CODEWITHSUNDEEP
linuxasterisktelecommunication
Ashwin Parmar
Ashwin Parmar

Reputation: 3045

How to update my Asterisk Dial-Plan for combination of Telephone Number?

I am trying to create dialplan for incoming/outgoing for given numbers:

+xx xxx [xxxxxxxxx|xxxxxxxx]

I have already configure my service provider information in sip.conf

[sipprovider]
type=friend
secret=xxxxx
defaultusername=xxxxx
host=xxx.xx.xx.xxx
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
insecure=port,invite
;fromdomain=xxx.xx.xx.xxx
context=default
nat=yes

Now, I want to create incoming/outgoing trunk, my extensions allow to dial international calls and incoming call received on given number.

+xx xxx [xxxxxxxxx|xxxxxxxx]

[default]
    switch => Realtime

    exten => 55,1,Verbose(1,Echo test application)
    exten => 55,n,Dial(SIP/sipprovider/0091XXXXX99999@sipprovider); Here is the outbound call, the exact dialstring depends on outgoing provider and channeltype
    exten => 55,n,Hangup()

Display: Calling.... and then, VM Play: Person you are calling is unavailable

Asterisk Console:

== Using SIP RTP CoS mark 5
    -- Executing [55@default:1] Verbose("SIP/3001-00000029", "1,Echo test application") in new stack
 Echo test application
    -- Executing [55@default:2] Dial("SIP/3001-00000029", "SIP/sipprovider/0091XXXXX99999@sipprovider") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/sipprovider/0091XXXXX99999@sipprovider
[Aug 17 18:29:02] WARNING[32467]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 17 18:29:02] WARNING[32467]: chan_sip.c:4053 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [55@default:3] Hangup("SIP/3001-00000029", "") in new stack
  == Spawn extension (default, 55, 3) exited non-zero on 'SIP/3001-00000029'
    -- Executing [h@default:1] Verbose("SIP/3001-00000029", "Hangup...") in new stack
Hangup...

Upvotes: 0

Views: 404

Answers (1)

Sergey S.
Sergey S.

Reputation: 181

Basically a dialstring can be in 'SIP/devicename/extension' or 'SIP/username@host' format. SIP/sipprovider/0091XXXXX99999@sipprovider is wrong.

"Retransmission timeout reached" means that asterisk tries to send an INVITE to sipprovider, but sipprovider's SIP port (5060 UDP) isn't accessible. You can see this in SIP debug.

Upvotes: 1

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