asterisk X avaya chamadas do avaya para o asterisk bina anonymous no aparelho

lucasvitor199
(usa Outra)

Enviado em 18/10/2018 - 16:40h

ao realizar chamadas do PABX avaya para o PABX asterisk bina "anonymous" no lugar do ramal, se eu alterar o parâmetro "send name" no avaya bina "asterisk".
preciso saber qual é o parâmetro dentro do asterisk para aparecer o numero do ramal.

vejo no log debug abaixo do asterisk que ele esta recebendo o nome do ramal avaya "teste"

-- Executing [2051@public:1] Dial("OOH323/Avaya-865", "SIP/2051,30") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10208
Adding codec 100010 (ilbc) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding codec 100028 (speex32) to SDP
Reliably Transmitting (no NAT) to 192.168.32.201:5060:
INVITE sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled SIP/2.0
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
Max-Forwards: 70
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>
Contact: <sip:asterisk@192.168.30.161:5060>
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.16.0
Date: Thu, 18 Oct 2018 17:09:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1303

v=0
o=root 340951957 340951957 IN IP4 192.168.30.161
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.30.161
t=0 0
m=audio 10208 RTP/AVP 97 0 8 3 18 4 4 112 112 5 5 7 7 110 110 111 111 9 9 102 102 115 115 116 116 117 117 10 10 118 118 119 119
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:117 speex/16000
a=rtpmap:10 L16/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:119 speex/32000
a=ptime:20
a=sendrecv

---
-- Called SIP/2051

<--- SIP read from UDP:192.168.32.201:1025 --->
SIP/2.0 100 Trying
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
User-Agent: Avaya one-X Deskphone 6.2.2.17 (40235)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.32.201:1025 --->
SIP/2.0 180 Ringing
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>;tag=-3bf389cf5bc8cc50-486e6158_T2051192.168.32.201
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
Allow: INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PRACK,PUBLISH,UPDATE
User-Agent: Avaya one-X Deskphone 6.2.2.17 (40235)
Contact: <sip:2051@192.168.32.201;transport=udp>
Accept-Language: en
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2051@192.168.32.201;transport=udp>
-- SIP/2051-0000030c is ringing
> 0x7f43701b3a20 -- Probation passed - setting RTP source address to 192.168.31.67:2112
Scheduling destruction of SIP dialog '03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.32.201:5060:
CANCEL sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled SIP/2.0
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
Max-Forwards: 70
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.16.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060' in 6400 ms (Method: INVITE)
== Spawn extension (public, 2051, 1) exited non-zero on 'OOH323/Avaya-865'

<--- SIP read from UDP:192.168.32.201:1025 --->
SIP/2.0 200 OK
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
User-Agent: Avaya one-X Deskphone 6.2.2.17 (40235)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.32.201:1025 --->
SIP/2.0 487 Request Terminated
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>;tag=-3bf389cf5bc8cc50-486e6158_T2051192.168.32.201
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
User-Agent: Avaya one-X Deskphone 6.2.2.17 (40235)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.32.201:5060:
ACK sip:2051@192.168.32.201;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.30.161:5060;branch=z9hG4bK6ae47cf0
Max-Forwards: 70
From: "teste" <sip:asterisk@192.168.30.161>;tag=as3b039e3c
To: <sip:2051@192.168.32.201;transport=udp;avaya-sc-enabled>;tag=-3bf389cf5bc8cc50-486e6158_T2051192.168.32.201
Contact: <sip:asterisk@192.168.30.161:5060>
Call-ID: 03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.16.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '03c0f9d82ea44cdc0afaade65b57f840@192.168.30.161:5060' in 6400 ms (Method: INVITE)