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Dial out ok, but can't dial in or other extensions

Offline alt.testing

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Dial out ok, but can't dial in or other extensions
« on: July 22, 2012, 07:44:02 AM »
Hi,
I am having trouble with the registration of extensions I believe.
I have trunk set up, and can successfully dial out, but not in or other extensions:

Really stumped - I have done this by hand editing files a while ago on a simple install without SME integration, and there wasn't much to do.

But, I am not sure what I need else to do here. My extensions are registered, I have trunks set up and registered with iinet , and route with dial _X!

There are some extra peers and things here, for troubleshooting...



Code: [Select]
meredith*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
iinet2In_0174/039023XXXX   203.55.xxx.xxx              5060     OK (22 ms)
iinet2In_0173/039023XXXX   203.55.xxx.xxx      N      5060     Unmonitored
iinet2In_0172/039023XXXX   203.55.xxx.xxx              5060     OK (22 ms)
iinet2In_0171/039023XXXX   203.55.xxx.xxx              5060     OK (22 ms)
iinet2In/039023XXXX        203.55.xxx.xxx       N      5060     OK (21 ms)
Pilot/039023XXXX           203.55.xxx.xxx       N      5060     OK (22 ms)
5250/5250                  192.168.100.250  D       A  5060     OK (8 ms)
5249/5249                  192.168.100.249  D       A  5060     OK (9 ms)
5248/5248                  192.168.100.248  D       A  5060     OK (7 ms)




I just get extension not found, and also 403 Forbidden by my asterisk? -   with sip debug





<------------>
Audio is at 203.206.xxx.xxx port 11020
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.55.229.197:5060:
INVITE sip:5249@203.55.229.197 SIP/2.0
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;branch=z9hG4bK1bb2a1a1;rport
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>
Contact: <sip:039023XXXX@203.206.xxx.xxx>
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;privacy=off;screen=no
Date: Sun, 22 Jul 2012 05:36:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 22934 22934 IN IP4 203.206.xxx.xxx
s=session
c=IN IP4 203.206.xxx.xxx
t=0 0
m=audio 11020 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 203.55.229.197:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;received=203.206.xxx.xxx;branch=z9hG4bK1bb2a1a1;rport=5060
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---

<--- SIP read from 203.55.229.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;received=203.206.xxx.xxx;branch=z9hG4bK1bb2a1a1;rport=5060
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>;tag=SDssem199-1172530212-1342935365656
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 102 INVITE
WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXh4xp1jx4Tyx784hBW",algorithm=MD5,realm="BroadWorks"
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.55.229.197:5060:
ACK sip:5249@203.55.229.197 SIP/2.0
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;branch=z9hG4bK1bb2a1a1;rport
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>;tag=SDssem199-1172530212-1342935365656
Contact: <sip:039023XXXX@203.206.xxx.xxx>
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;privacy=off;screen=no
Content-Length: 0


---
Audio is at 203.206.xxx.xxx port 11020
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.55.229.197:5060:
INVITE sip:5249@203.55.229.197 SIP/2.0
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;branch=z9hG4bK57b0b94b;rport
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>
Contact: <sip:039023XXXX@203.206.xxx.xxx>
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;privacy=off;screen=no
Authorization: Digest username="039023XXXX", realm="BroadWorks", algorithm=MD5, uri="sip:5249@203.55.229.197", nonce="BroadWorksXh4xp1jx4Tyx784hBW", response="175f014544f70b24e9aef915dc1e3ec1", qop=auth, cnonce="58d98d61", nc=00000001
Date: Sun, 22 Jul 2012 05:36:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 22934 22935 IN IP4 203.206.xxx.xxx
s=session
c=IN IP4 203.206.xxx.xxx
t=0 0
m=audio 11020 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 203.55.229.197:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.206.xxx.xxx:5060;received=203.206.xxx.xxx;branch=z9hG4bK57b0b94b;rport=5060
From: "039023XXXX" <sip:039023XXXX@ch03.iibusiness.net.au>;tag=as23ec6d13
To: <sip:5249@203.55.229.197>
Call-ID: 2e540ae20486bf6c459295af20de98d0@ch03.iibusiness.net.au
CSeq: 103 INVITE







extensions.conf





[general]

static=yes       ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[globals]

