Dears, kindly I have the below scenario: [SoftPhone1]<——Channel1——>[FreeSwitch1]<——Channel2——>[FreeSwitch2]<——Chanel3——>[SoftPhone2] Channel1 codec G722 Channel3 codec G722 I want to enforce Channel 3 between FreeSwitch instances to be GSM, I change the Dial plane of freeSwitch1 to and the same for FS2 <action application="set" data=“absolute_codec_string=GSM"/> <action application="bridge" data="sofia/internal/$1@10.8.0.22"/> However when I run call from FS2 to FS1, FS1 reply with the below putting “a=rtpmap:106 opus/48000/2” as the only available codec as below, your kind help will be appreciated recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H From: "Extension 1000" <sip:1000@10.8.0.29>;tag=U82NjmSa9S9Kg To: <sip:1000@10.8.0.22>;tag=g0eNeprr8e4Fg Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c CSeq: 89562813 INVITE Contact: <sip:1000@10.8.0.22:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 261 X-FS-Display-Name: 1000 X-FS-Display-Number: sip:1000@10.8.0.22 X-FS-Support: update_display,send_info Remote-Party-ID: "1000" <sip:1000@10.8.0.22>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22 s=FreeSWITCH c=IN IP4 10.8.0.22 t=0 0 m=audio 25836 RTP/AVP 106 101 a=rtpmap:106 opus/48000/2 a=fmtp:106 useinbandfec=1; minptime=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Thanks, Ahmed Habiba.