[asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1

I can create an audio call file and specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course),

[asterisk-users] Asterisk installation script on CentOS7 with systemd

Hello, For a new project, I'm adapting existing installation script to CentOS 7. I must admit I don't understand how to adapt things to systemd. Here are my questions: 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib. Do you think such directory and matching Makefile target

[asterisk-users] Polycom SoundStation IP 6000 does not register

Hi, I am pulling my hair for days now... I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with my Asterisk. There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... Simple Question: Does anybody have a

[asterisk-users] Opening video file to play

I am trying to play a video file. It is failing saying *File /tmp/video does not exist in any format *>* Unable to open /tmp/video (format ulaw|h263|h264)* I am setting the video codec h263 in the call file. File exists and is readable by all. looking at the code and attaching with the

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

Le 19/12/2016 à 17:10, Olivier a écrit : 2016-12-19 16:11 GMT+01:00 Jean Aunis >: Le 19/12/2016 à 15:54, Olivier a écrit : Running systemctl start asterisk fails with : Dec 19 15:43:08 foobar systemd: PID file

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

2016-12-19 16:11 GMT+01:00 Jean Aunis : > Le 19/12/2016 à 15:54, Olivier a écrit : > > Hello, > > For a new project, I'm adapting existing installation script to CentOS 7. > I must admit I don't understand how to adapt things to systemd. > > Here are my questions: > > 1. I

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd

Le 19/12/2016 à 15:54, Olivier a écrit : Hello, For a new project, I'm adapting existing installation script to CentOS 7. I must admit I don't understand how to adapt things to systemd. Here are my questions: 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib. Do you think

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

can you provide the configuration on sip.conf file? Do you have the following settings under the account number or ext number? host=dynamic nat=yes for instance my configuration sip.conf file is as follow: [1005] type=friend context=sip-phone call-limit=1 trustrpid=no

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

2016-12-19 16:26 GMT+01:00 Yves : > Hi, > > I am pulling my hair for days now... > > I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register > with my Asterisk. > > There are no SIP Packets arriving at my asterisk at all... and it has > nothing to do with a

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

On 12/19/2016 10:26 AM, Yves wrote: There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... I can ping the phone from the asterisk, If both of these items are true, then I'd look at the phone configurations. Does the provisioning

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

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

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

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