any firewall in between?<br><br><br><div><span class="gmail_quote">On 9/18/07, <b class="gmail_sendername">Richard</b> <<a href="mailto:trading@richms.com">trading@richms.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Sorry if this comes thru twice, I had the wrong account selected to send the<br>first time...<br><br><br>Callers to the number get ringing, I get stuff in my asterisk console, and<br>it calls my softphone and ata, but answering either gets silence, and the
<br>caller gets the ringing stop, if they wait ages they get the stanaphone<br>voicemail.<br><br>I have had the account for ages, and it never has worked, other sip incoming<br>works ok so I don't think its any issues, and the machine is the DMZ of the
<br>adsl router so it should be forwarded for everything.<br><br>These are the relevant snips of the file and the console output.<br><br>------sip.conf-----<br>[general]<br>context=mainmenu<br>allowguest=yes<br>allowoverlap=yes
<br>bindport=5060<br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>srvlookup=yes<br>pedantic=no<br>allow=all<br>allow=g729<br>rtptimeout=4 (tried this on the default of 30 and it just makes it take<br>longer to give the error, and I like it low incase the internet dies I don't
<br>end up talking to nothing for a long time without realizing it.)<br>compactheaders = yes<br><br><br>externip = 60.xxxxxx (our static IP is here)<br>localnet=<a href="http://192.168.0.0/255.255.0.0">192.168.0.0/255.255.0.0
</a>;<br>nat=yes<br>canreinvite=no<br><br>; richards stanaphone incoming to ext 8800<br>register => <a href="http://089xyz:xxxxxxxx@sip.stanaphone.com/8800">089xyz:xxxxxxxx@sip.stanaphone.com/8800</a><br>; richards italk to ext 8800
<br>register => <a href="http://64997xxxxx:xxxxx@akl.italk.co.nz/8800">64997xxxxx:xxxxx@akl.italk.co.nz/8800</a><br><br>------- later down in it.<br><br><br>[stanaphone-richard]<br>type=friend<br>username=089xxxxx<br>fromuser=089xxxxx (all the same, and as stanaphone give in the sip config)
<br>authname=089xxxxx<br>secret=xxxxxxxx (as stanaphone give in the sip config<br>host=<a href="http://sip.stanaphone.com">sip.stanaphone.com</a><br>allow=all (tried that since the softphoen uses pcm when it works - no<br>
change)<br>allow=g729<br>allow=gsm<br>dtmfmode=rfc2833<br>insecure=very<br>canreinvite=no<br>qualify=yes<br>nat=yes<br>port=5060<br>context=richardincoming<br>mohinterpret=better<br><br><br><br>I don't believe that the
extensions.conf is a problem since I have other<br>voips going to the same 8800 extension and being handled right.<br><br>What I get in the console on an incoming call to the stanaphone number is.<br><br><br> -- Executing [
8800@richardincoming:1] NoOp("SIP/089xxxxx-081c8b08",<br>"9974xxxx") in new stack<br> -- Executing [8800@richardincoming:2] NoOp("SIP/089xxxxx-081c8b08", "")<br>in new stack<br> -- Executing [
8800@richardincoming:3] Dial("SIP/089xxxxx-081c8b08",<br>"SIP/richard&SIP/richardsoftphone|15|tr") in new stack<br> -- Called richard<br> -- Called richardsoftphone<br> -- SIP/richardsoftphone-081d1348 is ringing
<br> -- SIP/richard-081cca70 is ringing<br> -- SIP/richard-081cca70 answered SIP/08923542-081c8b08<br>[Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting<br>call 'SIP/089xxxxx-081c8b08' for lack of RTP activity in 5 seconds
<br> == Spawn extension (richardincoming, 8800, 3) exited non-zero on<br>'SIP/089xxxxx-081c8b08'<br>[Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum<br>retries exceeded on transmission<br><a href="mailto:2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12">
2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12</a> for seqno 200 (Critical<br>Response)<br>[Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum<br>retries exceeded on transmission<br><a href="mailto:2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12">
2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12</a> for seqno 200 (Critical<br>Response)<br>[Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum<br>retries exceeded on transmission<br><a href="mailto:2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12">
2566BD0-E7EB11D3-B19BE26B-1D26484B@66.114.240.12</a> for seqno 200 (Critical<br>Response)<br><br>Those continue on for quite some time and then stop (will get about 7 or 8<br>of the critical error)<br><br><br>The lack of RTP everywhere makes it look to be a nat issue, but I have done
<br>everything I can think of to have that work, and the config is the same<br>other then host, username and password on italk which is working fine. I<br>have googled for the Maximum retries exceeded on transmission - I could only
<br>see some stuff related to broken sip phones, not a voip server.<br><br>Alternativly, since it seems that stanaphone is a bit of a hit and miss from<br>some other reading, is there any other functional US inwards provider for
<br>free that doesn't need a credit card that works well with asterisk? The<br>softphone works, but I really need to get it going to my phones in the house<br>instead. Soft client was closed when testing the asterisk.
<br><br>Many thanks.<br><br>Richard Malcolm-Smith...<br><br><br><br>_______________________________________________<br><br>Sign up now for AstriCon 2007! September 25-28th. <a href="http://www.astricon.net/">http://www.astricon.net/
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