[asterisk-users] stanaphone issues. can someone verify my config?

[Al lists]

any firewall in between?


On 9/18/07, Richard <trading at richms.com> 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:xxxxxxxx at sip.stanaphone.com/8800
> ; richards italk to ext 8800
> register => 64997xxxxx:xxxxx at akl.italk.co.nz/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 [8800 at richardincoming:1] NoOp("SIP/089xxxxx-081c8b08",
> "9974xxxx") in new stack
>     -- Executing [8800 at richardincoming:2] NoOp("SIP/089xxxxx-081c8b08",
> "")
> in new stack
>     -- Executing [8800 at richardincoming: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
> 2566BD0-E7EB11D3-B19BE26B-1D26484B at 66.114.240.12 for seqno 200 (Critical
> Response)
> [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum
> retries exceeded on transmission
> 2566BD0-E7EB11D3-B19BE26B-1D26484B at 66.114.240.12 for seqno 200 (Critical
> Response)
> [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum
> retries exceeded on transmission
> 2566BD0-E7EB11D3-B19BE26B-1D26484B at 66.114.240.12 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...
>
>
>
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