[asterisk-users] Testing Asterisk with media - sipp

Daniel - Asterisk earohuanca at gmail.com
Mon Jul 4 21:07:12 CDT 2011


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 <abalashov at evaristesys.com>:
> 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 <earohuanca at gmail.com> 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 at 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 '1-12768 at 127.0.0.1': 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:sipp at 127.0.0.1:5061>;tag=12768SIPpTag091
>> To: sut <sip:2005 at 192.168.1.18:5060>;tag=as3614adc3
>> Call-ID: 1-12768 at 127.0.0.1
>> 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



More information about the asterisk-users mailing list