I have a clean install of Issabel , asterisk 13, with 3 Motif Trunks (Google Voice)
When I make a call and while talking I was hearing the call drop
On further investigation the other party was hearing hold music, when the call dropped on my end.
What I come to find is that regardless of the ATA I use, and the DTMF mode If I speak very loudly or blow into the microphone the call drops on my end and is temporarily sent to music on the far end.
I should clarify I am using a custom context that bypasses most all normal contexts and therefore the transfer/hold may not work , but asterisk should not be detecting an even to initiate a transfer or hold.
Acabo de hacer una instalacion limpio de Isabel, asterisk 13, con 3 troncales Motif (Google Voice)
Cuando realizo una llamada y mientras de hablar , las llamadas se tumban.
Despues de investigar, encontre que al orto ladeo de la llamada escuchaban musica en espera, cuando se cayo en mi lado.
que encontre es sin iportar el ATA o modo de dtmf si hablo recio o soplo aire al microfono la llamada se tumba de mi lado y se escucha musica en el otro lado.
Debo decir que estoy usando mi propio contexto y creo que por eso se caiga pero no debe iniciar una transferencia/espera en cualquier caso . Obvio que asterisk detecta algo para iniciar musica en espera
Reliably Transmitting (NAT) to 199.222.111.99:5078:
OPTIONS sip:12345678900@199.222.111.99:5078;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 198.15.100.60:5060;branch=z9hG4bK4be459cc;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@198.15.100.60>;tag=as76b69ad3
To: <sip:12345678900@199.222.111.99:5078;user=phone;transport=udp>
Contact: <sip:Unknown@198.15.100.60:5060>
Call-ID: 1c480b5e55f8bcc25355d8ec583d818f@198.15.100.60:5060
CSeq: 102 OPTIONS
User-Agent: GoFrackYaSelf
Date: Wed, 04 Jul 2018 22:37:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:199.222.111.99:5078 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.15.100.60:5060;branch=z9hG4bK4be459cc;rport
From: "Unknown" <sip:Unknown@198.15.100.60>;tag=as76b69ad3
To: <sip:12345678900@199.222.111.99:5078;user=phone;transport=udp>;tag=453686759
Call-ID: 1c480b5e55f8bcc25355d8ec583d818f@198.15.100.60:5060
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 258
Content-Type: application/sdp
v=0
o=12345678900 5553588 5553588 IN IP4 192.168.1.21
s=ATA186 Call
c=IN IP4 192.168.1.21
t=0 0
m=audio 16384 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 11 lines) ---
Really destroying SIP dialog '1c480b5e55f8bcc25355d8ec583d818f@198.15.100.60:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 199.222.111.99:5078:
OPTIONS sip:19999999999@199.222.111.99:5078;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 198.15.100.60:5060;branch=z9hG4bK1f1eddb9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@198.15.100.60>;tag=as07ea5800
To: <sip:19999999999@199.222.111.99:5078;user=phone;transport=udp>
Contact: <sip:Unknown@198.15.100.60:5060>
Call-ID: 014133a635bf53a2286226f12c50df68@198.15.100.60:5060
CSeq: 102 OPTIONS
User-Agent: GoFrackYaSelf
Date: Wed, 04 Jul 2018 22:37:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:199.222.111.99:5078 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.15.100.60:5060;branch=z9hG4bK1f1eddb9;rport
From: "Unknown" <sip:Unknown@198.15.100.60>;tag=as07ea5800
To: <sip:19999999999@199.222.111.99:5078;user=phone;transport=udp>;tag=385127894
Call-ID: 014133a635bf53a2286226f12c50df68@198.15.100.60:5060
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 254
Content-Type: application/sdp
v=0
o=19999999999 5553672 5553672 IN IP4 192.168.1.21
s=ATA186 Call
c=IN IP4 192.168.1.21
t=0 0
m=audio 0 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 11 lines) ---
Really destroying SIP dialog '014133a635bf53a2286226f12c50df68@198.15.100.60:5060' Method: OPTIONS
<--- SIP read from UDP:199.222.111.99:5078 --->
INVITE sip
calledNumber)@198.15.100.60 SIP/2.0
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bKf0bed67b6befa4
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 3 INVITE
Contact: 12345678900 <sip:12345678900@199.222.111.99:5078;user=phone;transport=udp>
