Hello.Asterisk 13.7 (branch 13), res_pjsipLog file contains a lot of lines:[2016-01-11 15:33:00] ERROR[2862] res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact 17378/sip:17378@87.255.225.xxxx:5060[2016-01-11 15:33:..
Author : Dmitriy Serov
Home » Articles posted by Dmitriy Serov (paged 2)
The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP.There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use aster..
Hello.Recently I installed voipmonitor and voipmonitor-gui trial version. After examining it, I was amazed at the abundance of useful information that can and should be obtained from the work of Asterisk.1. The cost voipmonitor-gui too expensive ..
Hello.Continue to move from chan_sip to res_pjsip. For the work of my algorithms is very important to know the IP address of all trunks and endpoints (phones).In the case of chan_sip, I used PeerStatus AMI event through which was received the fact..
Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_..
Hello. Continuea months-longstrugglethat is associatedwith the transfer from chan_sip to res_pjsip. Where are many gates (GSM gate) that do not supportauthentication whensendingMESSAGE. For example, 4goip when relay incoming SMS. Using chan_sip it ..
Hello.I have plain text password for endpoint with outbound registration (someone elses server). My aim is to write it in pjsip.conf.md5 means that I know realm. I do not always know it.Is where any way?Dmit..
Hello.Voice quality when calling – this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.Because the overall quality of a call can only be determined upon completion, I did it in the HangUp hand..
Hello.Is there an analog option outofcall_message_context for pjsip?or: how to determine that the call is an outbound text message?Dmit..
Hello.Asterisk 13.2, PJSIP.Problem: I do not get any AMI events when changing the status of the contact.When using chan_sip I got peerstatus event. When using res_pjsip and devices (endpoint configuration) I got peerstatus event. When using res_pjsip..