Topic: Incoming call Problems

[Tib]

I have two DiD numbers I would like to utilize.

I have looked at the old docs page (aelintra seems to be down) and the pics etc don't seem to match up with the ver I have sail-2.2.1-673.

Could someone please give me a hand to get these inbound numbers working.

When I call these DiD numbers I get a "The extention you are calling does not exist ... etc"

Regards,

Tib

[Tib]

I have these 2 examples from Mynetfone that I have found

Sip.conf

Code: [Select]

;
; SIP Configuration for Asterisk
;
; Useful CLI commands to check peers/users:
;   sip show peers              Show all SIP peers (including friends)
;   sip show users              Show all SIP users (including friends)
;   sip show registry           Show status of hosts we register with
;
;   sip debug                   Show all SIP messages
;

[general]
context=sip-in                  ; Default context for incoming calls

port=5060                       ; UDP Port to bind to (SIP standard port is
                                ; 5060)

bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)

srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls

                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet

language=en                     ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers
                               
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP
                                ; activity when we're not on hold

rtpholdtimeout=300              ; Terminate call if 300 seconds of no RTP
                                ; activity

defaultexpirey=240              ; SIP registration timeout is 240 seconds

pedantic=no                     ; Don't check SIP headers for strict
                                ; compatibility

allow=all                       ; Allow all codecs

                                ; Register with MyFone SIP proxy.
                                ; Calls cannot be made/received without
                                ; registering with MyFone SIP proxy
register => <MyNetFoneNumber>@sip.myfone.com.au:<MyNetFonePassword>:<MyNetFoneNumber>@sip.myfone.com.au/<MyNetFoneNumber>


[myfone-sip]                    ; MyFone SIP proxy

type=friend
secret=<MyNetFonePassword>
username=<MyNetFoneNumber>
fromuser=<MyNetFoneNumber>
authname=<MyNetFoneNumber>
host=sip.myfone.com.au
insecure=very
nat=yes
disallow=all
allow=g729
qualify=yes
dtmfmode=rfc2833
canreinvite=no
context=myfone-inbound          ; Context to handle inbound calls from MyFone
                                ; SIP proxy


[2222]                          ; SIP phone connected to Asterisk.
                                ; It has extension number 2222 on Asterisk
type=friend
username=2222
fromuser=2222
host=dynamic
dtmfmode=rfc2833
secret=<Password>
disallow=all
allow=g729
context=default                 ; Context to handle calls made from SIP phone connected
                                ; to Asterisk
canreinvite=no

Extensions.conf

Code: [Select]

; Handle calls coming in from MyFone SIP proxy
;
[myfone-inbound]

; Incoming calls from MyFone to Asterisk are directed to the extension
; 2222 on Asterisk
; This is a SIP phone in this sample config.
;
exten => <MyNetFoneNumber>,1,Dial(SIP/2222,30)
exten => <MyNetFoneNumber>,2,Congestion
exten => <MyNetFoneNumber>,102,Busy

; Note: In this sample config, in the settings for extension 2222 its
; context is set as "default".
; So any outgoing calls made from extension 2222 are processed in this
; section.
;
[default]

; Send all outbound calls with prefix 3 via MyFone. Strip the prefix 3
; before sending call to MyFone SIP proxy
; Ex: if 30291234567 is dialled then the 3 is removed and 0291234567 is
; sent to the MyFone SIP proxy.
;
; Note :{EXTEN:1} => strip the first digit of number contained in
; variable EXTEN
;    myfone-sip is defined in sip.conf, 30 it the timeout
;
exten => _3.,1,Dial(SIP/${EXTEN:1}@myfone-sip,30)
exten => _3.,2,Congestion
exten => _3.,102,Busy

Can someone please help me with the incoming setup of this.

I seem to be running around in circles and getting no where.

My trunk setup atm

Code: [Select]

type=peer
host=sip10.mynetfone.com.au
qualify=yes
canreinvite=no
dtmfmode=rfc2833
fromdomain=sip10.mynetfone.com.au
username=09129509
fromuser=09129509
secret=<Mypassword>
authname=09129509
insecure=yes
nat=yes
disallow=all
allowg729
allow=alaw
allow=ulaw
registration string

Code: [Select]

09129509@sip10.mynetfone.com.au:<Mypassword>:09129509@sip10.mynetfone.com.au/09129509
Registration is ok and outbound calls with Mynetfone is working just no inbound calls.

In and outbound work on the soft phone they provide.

Regards,

Tib

[SARK devs]

set up a DiD with the number 09129509.  Route it to your extension and you should be good to go

Kind Regards

S

[Teviot]

Hi All

I to have an issue with incoming calls.

I have a UK number woth "voipfone.co.uk" and can make calls but unable to recieve calls.

The entry i have in the trunk is as follows. This is the left hand side (below) and nothing on the right hand side.

Code: [Select]

type=peer
host=sip.voipfone.co.uk
qualify=3000
canreinvite=no
username= "userid"
fromuser= "userid"
secret= "password"
disallow=all
allow=alaw
allow=ulaw
Can somebody point out what i am missing or have over looked.

Regards
Teviot

[SARK devs]

Usually this is due to what DNID (Dialed Number ID) the carrier is delivering (which also depends upon the registration string).

They will either deliver against the real number they have allocated to you (10 or 11 digits in UK) or they will deliver against the username field from your sip entry.

If you really want to find out which it is then you will need to look at the SIP invite they send you when a call comes in...  It will be of the form some_value@your.ip.address.  The value is effectively the dialled number (or DiD) that you want to catch so that's what you set in your PTT_DiD.   However, since in reality, it can only be eithet the number or the username, it's usually quicker to just build the PTT_DiD and see which one works.

Best

S

 

[Teviot]

Being new to SAIL or SARK on SME, how do i go about doing that?  The manual doesn't appear to follow what I see from ther server-manager panel unless I'm just not seeing the right place to set that up.

Also, selintra, Can I email or phone you to discuss the details of this?

Teviot

[SARK devs]

Being new to SAIL or SARK on SME, how do i go about doing that?

sip debug at the asterisk console.

Can I email or phone you to discuss the details of this?

I'm afraid not. At least, not without a support contract.  We do our best to assist our free-user community through the forum here, but we also have to eat.  Our business proper runs on equipment sales and service/support contracts from our paying customers, distributors and official resellers.  We will sometimes enter into direct contact with a non-paying user if we suspect they have uncovered a bug in the product, because that benefits everyone, but we can't do it for general usage questions/issues. We just don't have the resource to devote free time on a one-to-one basis.  If you want to discuss formal support for your project then please call the office or drop an email to admin@aelintra.com.
 
Kind Regards

S