Hi guys, I am testing asterisk and linphonec for their flexibility for certain voip scenarios and I am running into this:
Tried to get linphone to use dtmf info method towards Asterisk, but everytime it still advertises 101 NTE. Tried first with the distro linphone (3.3.2-3) and faced the problem. Then just installed from source 3.4.3 and oRTP with same result. What can I do to get INFO to work? Any tip is appreciated. See the INVITE: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5071;rport;branch=z9hG4bK1155263762 From: <sip:[email protected]:5071>;tag=1343269921 To: <sip:[email protected]> Call-ID: 1249165265 CSeq: 20 INVITE Contact: <sip:[email protected]> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.4.3 (eXosip2/3.3.0) Subject: Phone call Content-Length: 316 v=0 o=linphone1 2260 2260 IN IP4 192.168.0.198 s=Talk c=IN IP4 192.168.0.198 t=0 0 m=audio 7078 RTP/AVP 112 111 110 0 8 101 a=rtpmap:112 speex/32000 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ----------- On my config I have: [net] download_bw=0 upload_bw=0 firewall_policy=0 mtu=0 [sip] sip_port=5071 guess_hostname=1 contact=sip:[email protected] inc_timeout=15 use_info=1 use_rfc2833=0 use_ipv6=0 register_only_when_network_is_up=1 default_proxy=-1 [rtp] audio_rtp_port=7078 video_rtp_port=9078 audio_jitt_comp=60 video_jitt_comp=0 nortp_timeout=30 ----------On Asterisk sip.conf I have: > [linphone1] > type=friend > secret=12345 > host=dynamic > port=5071 > callerid=linphone1 <2021> > dtmfmode=info > disallow=all > allow=ulaw _______________________________________________ Linphone-users mailing list [email protected] https://lists.nongnu.org/mailman/listinfo/linphone-users
