Codec Priority is just arranging our List of available codecs. Switch / other parties will choose the codec, based on which codec is matched by sent priority codec list with them in ascending order. If PCMU codec is only available codec on answerer side, then they will choose only PCMU even after you set the priority list. PCMU will selected for the call even we placed PCMU codec on the last row in the priority list. ![]() If you want to disable PCMU/Any codec on PJSIP library side, then set the default value as 0 when setting the priority. Resident evil 3 pc full rip cd. Did you check codec priority value like below? The codec priority will be arranged like below, CODEC_NAME/CODEC_TYPE PRIORITY_VALUE G729/8000/1 133 GSM/8000/1 132 speex/8000/1 131 speex/16000/1 131 speex/48000/1 130 PCMU/8000/1 128 iLBC/8000/1 127 So try with disabling PCMU codec. If then freeswitch chooses G729 codec, sure the priority value is wrongly assigned. The higher priority value is takes first place. So try with high value for G729 codec than other codecs. I put my observation above, kindly try this and ask your queries on comments! ![]() Making changes only to the codec priority on the phone. How to configure SIP Trunking for Asterisk IP PBX based systems Our service is 100% compatible with Asterisk using either standard SIP registration or IP authentication where SIP trunks are configured as such. Asterisk is the base software behind many open-source PBX distributions, including FreePBX, Trixbox and Elastix, and is also the enabler behind many other ITSPs and commercial PABX manufacturers. This guide is aimed at Asterisk's SIP stack via the sip.conf file, it does not deal with real-time configuration via a back-end database, however, the principles are the same and the appropriate options should be transposed as such. The configuration of SIP trunks for the PJSIP stack within Asterisk versions 12, 13 and greater is forthcoming and will be posted in a separate FAQ entry. Once you have set up and configured Asterisk, you can use the following details to start making calls. These details are visible on your customer control panel if you have been allocated a SIP trunk.
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