        LOCALIP=192.168.100.1
        ALLOWHASHXFER=disabled
        BLINDBUSY=Operator
        BOUNCEALERT=
        CALLRECORD1=None
        EXTLEN=
        FAX=5001
        FAXDETECT=2
        INTRINGDELAY=15
        LTERM=NO
        MTIME=disabled
        OPERATOR=0
        PAGEPASS=
        PLAYBEEP=YES
        PLAYBUSY=YES
        PLAYCONGESTED=YES
        RINGDELAY=0
        SYSOP=5248
        VOIPMAX=4
        VOICEINSTR=YES
        DYNAMIC_FEATURES=>automon

;
;       Customer supplied Globals & Header data
;
        TESTGLOBAL=15
;
;       End of Customer supplied Globals & Header data
;

[from-internal]

        include => internal
[internal]

        include => internal-presets
        include => extensions
        include => utilities

        exten => _X.,1,agi(selintra,OutCluster,${EXTEN})

[priv_sibling]

        exten => _X.,1,agi(selintra,OutCluster,${EXTEN})

[qrxvtmny]

        include => internal-presets
        include => extensions
        include => conferences
        exten => _**XX.,1,Pickup(${EXTEN:2})
        include => utilities
        exten => _X!,1,agi(selintra,OutRoute,inbound)

        exten => t,1,Hangup
        exten => h,1,Hangup
        exten => i,1,Playtones(congestion)
[qrxvtmny-callback]

        exten => _X.,1,DISA(no-password,qrxvtmny)


;
[extensions]
        include => internal-presets
        include => parkedcalls

        exten => ,1,ParkedCall()
        exten => ,hint,park:@parkedcalls

        exten => _*8XX.,1,Pickup(${EXTEN:2}@PICKUPMARK)
        exten => 0,1,Goto(${SYSOP},1) ; accept zero as operator request

        exten => 5248,hint,SIP/5248
        exten => 5248,1,agi(selintra,InCall,)
        exten => *5248,1,Playback(silence/2)
        exten => *5248,n,Voicemail(5248,su)
        exten => *5248,n,Hangup
        exten => 5249,hint,SIP/5249
        exten => 5249,1,agi(selintra,InCall,)
        exten => *5249,1,Playback(silence/2)
        exten => *5249,n,Voicemail(5249,su)
        exten => *5249,n,Hangup
        exten => 5250,hint,SIP/5250
        exten => 5250,1,agi(selintra,InCall,)
        exten => *5250,1,Playback(silence/2)
        exten => *5250,n,Voicemail(5250,su)
        exten => *5250,n,Hangup
        exten => o,1,Background(pbx-transfer)
        exten => o,2,GoTo(0,1)
        exten => t,1,Hangup
        exten => h,1,Hangup
        exten => i,1,Playtones(congestion)

        exten => i,2,Hangup

[queues]
        exten => 5248,1,agi(selintra,Dial,5248,queue)
        exten => 5249,1,agi(selintra,Dial,5249,queue)
        exten => 5250,1,agi(selintra,Dial,5250,queue)
        exten => t,1,Hangup
        exten => h,1,Hangup
        exten => i,1,Playtones(congestion)


[conferences]

        exten => _30[0-7],1,Meetme(${EXTEN},Mp)

[from-trunk]
        include => from-pstn

[from-pstn]
        include => mainmenu

[mainmenu]

        include => Spitfire_sip

        exten => 039023XXXX,1,agi(selintra,Inbound,${EXTEN},039023XXXX)
        exten => 0390790170,1,agi(selintra,Inbound,${EXTEN},0390790170)
        exten => 0390790171,1,agi(selintra,Inbound,${EXTEN},0390790171)
        exten => 0390790172,1,agi(selintra,Inbound,${EXTEN},0390790172)
        exten => 0390790173,1,agi(selintra,Inbound,${EXTEN},0390790173)
        exten => 0390790174,1,agi(selintra,Inbound,${EXTEN},0390790174)

        exten => fax,1,GoToIf($["$FAX" = ""]?3:2)     ;no FAX defined - hangup
        exten => fax,2,GoTo(extensions,${FAX},1)
        exten => fax,3,Playtones(congestion)

        exten => t,1,GotoIf($["${OPEN}" = "YES"]?t,4)
        exten => t,2,Voicemail(${SYSOP},su)
        exten => t,3,Hangup
        exten => t,4,Goto(extensions,${SYSOP},1)
        exten => t,5,Hangup
        exten => h,1,Hangup
        exten => i,1,Playtones(congestion)





sip.conf

[general]
context=mainmenu
maxexpirey=180
defaultexpirey=160
limitonpeers=yes
notifybusy=yes
notifyringing=yes
notifyhold=yes
allowguest=no
alwaysauthreject=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
externip=203.206.183.139
localnet=192.168.100.1/255.255.255.0

