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

2017-01-03 Thread Dmitriy Serov
Joshua, issue has been filed. Thank you! https://issues.asterisk.org/jira/browse/ASTERISK-26689 03.01.2017 20:58, Joshua Colp пишет: On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1]

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

2017-01-03 Thread Joshua Colp
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: > Yes, this means the remote end was not sending any audio stream. > But it shouldn't. > According to [1] before remote end send OK or ACK there is one way SDP, > no any audio stream. > PJSIP channel (option rtp_timeout) does not take this

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

2016-12-19 Thread Dmitriy Serov
Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016

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

2016-12-19 Thread Jean Aunis
This means the remote end was not sending any audio stream, or the audio stream was not received by Asterisk. The problem may have many different reasons, but often it is a network-related issue. Le 16/12/2016 à 21:19, Dmitriy Serov a écrit : Today I faced a problem. Please help to solve

[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