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asterisk-config/sip.conf at master · RangeNetworks ...
https://github.com/RangeNetworks/asterisk-config/blob/master/sip.conf
View blame. ; WARNING do not change this file, but instead use sip-custom-register.conf and sip-custom-contexts.conf. ; as this will limit the amount of conflicts when upgrading. [general] bindport=5060 ; asterisk 1.6. ; UDP Port to bind to (SIP standard port for unencrypted UDP. ; and TCP sessions is 5060) ; bindport is the local UDP port that ...
SIP - asteriskdocs.org
http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-A-SECT-2.html
The tos_sip, tos_audio, and tos_video settings control the TOS bits for the SIP messages, the RTP audio ... If you do not wish to have plain-text secrets in your sip.conf files, you can use md5secret to configure the MD5 hash that can be used for authentication. To generate the MD5 hash from the Linux console, use the following command:
asterisk/sip.conf.sample at master - GitHub
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample
Mirror of the official Asterisk (https://www.asterisk.org) Project repository. No pull requests here please. Use Gerrit: - asterisk/sip.conf.sample at master · asterisk/asterisk
How to enable TOS for RTP audio packets - Kolmisoft Wiki
http://wiki.kolmisoft.com/index.php/How_to_enable_TOS_for_RTP_audio_packets
You need to connect server via SSH and uncomment ;tos_audio=ef line in /etc/asterisk/sip.conf
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
Besides NAT problems I've also faced this issues on 3 cases: 1) Missconfigured parameter localnet: on /etc/asterisk/sip.conf make sure you set the network address for the phones. You can alos add multiple networks, for example: localnet=172.16.1.0/24 localnet=192.168.1.0/24
One-way audio during outbound calls - Asterisk SIP ...
https://community.asterisk.org/t/one-way-audio-during-outbound-calls/82347
Also using IAX2 solved my security problem where SIP scanners were trying to scan and extensions so I have only white-listed my ISP’s VoIP server. Bellow is my trunk config as Asterisk general and SIP config: sip.conf: [global] type=global user_agent=FPBX-15.0.16.38(16.7.0) default_outbound_endpoint=dpma_endpoint
TOS QoS Values Not Changing - FreePBX Community Forums
https://community.freepbx.org/t/tos-qos-values-not-changing/42765
The following values are present in sip_general_additional.conf in the latest stable FreePBX 14 Asterisk 13 release. tos_sip=cs3 tos_audio=ef tos_video=af41 Adding new values to Settings–>SIP Settings–>Chan-SIP does not seem to override these values. Also, a wireshark analysis of the UDP packets reveals they are being sent with a DSCP tag of 0x05. I cannot …
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