I agree, seems odd you didn't have a [peername] section for your softphone in your sip.conf.
aren't 404 errors a likely symptom of this? :) Mojo Steve Totaro wrote: > Pete, > > You are connecting via a SIP softphone correct? Where is that in your > sip.conf? > > On Mon, Mar 17, 2008 at 11:42 AM, Pete Kay <[EMAIL PROTECTED]> wrote: > >> Hi, >> >> My sip.conf has the allow=gsm as shown in the following: >> >> >> [general] >> port = 5060 >> bindaddr = 0.0.0.0 >> context = others >> >> register =>outraspace:[EMAIL PROTECTED]/outraspace >> nat=yes >> externip=58.251.75.251 >> >> localnet=192.168.1.0/255.255.255.0 >> canreinvite=no >> disallow=all >> allow=ulaw >> allow=alaw >> allow=gsm >> qualify=yes >> >> All the sound files are in /var/lib/asterisk/sounds instead. Is it correct? >> >> I have tried both Wengo and xlite, but same result. >> >> I can't figure out what caused the 404 error. Any idea? >> >> >> Thank you so much for your help. >> >> Pete >> >> >> >> On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin Hoffmeister >> <[EMAIL PROTECTED]> wrote: >> >> >>> Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay: >>> >>>> Hi, >>>> >>>> >>>> Here is the SIP debug output for the playback test. Thank you so much >>>> for your help. >>>> >>> Hi Pete, >>> >>> >>> >>>> <------------> >>>> [Mar 18 05:33:08] -- Executing [EMAIL PROTECTED]:1] >>>> Answer("SIP/2000-081e0738", "") in new stack >>>> [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028 >>>> [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP >>>> [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP >>>> [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP >>>> >>> I do not see "gsm" here. Any reason not to allow that codec? Or did I >>> miss something? You wrote you enabled it, so it should be here IMO. >>> >>> >>> >>>> <--- Transmitting (NAT) to 192.168.1.102:5060 ---> >>>> SIP/2.0 404 Not Found >>>> Via: SIP/2.0/UDP >>>> >>>> >> 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102;rport=5060 >> >>>> From: 2001 <sip:[EMAIL PROTECTED]>;tag=2612560371 >>>> To: <sip:[EMAIL PROTECTED]>;tag=as0ca1ddb0 >>>> Call-ID: [EMAIL PROTECTED] >>>> CSeq: 20 OPTIONS >>>> User-Agent: Asterisk PBX >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>>> Supported: replaces >>>> Accept: application/sdp >>>> Content-Length: 0 >>>> >>> "404" does not sound good. Please, look which sound files exist on your >>> system (e.g. what does >>> find /usr/share/asterisk -file "vm-goodbye*" >>> say?) >>> >>> Another point: Which client do you use, is it Wengo or is it Xlite? Or >>> both? In that case: Any differences? >>> >>> >>> >>> >>> BR >>> Anselm >>> >>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users