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Russell Horn

2006-Jun-13 10:22 UTC

[Asterisk-Users] No incoming sip calls

Hi folks - I've recently returned to asterisk after an eighteen month break.

I've two sip providers - gradwell in the UK (inbound and outbound)
and talklite in the US (outbound only).

I've managed to get outbound dialing working but am not receiving any
calls from gradwell.

I've included my sip.conf and extensions.conf as well as the output
from tethereal. When a call is placed to rgadwell I'm seeing no sip
traffic whatsoever on asterisk. My aim is to have inbound calls ring
SIP extension 2201

I'm guessing this is something pretty straightforward, but any help
would be much appreciated.

Thanks,

Russell.

sip.conf

[general]
context=incoming                ; Default context for incoming calls
register => 7960xxx:yyyy@sip.gradwell.net/2001
register => 9479xxx:zzzz@sip.talklite.net
port=5060                       ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0                ; address to bind to (0.0.0.0 binds to all)
nat=yes                 ; NAT settings
allow=all

[Gradwell]
type=peer
username=796xxxx
fromuser=796xxxx
secret=yyyy
host=sip.gradwell.net
context=flat
fromdomain=sip.gradwell.net
nat=yes
allow=all
canreinvite=no
dtmfmode=inband
qualify=yes

[talklite]
type=peer
username=9479xxxx
qualify=yes
secret=zzzz
host=sip.talklite.net
canreinvite=yes
disallow=all
allow=ulaw

[2201]
type=friend
context=flat
username=albanach
secret=aaaa
defaultip=192.168.1.100
qualify=yes
type=friend
callerid="Russell Horn" <0000>
host=dynamic
nat=no                       ; X-Lite is behind a NAT router
canreinvite=yes               ; Typically set to NO if behind NAT
allow=all


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

extensions.conf

[general]
static=yes
writeprotect=no

[globals]
TRUNK=Gradwell
TRUNKMSD=1                                      ; MSD digits to strip
(usually 1 or 0)

PHONES1=SIP/2201


[flat]
include => home
include => outgoing

[home]
exten => 2201,1,Dial(${PHONES1},20,Ttm)
exten => 2201,2,Macro(vmessage,${PHONES1VM})
exten => 2201,3,Hangup

[outgoing]
ignorepat => 9
ignorepat => 8
exten => _9.,1,Dial(SIP/${EXTEN:1}@Talklite)
exten => _8.,1,Dial(SIP/${EXTEN:1}@Gradwell)

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

linux:/etc/asterisk # tethereal -R "sip"
Capturing on eth0
  0.000000 207.44.248.78 -> 192.168.1.102 SIP Request: OPTIONS
sip:s@192.168.1.102
  0.000831 192.168.1.102 -> 207.44.248.78 SIP Status: 404 Not Found
  1.350584 192.168.1.102 -> 192.168.1.100 SIP Request: OPTIONS
sip:2201@192.168.1.100:5060
  1.350730 192.168.1.102 -> 207.44.248.78 SIP Request: OPTIONS
sip:sip.talklite.net
  1.350887 192.168.1.102 -> 193.111.200.56 SIP Request: OPTIONS
sip:sip.gradwell.net
  1.369388 192.168.1.100 -> 192.168.1.102 SIP Status: 200 OK
  1.455492 207.44.248.78 -> 192.168.1.102 SIP Status: 404 Not Found
  1.502618 193.111.200.56 -> 192.168.1.102 SIP Status: 404 Invalid
account for voicemail
  1.552845 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
  1.654933 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying    (1
bindings)
  1.655832 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
  1.657951 207.44.248.78 -> 192.168.1.102 SIP Status: 401 Unauthorized
   (1 bindings)
  1.658229 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
  1.770875 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying    (1
bindings)
  1.773894 207.44.248.78 -> 192.168.1.102 SIP Status: 200 OK    (1 bindings)
  1.792718 193.111.200.56 -> 192.168.1.102 SIP Status: 401
Unauthorized    (0 bindings)
  1.793529 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
  1.937253 193.111.200.56 -> 192.168.1.102 SIP Status: 200 OK    (1 bindings)

Jonathan Attwood

2006-Jun-13 11:33 UTC

head link

[Asterisk-Users] No incoming sip calls

Could your register line require attention ? (2001?)

 7960xxx:yyyy@sip.gradwell.net/2001 - I thought your target was 2201?

