Thank you Alex, It's running without errors now and I can see the media flowing with 'rtp set debug on' but I can't still hear anything on the Asterisk's peers, any advice?
Elder 2011/7/4, Alex Balashov <[email protected]>: > 488 means no mutually acceptable codecs were negotiated between the > endpoints. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk <[email protected]> wrote: > >> I'm trying to get working SIPp with media but something is wrong (it's >> working well without media), please help: >> >> This is the command I send at SIPp server: >> ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err >> >> This is the result I see: >> Last Error: Aborting call on unexpected message for Call-Id >> '19-12768@12... >> >> What I see at sipp's logs: >> >> 2011-06-28 14:32:57:624 1309289577.624809: Aborting call on >> unexpected message for Call-Id '[email protected]': while expecting '100' >> (index 1), received 'SIP/2.0 488 Not acceptable here >> >> Via: SIP/2.0/UDP >> 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253 >> From: sipp <sip:[email protected]:5061>;tag=12768SIPpTag091 >> To: sut <sip:[email protected]:5060>;tag=as3614adc3 >> Call-ID: [email protected] >> CSeq: 1 INVITE >> Server: Asterisk PBX 1.8.4.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH >> Supported: replaces, timer >> Content-Length: 0 >> >> This is my asterisk 1.8's configuration: >> >> sip.conf >> [sipp] >> type=friend >> context=sipp >> host=dynamic >> port=6000 >> user=sipp >> canreinvite=no >> disallow=all >> allow=ulaw >> >> extensions.conf: >> [sipp] >> exten => 2005,1,Answer >> same=>n,Dial(SIP/intern,30) >> same=>n,Hangup() >> >> exten => 2006,1,Answer() >> same=> n,WaitMusicOnHold(4) >> same=> n,Hangup() >> >> >> I'm using sipp.3.1.src.tar.gz and I have installed it this way: >> ..sip.svn# make pcapplay >> >> Thanks in advance. >> >> Elder >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > -- Enviado desde mi dispositivo móvil -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
