plan?

Hopefully this is something simple, but having looked through lots of other posts I cant see anywhere that has an answer. Some people setup BLFs and DTMF buttons but having to setup and look after two sets of buttons is not somethign we can do as we dont have enough buttons on the Sangoma 500’s, plus i’m sure the client would think it strange we cant do the same attended transfer in the BLF as by pressing *2.

I know we can do extra things in the BLF like short and long presses but I would expect an attended transfer (by whatever means, BLF or *2) to go to the same transfer_context. Can I do something funky in the basefile edit template or any other file to tweak the BLF transfer context?

I am using a new FreePBX install
FreePBX 14.0.13.4
Current Asterisk Version: 13.22.0

My config edit file changes -

globals_custom.conf

TRANSFER_CONTEXT = custom-test_transfer

extensions_custom.conf

[custom-test_transfer]
exten => _X.,1,NoOp(Entering Daves custom-test_transfer)
this is not the full dial plan, this is just to show in the logs below the noop line during a dtmf attended transfer, and that it doesnt show during a BLF attended transfer.

Logs during a DTMF Attended transfer using *2 EXT #

[2019-09-06 08:48:21] VERBOSE[5620][C-0000001e] app_dial.c: PJSIP/4001-00000038 is ringing
[2019-09-06 08:48:22] VERBOSE[5620][C-0000001e] app_dial.c: PJSIP/4001-00000038 answered IAX2/sbc3-2340
[2019-09-06 08:48:22] VERBOSE[5622][C-0000001e] bridge_channel.c: Channel PJSIP/4001-00000038 joined ‘simple_bridge’ basic-bridge <7bdc2e82-aaa2-456d-af5d-fa141463d643>
[2019-09-06 08:48:22] VERBOSE[5620][C-0000001e] bridge_channel.c: Channel IAX2/sbc3-2340 joined ‘simple_bridge’ basic-bridge <7bdc2e82-aaa2-456d-af5d-fa141463d643>
[2019-09-06 08:48:26] VERBOSE[5622][C-0000001e] bridge_basic.c: Channel PJSIP/4001-00000038: Started DTMF attended transfer.
[2019-09-06 08:48:26] VERBOSE[5622][C-0000001e] file.c: <PJSIP/4001-00000038> Playing ‘pbx-transfer.ulaw’ (language ‘en_GB’)
[2019-09-06 08:48:26] VERBOSE[5620][C-0000001e] res_musiconhold.c: Started music on hold, class ‘default’, on channel ‘IAX2/sbc3-2340’
[2019-09-06 08:48:31] VERBOSE[5625][C-0000001e] bridge_channel.c: Channel Local/4000@custom-test_transfer-0000004f;1 joined ‘simple_bridge’ basic-bridge <365c2d13-a28d-4c97-8987-9e82c9f6f199>
[2019-09-06 08:48:31] VERBOSE[5624][C-0000001e] pbx.c: Executing [4000@custom-test_transfer:1] NoOp(“Local/4000@custom-test_transfer-0000004f;2”, "Entering Daves custom-test_transfer") in new stack
[2019-09-06 08:48:31] VERBOSE[5624][C-0000001e] pbx.c: Executing [4000@custom-test_transfer:2] NoOp(“Local/4000@custom-test_transfer-0000004f;2”, “The caller id number before dial is: 4001”) in new stack

Logs during a BLF Attended transfer to the same extension - we dont see it picking up the custom-test_transfer

