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

Sun Dec 16 05:44:35 CST 2007

Sorry, being really busy recently and only now have the time to dedicate to
this (finished uni for the summer break)

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.

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.

; richards stanaphone incoming

;register => 0892xxxx: (MY PASSWORD)@sip.stanaphone.com/0892xxxx

register => 0892xxxx: (MY PASSWORD)@sip.stanaphone.com/101

(tried it both ways, having the stanaphone number as extension makes no
difference)

101 just goto's a thing that answers, plays a voice and thenputs it on hold
which work on all other sip providers.

[stanaphone-richard]

type=friend

username=0892xxxx

secret=(MY PASSWORD)

host=sip.stanaphone.com

allow=all

;allow=g729

;allow=gsm

dtmfmode=rfc2833

insecure=very

canreinvite=no

qualify=yes

nat=yes

port=5060

context=richardincoming

mohinterpret=better

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Al lists
Sent: Monday, September 24, 2007 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stanaphone issues. can someone verify my
config?

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  <http://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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