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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sorry, being really busy recently and only now have the time to
dedicate to this (finished uni for the summer break)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The asterisk is running on the machine that does the nat for the
network here at home, it is set as the dmz on the adsl router so all ports
should be coming into it.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I have done a sip debug and copied it (and sanitized it) and put
it here – well up till all the retrys start to appear.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>; richards stanaphone incoming<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>;register => 0892xxxx: (MY PASSWORD)@sip.stanaphone.com/0892xxxx<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>register => 0892xxxx: (MY PASSWORD)@sip.stanaphone.com/101<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>(tried it both ways, having the stanaphone number as extension
makes no difference)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>101 just goto’s a thing that answers, plays a voice and thenputs
it on hold which work on all other sip providers.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>[stanaphone-richard]<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>type=friend<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>username=0892xxxx<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>secret=(MY PASSWORD)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>host=sip.stanaphone.com<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>allow=all<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>;allow=g729<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>;allow=gsm<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>dtmfmode=rfc2833<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>insecure=very<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>canreinvite=no<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>qualify=yes<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>nat=yes<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>port=5060<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>context=richardincoming<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>mohinterpret=better<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Al lists<br>
<b>Sent:</b> Monday, September 24, 2007 7:33 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] stanaphone issues. can someone verify my
config?<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>any firewall in between?<br>
<br>
<o:p></o:p></p>
<p class=MsoNormal><span class=gmailquote>On 9/18/07, <b>Richard</b> <<a
href="mailto:trading@richms.com">trading@richms.com</a>> wrote:</span><o:p></o:p></p>
<p class=MsoNormal>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>
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