any firewall in between?
On 9/18/07, Richard <[EMAIL PROTECTED]> wrote: > > Sorry if this comes thru twice, I had the wrong account selected to send > the > first time... > > > Callers to the number get ringing, I get stuff in my asterisk console, and > it calls my softphone and ata, but answering either gets silence, and the > caller gets the ringing stop, if they wait ages they get the stanaphone > voicemail. > > I have had the account for ages, and it never has worked, other sip > incoming > works ok so I don't think its any issues, and the machine is the DMZ of > the > adsl router so it should be forwarded for everything. > > These are the relevant snips of the file and the console output. > > ------sip.conf----- > [general] > context=mainmenu > allowguest=yes > allowoverlap=yes > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=yes > pedantic=no > allow=all > allow=g729 > rtptimeout=4 (tried this on the default of 30 and it just makes it take > longer to give the error, and I like it low incase the internet dies I > don't > end up talking to nothing for a long time without realizing it.) > compactheaders = yes > > > externip = 60.xxxxxx (our static IP is here) > localnet=192.168.0.0/255.255.0.0; > nat=yes > canreinvite=no > > ; richards stanaphone incoming to ext 8800 > register => 089xyz:[EMAIL PROTECTED]/8800 > ; richards italk to ext 8800 > register => 64997xxxxx:[EMAIL PROTECTED]/8800 > > ------- later down in it. > > > [stanaphone-richard] > type=friend > username=089xxxxx > fromuser=089xxxxx (all the same, and as stanaphone give in the sip config) > authname=089xxxxx > secret=xxxxxxxx (as stanaphone give in the sip config > host=sip.stanaphone.com > allow=all (tried that since the softphoen uses pcm when it works - no > change) > allow=g729 > allow=gsm > dtmfmode=rfc2833 > insecure=very > canreinvite=no > qualify=yes > nat=yes > port=5060 > context=richardincoming > mohinterpret=better > > > > I don't believe that the extensions.conf is a problem since I have other > voips going to the same 8800 extension and being handled right. > > What I get in the console on an incoming call to the stanaphone number is. > > > -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/089xxxxx-081c8b08", > "9974xxxx") in new stack > -- Executing [EMAIL PROTECTED]:2] NoOp("SIP/089xxxxx-081c8b08", > "") > in new stack > -- Executing [EMAIL PROTECTED]:3] Dial("SIP/089xxxxx-081c8b08", > "SIP/richard&SIP/richardsoftphone|15|tr") in new stack > -- Called richard > -- Called richardsoftphone > -- SIP/richardsoftphone-081d1348 is ringing > -- SIP/richard-081cca70 is ringing > -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 > [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: > Disconnecting > call 'SIP/089xxxxx-081c8b08' for lack of RTP activity in 5 seconds > == Spawn extension (richardincoming, 8800, 3) exited non-zero on > 'SIP/089xxxxx-081c8b08' > [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum > retries exceeded on transmission > [EMAIL PROTECTED] for seqno 200 (Critical > Response) > [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum > retries exceeded on transmission > [EMAIL PROTECTED] for seqno 200 (Critical > Response) > [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum > retries exceeded on transmission > [EMAIL PROTECTED] for seqno 200 (Critical > Response) > > Those continue on for quite some time and then stop (will get about 7 or 8 > of the critical error) > > > The lack of RTP everywhere makes it look to be a nat issue, but I have > done > everything I can think of to have that work, and the config is the same > other then host, username and password on italk which is working fine. I > have googled for the Maximum retries exceeded on transmission - I could > only > see some stuff related to broken sip phones, not a voip server. > > Alternativly, since it seems that stanaphone is a bit of a hit and miss > from > some other reading, is there any other functional US inwards provider for > free that doesn't need a credit card that works well with asterisk? The > softphone works, but I really need to get it going to my phones in the > house > instead. Soft client was closed when testing the asterisk. > > Many thanks. > > Richard Malcolm-Smith... > > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --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 >
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