Carlos Alberto Bernat Orozco
2006-Apr-20 07:24 UTC
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ;domain=mydomain.tld ;domain=mydomain.tld,mydomain-incoming ;domain=1.2.3.4 ;allowexternalinvites=no ;autodomain=yes ;pedantic=yes ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yes disallow=all allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes And these are the extensions: [xxxx] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx [xxxx2] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx As you can see I put the jitterbuffer, maxjitterbuffer and rtptimeout options. I think with this, the call has a huge improvement and I still reading about it. This is the CLI output with different commands: sip show peers Name/username Host Dyn Nat ACL Port Status usuario2/usuario2 10.xxx.xxx.xxx D 5060 Unmonitored usuario1/usuario1 10.xxx.xxx.xxx D 5060 Unmonitored 2 sip peers [2 online , 0 offline] sip show users Username Secret Accountcode Def.Context ACL NAT usuario2 usuario2 default No RFC3581 usuario1 usuario1 default No RFC3581 --- (8 headers 0 lines)--- Looking for 200.xxx.xxxx.xxx in default (domain ) Transmitting (no NAT) to 10.xxx.xxx.xxx:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.xxx.xxx.xxx ;rport;branch=z9hG4bK0a0101e20000001044479388000070d3000000d4;received10.xxx.xxx.xxx From: <sip:usuario2@200.xxx.xxx.xxx>;tag=312051512495 To: <sip:200.xxx.xxx.xxx>;tag=as767ed6bb Call-ID: DBBDE928-A279-4194-B78C-319FF0FCCDD9@10.xxx.xxx.xxx CSeq: 150 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:200.xxx.xxx.xxx> Accept: application/sdp Content-Length: 0 But I have another question. Our users surf the Internet by cable modems and we have a CMTS Motorola BSR 1000 with QoS options. I know I can configure it to manage QoS but I don't know very well how to do it. If somebody knows any tutorial or experiences administrating this device, please let me know Thanks again Carlos Bernat> Message: 8 > Date: Wed, 19 Apr 2006 15:46:21 -0500 > From: "Cavanna, Richard" <RCavanna@sychip.com> > Subject: [Asterisk-Users] RE: Delayed voice for 10 secs > To: <asterisk-users@lists.digium.com> > Message-ID: > <AB220F8DE6CD4F489EAB48B0020C64A976352A@tx01mailbox1.sychip.com> > Content-Type: text/plain; charset="us-ascii" > > Please post pertinent config files and a CLI output so the list can help > with the 10 sec delay > > You set codec selection in SIP.conf. This selects preferred codec from > top to bottom as well as jitter buffer settings and the RTP timeout. > > Sip.conf > disallow=all > allow=g729 > allow=gsm > allow=ulaw > jitterbuffer=yes > ;forcejitterbuffer=yes > maxjitterbuffer=1500 > rtptimeout=60 > > > As for the DTMF issue try to use rfc2833 > > in sip.conf define your extention > > [XXXX] > username=XXXX > type=friend > secret=XXXXX > qualify=no > port=5060 > nat=yes > mailbox=XXXX@device > host=dynamic > dtmfmode=rfc2833 > context=from-internal > canreinvite=no > callerid=device <XXXX> > > Rich > > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060420/25b1d583/attachment.htm
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