[2019-09-06 08:51:06] VERBOSE[6110][C-00000021] app_dial.c: PJSIP/4001-0000003e is ringing
[2019-09-06 08:51:08] VERBOSE[6110][C-00000021] app_dial.c: PJSIP/4001-0000003e answered IAX2/sbc3-6413
[2019-09-06 08:51:08] VERBOSE[6112][C-00000021] bridge_channel.c: Channel PJSIP/4001-0000003e joined ‘simple_bridge’ basic-bridge <4b2ab841-5127-458d-92f1-be575877046f>
[2019-09-06 08:51:08] VERBOSE[6110][C-00000021] bridge_channel.c: Channel IAX2/sbc3-6413 joined ‘simple_bridge’ basic-bridge <4b2ab841-5127-458d-92f1-be575877046f>
[2019-09-06 08:51:13] VERBOSE[6110][C-00000021] res_musiconhold.c: Started music on hold, class ‘default’, on channel ‘IAX2/sbc3-6413’
[2019-09-06 08:51:13] VERBOSE[15638] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘servername.domain.co.uk’
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@from-internal:1] GotoIf(“PJSIP/4001-0000003f”, “1?ext-local,4000,1:followme-check,4000,1”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx_builtins.c: Goto (ext-local,4000,1)
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@ext-local:1] Set(“PJSIP/4001-0000003f”, “__RINGTIMER=30”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@ext-local:2] Macro(“PJSIP/4001-0000003f”, “exten-vm,4000,4000,0,0,0”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [s@macro-exten-vm:1] Macro(“PJSIP/4001-0000003f”, “user-callerid,”) in new stack

Thanks for your help,
Regards
David

Posts: 1

Participants: 1

Read full topic

@oops7812 wrote:

Has someone experience or even better a working configuration?

Please note: there are many advices and working configurations for the Deutsche Telekom, but all I found not belongs to this contract type “Cloud PBX”.

I found many examples for “DeutschlandLAN All-IP”, “DeutschlandLAN IP-Start” or “DeutschlandLAN Sip-Trunk” and so on…

And please don’t ask why we choose this contract type. Our old contract was terminated and we have done the wrong decision. Now I try to make the best of it and FreePBX is my choice, but I can’t get registered

Posts: 1

Participants: 1

Read full topic

@yitzflam wrote:

Where is FindMeFollowMe information stored? I found asterisk.findmefollow DB in CLI, which contains correct info. When i tried changing info, all changes seem to have been made, no errors, and updated information is displayed in table, but when I dial or check extension via GUI it writes back changes. also, where is (DB(AMPUSER/${AMPUSER}/followme)?
Thank you

Posts: 2

Participants: 2

Read full topic

@kbyrum wrote:

Hi - I have installed FreePBX 14 / Asterisk 16 on a physical machine. I setup one phone and assigned it an extension. The phone is connected using PJSIP (5060). I enabled voice mail on the extension. I can dial voice mail using *97 but when it answers the login always fails. It seems as if the voice mail is not recognizing any tones. I bypassed the password by setting it blank and when it wont recognize any key press for menu options. I have tried setting the dtmf signaling to RFC 4733, Auto, SIP Info and Inband audio but nothing seems to work. Does any one have any suggestions on how to troubleshoot this? Thanks in advance.

Posts: 1

Participants: 1

Read full topic

@nickb wrote:

I’m running FreePBX 12.7.6-1904-1.sng7 using the commercial EPM, on a system with 73 extensions. Most of these (about 65) are managed by the EPM; a few unsupported devices are manually configured. The system is fully updated in Yum and Module Admin.

Nearly all devices are Cisco SPA500 series. Recently we added six Grandstream GXP-1628 devices, configured via commercial EPM.

These devices successfully provision (verified in apache log) and have the correct information in the provisioning files for SIP server and ports. I am using PJSIP.

All six devices will only show “Unavailable” in SIP Peers. Attempting to dial one of these extensions from another results in the call going directly to voicemail. They can, however, dial voicemail, receive a password prompt from the server, subsequently interacting with the voicemail subsystem.

Suspecting a NAT issue in the router at the client site, we verified SIP ALG was disabled, and changed NAT settings to the recommended values for a SonicWall per the documentation that the site network admin referenced (I do not have that documentation). The UDP NAT timeout is 30 seconds in the router, so I changed Register Expiration to 1 minute, and ReRegister Before Expiration to 40 seconds in the Basefile Editor for the Grandstream template, this should result in re-registration before the 30 second UDP NAT timeout happens.

However, not even the initial registration is received when the phones are booted. I did reboot the FreePBX server as well, but no change in behavior occured after rebooting.

I’ve also verified that the site IP address is in the “Local” networks group in the Firewall, and is whitelisted in Intrusion Detection.

Any suggestions as to why the phones may not be registering at all?

Posts: 7

Participants: 3

Read full topic