[asterisk-users] Asterisk 11 with Dovecot IMAP Email

2016-12-16 Thread Tammy Firefly 
Hi all,
Has anyone gotten dovecot 2.2 working with the asterisk 11 imap
voicemail system?  I can login via thunderbird or telnet to the imap
server just fine.

The below is logged:

[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:3176 mm_log: IMAP Error:
Can't open mailbox
{localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX:
invalid remote specification
[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2906 init_mailstream:
Can't connect to imap server
{localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX
[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2424 __messagecount:
Houston we have a problem - IMAP mailstream is NULL

As of now its not even sending a imap request to dovecot.

I am using a master user which I can successfully login as.  Ive tried
specifying individual usernames/passwords in the voicemail.conf file.
This doesnt work either.

Thanks
--Tammy



signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

2016-12-16 Thread Dmitriy Serov 

Today I faced a problem. Please help to solve this problem.

Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware 
v2.06(AAGJ.9)C1


Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk).
Call using early media (183 Session in progress) and rtp_timeout=10.
After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] 
res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-027b' for lack 
of RTP activity in 10 seconds


SIP dump is attached.

According to [1] before called user agent send OK or ACK there is one 
way SDP.
In sip dump (attached) i didn't find such SIP packets. Whether that call 
was canceled due to RTP inactivity?


Any help is welcome.

[1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt

INVITE sip:8xxx6yyy...@txxx37.ru SIP/2.0
Via: SIP/2.0/UDP 
11.111.11.11:5060;rport;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Max-Forwards: 70
From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: sip:8xxx6yyy...@txxx37.ru
Contact: "007" 
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
CSeq: 10072 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 90
Min-SE: 90
User-Agent: Keenetic Plus DECT
Authorization: Digest username="login", realm="ruvoip.net", 
nonce="1481885583/bcb53e85a740689479f116a96fc7086b", 
uri="sip:8xxx6yyy...@txxx37.ru", response="843f8211896b5b05fcf3a633d6d8eedf", 
algori
Content-Type: application/sdp
Content-Length:   326

v=0
o=- 3690874445 3690874445 IN IP4 11.111.11.11
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 0 8 109 96
c=IN IP4 11.111.11.11
b=TIAS:64000
a=rtcp:4023 IN IP4 11.111.11.11
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:109 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

[2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio TOS bits 
184
[2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio CoS mark 5
[2016-12-16 13:53:03] VERBOSE[13985] res_pjsip_logger.c: <--- Transmitting SIP 
response (352 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: 
CSeq: 10072 INVITE
Server: ruVoIP.net PBX
Content-Length:  0


[2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP 
response (849 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15
CSeq: 10072 INVITE
Server: ruVoIP.net PBX
Contact: 
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, MESSAGE, REFER, REGISTER
Content-Type: application/sdp
Content-Length:   266

v=0
o=- 3690874445 3690874447 IN IP4 222.222.222.22
s=ruVoIP.net PBX
c=IN IP4 222.222.222.22
t=0 0
m=audio 25094 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[2016-12-16 13:53:05] VERBOSE[6631] res_pjsip_logger.c: <--- Received SIP 
request (812 bytes) from UDP:11.111.11.11:5060 --->
UPDATE sip:222.222.222.22:5060 SIP/2.0
Via: SIP/2.0/UDP 
11.111.11.11:5060;rport;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk
Max-Forwards: 70
From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: sip:8xxx6yyy...@txxx37.ru;tag=f30b688b-3358-4107-9992-fb6e3923bc15
Contact: "007" 
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
CSeq: 10073 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 90
Min-SE: 90
Content-Type: application/sdp
Content-Length:   271

v=0
o=- 3690874445 3690874446 IN IP4 11.111.11.11
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 96
c=IN IP4 11.111.11.11
b=TIAS:64000
a=rtcp:4023 IN IP4 11.111.11.11
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv

[2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP 
response (910 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15
CSeq: 10073 UPDATE
Session-Expires: 90;refresher=uac
Require: timer
Contact: 
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, MESSAGE,