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest username="12345678900",realm="asterisk",nonce="77301c66",uri="sipcalledNumber)@My.frggindomain.com",response="00db2684470ebcc63bbb463d4e621519"
Expires: 10
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: 100rel,replaces
Content-Length: 222
Content-Type: application/sdp
v=0
o=12345678900 5554337 5554337 IN IP4 199.222.111.99
s=ATA186 Call
c=IN IP4 199.222.111.99
t=0 0
m=audio 16386 RTP/AVP 0 101
a=sendonly
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 10 lines) ---
Sending to 199.222.111.99:5078 (NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7fba64013910 -- Strict RTP learning after remote address set to: 199.222.111.99:16386
Peer audio RTP is at port 199.222.111.99:16386
<--- Transmitting (NAT) to 199.222.111.99:5078 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bKf0bed67b6befa4;received=199.222.111.99;rport=5078
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 3 INVITE
Server: GoFrackYaSelf
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sipcalledNumber)@198.15.100.60:5060>
Content-Length: 0
<------------>
Audio is at 13320
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 199.222.111.99:5078 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bKf0bed67b6befa4;received=199.222.111.99;rport=5078
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 3 INVITE
Server: GoFrackYaSelf
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sipcalledNumber)@198.15.100.60:5060>
Content-Type: application/sdp
Content-Length: 252
v=0
o=root 502413054 502413055 IN IP4 198.15.100.60
s=Asterisk PBX 13.18.5
c=IN IP4 198.15.100.60
t=0 0
m=audio 13320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=recvonly
<------------>
-- Started music on hold, class 'default', on channel 'Motif/(calledNumber)@voice.google.com-6794'
Retransmitting #1 (NAT) to 199.222.111.99:5078:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bKf0bed67b6befa4;received=199.222.111.99;rport=5078
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 3 INVITE
Server: GoFrackYaSelf
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sipcalledNumber)@198.15.100.60:5060>
Content-Type: application/sdp
Content-Length: 252
v=0
o=root 502413054 502413055 IN IP4 198.15.100.60
s=Asterisk PBX 13.18.5
c=IN IP4 198.15.100.60
t=0 0
m=audio 13320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=recvonly
<--- SIP read from UDP:199.222.111.99:5078 --->
ACK sipcalledNumber)@198.15.100.60 SIP/2.0
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bKa83e6135d82f763
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 3 ACK
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest username="12345678900",realm="asterisk",nonce="77301c66",uri="sipcalledNumber)@My.frggindomain.com",response="00db2684470ebcc63bbb463d4e621519"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:199.222.111.99:5078 --->
BYE sipcalledNumber)@198.15.100.60 SIP/2.0
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bK27b5643ebf305588
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 4 BYE
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest username="12345678900",realm="asterisk",nonce="77301c66",uri="sipcalledNumber)@My.frggindomain.com",response="6afa45e41cb00ddead5415c48155a3dd"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 199.222.111.99:5078 (NAT)
Scheduling destruction of SIP dialog '3094271668@192.168.1.21' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 199.222.111.99:5078 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 199.222.111.99:5078;branch=z9hG4bK27b5643ebf305588;received=199.222.111.99;rport=5078
From: 12345678900 <sip:12345678900@My.frggindomain.com:5078;user=phone>;tag=453686759
To: <sipcalledNumber)@My.frggindomain.com;user=phone>;tag=as4a178bcb
Call-ID: 3094271668@192.168.1.21
CSeq: 4 BYE
Server: GoFrackYaSelf
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/12345678900-00000002 left 'simple_bridge' basic-bridge <b2e12a42-df88-4d97-bed2-20d533d813ba>
== Spawn extension (from-gv-extension, (calledNumber), 1) exited non-zero on 'SIP/12345678900-00000002'
-- Channel Motif/(calledNumber)@voice.google.com-6794 left 'simple_bridge' basic-bridge <b2e12a42-df88-4d97-bed2-20d533d813ba>
-- Stopped music on hold on Motif/(calledNumber)@voice.google.com-6794