#include sark_customer_sip_headers.conf
register => 039023XXXX@ch03.iibusiness.net.au:secret:039023XXXX@sip.vic.iibusiness.net.au
register => 039023XXXX@ch03.iibusiness.net.au:secret:039023XXXX@sip.vic.iibusiness.net.au/0390790173
#include sark_customer_sip_registrations.conf
;Internal IP phones

[5248]
type=peer
username=5248
secret=5248secret
mailbox=5248
host=dynamic
qualify=3000
canreinvite=no

callerid="5248" <5248>
pickupgroup=1

callgroup=1
call-limit=99
subscribecontext=extensions
context=from-internal

deny=0.0.0.0/0.0.0.0
permit=192.168.100.1/255.255.255.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dial=SIP/5248


[5249]
type=peer
username=5249
secret=5249secret
mailbox=5249
host=dynamic
qualify=3000
canreinvite=no
callerid="5249" <5249>
pickupgroup=1

callgroup=1
call-limit=99

subscribecontext=extensions
context=from-internal

deny=0.0.0.0/0.0.0.0
permit=192.168.100.1/255.255.255.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dial=SIP/5249


[5250]
type=peer
username=5250
secret=5250secret
mailbox=5250
host=dynamic
qualify=3000
canreinvite=no
context=internal
callerid="5250" <5250>
pickupgroup=1

callgroup=1
call-limit=99
subscribecontext=extensions
deny=0.0.0.0/0.0.0.0
permit=192.168.100.1/255.255.255.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw


;External Sip lines
;
;==========================================================================
;      This is where we deal with Sipura 3000 and SIP trunks (if any).
;      We generate 2 entries (peer and user) for Spa3K but only a peer
;      entry for SIP devices.
;==========================================================================
;

[Pilot]
type=friend
host=203.55.xxx.xxx
realm=ch03.iibusiness.net.au
fromdomain=ch03.iibusiness.net.au
pedantic=no
trustrpid=yes
sendrpid=yes
qualify=yes
registersip=yes
canreinvite=no
username=039023XXXX
fromuser=039023XXXX
defaultuser=039023XXXX
secret=secret
disallow=all
allow=gsm

allow=alaw
allow=ulaw
context=from-trunk
nat=yes

[iinet2In]
type=peer
host=203.55.xxx.xxx
realm=ch03.iibusiness.net.au
fromdomain=sip.vic.iibusiness.net.au
pedantic=no
trustrpid=yes
sendrpid=yes
qualify=yes
registersip=yes
canreinvite=no
username=039023XXXX
fromuser=039023XXXX
defaultuser=039023XXXX
secret=secret
disallow=all
allow=gsm

allow=alaw
allow=ulaw
context=from-trunk
nat=yes

[iinet2In_0171]
type=peer
host=sip.vic.iibusiness.net.au
realm=ch03.iibusiness.net.au
fromdomain=ch03.iibusiness.net.au
qualify=3000
canreinvite=no
username=039023XXXX
fromuser=039023XXXX
secret=secret
disallow=all
allow=gsm

allow=alaw
allow=ulaw

[iinet2In_0172]
type=peer
host=sip.vic.iibusiness.net.au
realm=ch03.iibusiness.net.au
fromdomain=ch03.iibusiness.net.au
qualify=3000
canreinvite=no
username=039023XXXX
fromuser=039023XXXX
secret=secret
disallow=all
allow=gsm

allow=alaw
allow=ulaw

[iinet2In_0173]
type=friend
host=203.55.xxx.xxx
realm=ch03.iibusiness.net.au
fromdomain=ch03.iibusiness.net.au
pedantic=no
trustrpid=yes
sendrpid=yes
qualify=no
registersip=yes
canreinvite=no
username=039023XXXX
fromuser=039023XXXX
defaultuser=039023XXXX
secret=secret
disallow=all
allow=gsm

allow=alaw
allow=ulaw
context=from-trunk
nat=yes
« Last Edit: July 22, 2012, 07:47:42 AM by alt.testing »

Offline m4st3rc1p0

  • 14
  • +0/-0
Re: Dial out ok, but can't dial in or other extensions
« Reply #1 on: September 03, 2012, 11:33:39 AM »
can you show your route parameters, this is most likely on your routing or dial plan.

guest22

Re: Dial out ok, but can't dial in or other extensions
« Reply #2 on: September 03, 2012, 01:28:48 PM »
Quote
and route with dial _X!
Quote

If your internal extensions have an extentions length of 4 (e.g. 5250) then set your dial pattern for outside lines to:

XXXXX.