[RESOLVED] Inbound Calls Not Ringing Phones

Asterisk Asterisk Support
1

Hi All,

Im having some trouble in that inbound calls arnt ringing the sip extensions, I suspect it’s my own configuration as I can make outbound calls fine. If someone could shed some light it would be greatly appreciated. I have include my sip.conf and extensions.conf for reference which private informtion *'ed out.

UPDATE: Just spoke to our IPDC provider, apparently we need to send the invite request twice as the first is always rejected and the second is accepted, any ideas?

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = others
srvlookup=yes
register => ****:****@****/****

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[ext-sip-account]
type=friend
context=from-voip-provider
username=0001*****
fromuser=0001*****
secret=****
host=****
fromdomain=****
qualify=yes
insecure=very
nat=yes
callerid=unknown

extensions.conf

[others]

[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Answer()
exten => 2001,3,Playback(hello-world)
exten => 2001,4,Hangup()
exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account)
include => from-voip-provider

[from-voip-provider]
exten =>01453700060,1,Dial(SIP/2000)

Any help anyone can provide would be great!

2

How about

[my-phones] exten => s,1,Goto(my-phones,2000,1) exten => 2000,1,Dial(SIP/2000) exten => 2001,1,Answer() exten => 2001,3,Playback(hello-world) exten => 2001,4,Hangup() exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account) include => from-voip-provider

3

Unfortunately I’ve tried that, I think it might have something to do with the update i posted

4

It is standard to send invite without authentication followed with invite with authentication, the first one being rejected as unuathenticated. In other cases, there is no requirement to continue after a rejection.

How is the first invite being rejected?

5

Ok well in that case a second invite isn’t being sent, at least I can’t see one in the sip trace

6

Hi All,

Just an update with some more information i’ve been able to get hold of, I still can’t get this working im afraid.

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = others
register => 0001XXXXXXXXXXXX985:BGIxK1fXXXXXXXXXXXX1Br6X6V1@gw3.theiptele.com/2000

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

[ext-sip-account]
type=friend
context=from-voip-provider
username=0001XXXXXXXXXX85
secret=BGIxK1fCXXXXXXXXXXXXr6X6V1
host=88.215.60.40
fromuser=000XXXXXXXX3985
fromdomain=87.194.166.89
qualify=yes
insecure=very
nat=no
canreinvite=yes
dtmfmode=rfc2883

extensions.conf

[others]

[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account)

[from-voip-provider]
exten => 01453700060,1,Dial(SIP/2000)

Info from VOIP provider

  1.  Gamma Telecom IPDC SIP gateway: gw3.theiptele.com 
       Please note that IPDC authentication is based on User authentication, the solution does not require the CPE's IP Address.

  2.  The SIP header requirements in the INVITE packets originated from the CPE should be set as follows: 

        -  SDP payload must be present, 

        -  From: header must contain public facing interface, originating CLI in the display name field and SIP username in the userpart (e.g.: "From: 'CLI_number' 

          'SIP_username'@'public_facing_interface_IP_address'>") The SIP username is the same as the extension number mentioned above,

        -  To: header must contain IPDC SIP gateway address or domain and the called number with leading zero or in E164 format. 

 3.  The CLI numbers are validated against the Gamma assigned geo number ranges for each endpoint and will be accepted in the following formats:

        -  "+44"+area_code+phone_number

        -  "0"+area_code+phone_number

        -  country_code+area_code+phone_number

        These formats can be presented in:

        -  the PAI (P-Asserted ID) header of the SIP Invite message

        -  the R-PID (Remote Party ID) header of the SIP Invite message

        -  the From header of the SIP Invite message

        (Please note that we do not support P-Preferred ID headers for CLI presentation.)

 4.  The Traffic on the following ports must be forwarded through relevant routers and firewalls on the customer premises to allow access to the Gamma IP DirectConnect Network

      TCP and UDP 5060 egress/ingress to gw3.theiptele.com 

      UDP all ports between 1024 - 65535 egress/ingress to gw3.theiptele.com
7

This has now been resolved, I’ve included my set-up just in case anyone stumbles across this with a similar issue.

sip.conf

[code][general]
context=others
allowoverlap=no
port=5060
bindaddr=0.0.0.0
srvlookup=yes
externip=XX.XX.XX.XX
localnet=192.168.0.0/255.255.255.0
allowexternal=yes
allowguest=yes
register => 000157XXXXXXXXX85:BGIxXXXXXXxxxxxxV1@gwXXXXXe.com/0001XXXXXXXXXX85

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
qualify=yes
nat=yes

[ext-sip-account]
type=friend
host=gw3.theiptele.com
username=0001xxxxxxxxxxx85
fromuser=01453xxxxx0
fromdomain=xx.xx.xx.xx
secret=BGxxxxxxxxxxrxxV1
context=from-voip-provider
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
insecure=invite
nat=yes
[/code]

extensions.conf

[code][others]

[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account)

[from-voip-provider]
exten => 441453XXXXXX0,1,Dial(SIP/01453XXXXXX@ext-sip-account)
[/code]