Call failed using Asterisk

0
votes

I am trying on 2 PCs, First acts as sip server & client (has asterisk & twinkle installed) and the other as a client only(has twinkle only installed) . I try to make a call between them using Ethernet cable -No internet- so I established the wired connection and I gave each of them an address , I gave the 1st one with asterisk installed ip 192.168.0.1 & the 2nd one 192.168.0.2. Fist one has username 100 & 2nd one is 101.

I wrote in terminal "sudo asterisk -rvvvvvvv" then "sip reload" then "dialplan reload" then "sip set debug on" & I establish the wired connection so I find the 2 Twinkles on the 2 Pcs registered successfully to asterisk but when I try to make a call between them , Twinkle said that " call failed 404 not found " i think then the problem is in the extensions.conf but i can not figure out what is wrong, any one can help me .?
Here is the full output on terminal mediafire.com/?6g0uuhkai5vcahk

Also tell me if there is anything wrong in the file as I am beginner.

Thanks in advance Here are the configuration files:

sip.conf

[general]

bindport=5060

udpbindaddr=192.168.0.1:5060

allowguest=yes

disallow=all

allow=gsm

delayreject=yes

nochecksums=no

pedantic=no

srvlookup=yes

autodomain=yes

sipdebug = yes

domain=192.168.0.1

nat=no

notifyringing=yes

notifyhold=yes

register => 100:[email protected]/internal-phones

register => 101:[email protected]/internal-phones

peer auth=100:[email protected]

peer auth=101:[email protected]

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[192.168.0.1]

usereqphone = yes

nat=no

fromdomain=192.168.0.1

fromuser=100

secret=sarasara

username=100

context=internal-phones

authname=100

dtmfmode = rfc2833

canreinvite=yes

notifyringing=yes

notifyhold=yes

peer auth=100:[email protected]

peer auth=101:[email protected]

disallow=all

allow=gsm

[100]

type=friend

context=internal-phones

secret=sarasara

nat=no

qualify=no

host=dynamic

dtmfmode = rfc2833

permit=192.168.0.1

[101]

type=friend

context=internal-phones

secret=saadsaad

qualify=no

host=dynamic

nat=no

dtmfmode = rfc2833

permit=192.168.0.1

extensions.conf

[globals]

[general]

exten => 100,1,Dial(SIP/100,60)

exten => 101,1,Dial(SIP/101,60)

exten => s,1,hangup

[internal-phones] exten => 100,1,Dial(SIP/100,60)

exten => 101,1,Dial(SIP/101,60)

exten => s,1,hangup

1
configurationsipasteriskethernetsip-server
user "asterisk -r" to see what it actualy do.If nothing shown,try also use sip set debug on/core set verbose 5 on that console. There are no any way answer your question based on info provided. – arheops
I wrote in terminal "sudo asterisk -rvvvvvvv" then "sip reload" then "dialplan reload" then "sip set debug on" & I establish the wired connection so I find the 2 Twinkles on the 2 Pcs registered to asterisk but when I try to make a call between them , Twinkle said that " call failed 404 not found " Here is the full output on terminal mediafire.com/?6g0uuhkai5vcahk – Sarsoura
sorry, this site is not asterisk mailing list nor some debug source. very likly you missed something with context. If you not able read books, you have hire expert. – arheops

1 Answers

0
votes

Enable SIP debugging (sip set debug on) to view the SIP response for the call. (503, 403 or 404 ?). Also, make sure your soft-phone is listening on an udp port other than 5060, which will conflict with asterisk.