On 13/06/06, Russell Horn <albanach@gmail.com>
wrote:> Hi folks - I've recently returned to asterisk after an eighteen month
break.
>
> I've two sip providers - gradwell in the UK (inbound and outbound)
> and talklite in the US (outbound only).
>
> I've managed to get outbound dialing working but am not receiving any
> calls from gradwell.
>
> I've included my sip.conf and extensions.conf as well as the output
> from tethereal. When a call is placed to rgadwell I'm seeing no sip
> traffic whatsoever on asterisk. My aim is to have inbound calls ring
> SIP extension 2201
>
> I'm guessing this is something pretty straightforward, but any help
> would be much appreciated.
>
> Thanks,
>
> Russell.
>
> sip.conf
>
> [general]
> context=incoming                ; Default context for incoming calls
> register => 7960xxx:yyyy@sip.gradwell.net/2001
> register => 9479xxx:zzzz@sip.talklite.net
> port=5060                       ; UDP Port to bind to (SIP standard
> port is 5060)
> bindaddr=0.0.0.0                ; address to bind to (0.0.0.0 binds to all)
> nat=yes                 ; NAT settings
> allow=all
>
> [Gradwell]
> type=peer
> username=796xxxx
> fromuser=796xxxx
> secret=yyyy
> host=sip.gradwell.net
> context=flat
> fromdomain=sip.gradwell.net
> nat=yes
> allow=all
> canreinvite=no
> dtmfmode=inband
> qualify=yes
>
> [talklite]
> type=peer
> username=9479xxxx
> qualify=yes
> secret=zzzz
> host=sip.talklite.net
> canreinvite=yes
> disallow=all
> allow=ulaw
>
> [2201]
> type=friend
> context=flat
> username=albanach
> secret=aaaa
> defaultip=192.168.1.100
> qualify=yes
> type=friend
> callerid="Russell Horn" <0000>
> host=dynamic
> nat=no                       ; X-Lite is behind a NAT router
> canreinvite=yes               ; Typically set to NO if behind NAT
> allow=all
>
>
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
>
> extensions.conf
>
> [general]
> static=yes
> writeprotect=no
>
> [globals]
> TRUNK=Gradwell
> TRUNKMSD=1                                      ; MSD digits to strip
> (usually 1 or 0)
>
> PHONES1=SIP/2201
>
>
> [flat]
> include => home
> include => outgoing
>
> [home]
> exten => 2201,1,Dial(${PHONES1},20,Ttm)
> exten => 2201,2,Macro(vmessage,${PHONES1VM})
> exten => 2201,3,Hangup
>
> [outgoing]
> ignorepat => 9
> ignorepat => 8
> exten => _9.,1,Dial(SIP/${EXTEN:1}@Talklite)
> exten => _8.,1,Dial(SIP/${EXTEN:1}@Gradwell)
>
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
>
> linux:/etc/asterisk # tethereal -R "sip"
> Capturing on eth0
>  0.000000 207.44.248.78 -> 192.168.1.102 SIP Request: OPTIONS
> sip:s@192.168.1.102
>  0.000831 192.168.1.102 -> 207.44.248.78 SIP Status: 404 Not Found
>  1.350584 192.168.1.102 -> 192.168.1.100 SIP Request: OPTIONS
> sip:2201@192.168.1.100:5060
>  1.350730 192.168.1.102 -> 207.44.248.78 SIP Request: OPTIONS
> sip:sip.talklite.net
>  1.350887 192.168.1.102 -> 193.111.200.56 SIP Request: OPTIONS
> sip:sip.gradwell.net
>  1.369388 192.168.1.100 -> 192.168.1.102 SIP Status: 200 OK
>  1.455492 207.44.248.78 -> 192.168.1.102 SIP Status: 404 Not Found
>  1.502618 193.111.200.56 -> 192.168.1.102 SIP Status: 404 Invalid
> account for voicemail
>  1.552845 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
> sip:sip.talklite.net
>  1.654933 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying    (1
bindings)
>  1.655832 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
> sip:sip.gradwell.net
>  1.657951 207.44.248.78 -> 192.168.1.102 SIP Status: 401 Unauthorized
>   (1 bindings)
>  1.658229 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
> sip:sip.talklite.net
>  1.770875 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying    (1
bindings)
>  1.773894 207.44.248.78 -> 192.168.1.102 SIP Status: 200 OK    (1
bindings)
>  1.792718 193.111.200.56 -> 192.168.1.102 SIP Status: 401
> Unauthorized    (0 bindings)
>  1.793529 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
> sip:sip.gradwell.net
>  1.937253 193.111.200.56 -> 192.168.1.102 SIP Status: 200 OK    (1
bindings)
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Kevin P. Fleming

2006-Jun-14 13:59 UTC

head link

[Asterisk-Users] No incoming sip calls

----- Russell Horn <albanach@gmail.com> wrote:
> bindaddr=0.0.0.0                ; address to bind to (0.0.0.0 binds to
> all)
> externalip=yyy.yyy.yyy.yy
> nat=yes                 ; NAT settings
> allow=all
> canreinvite=no
Check your sip.conf.sample again... it's not 'externalip'.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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