How to connect Asterisk to a Voip Provider? Phonemaxpr

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How to connect Asterisk to a Voip Provider? Phonemaxpr

Postby jcalderin » Wed Dec 30, 2009 9:10 pm

Currently installed the 1.5.0 version of AsteriskNOW with several phone extensions in my company, everything works fine on the local network but when trying to connect to my voip provider (Phonemaxpr) to make calls to the outside world can not do. He says: Your call can not be completed as dialed, please check the number and dial again.

This is the current configuration of my Trunk allegedly is connected to my voip provider "phonemaxpr.com"

Outgoing Settings

Trunk Name: Phonemaxpr

PEER Details:

allow=ulaw&gsm
canreinvite=no
context=from-phonemaxpr
disallow=all
fromuser=Myusername
host=phonemaxpr.com
nat=yes
qualify=yes
secret=mypassword
sendrpid=yes
trustrpid=yes
type=friend


Incoming Settings

Nothing


Registration

myusername:mypassword@phonemaxpr.com

This is what happens in the Asterisk terminal when I try to make a call to the outside world:

-- Executing [7872252930@from-internal:1] ResetCDR("SIP/200-b7804d20", "") in new stack
-- Executing [7872252930@from-internal:2] NoCDR("SIP/200-b7804d20", "") in new stack
-- Executing [7872252930@from-internal:3] Wait("SIP/200-b7804d20", "1") in new stack
-- Executing [7872252930@from-internal:4] Playback("SIP/200-b7804d20", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
-- <SIP/200-b7804d20> Playing 'silence/1' (language 'en')
-- <SIP/200-b7804d20> Playing 'cannot-complete-as-dialed' (language 'en')
-- <SIP/200-b7804d20> Playing 'check-number-dial-again' (language 'en')
-- Executing [7872252930@from-internal:5] Wait("SIP/200-b7804d20", "1") in new stack
-- Executing [7872252930@from-internal:6] Congestion("SIP/200-b7804d20", "20") in new stack
== Spawn extension (from-internal, 7872252930, 6) exited non-zero on 'SIP/200-b7804d20'
-- Executing [h@from-internal:1] Macro("SIP/200-b7804d20", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/200-b7804d20", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/200-b7804d20", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/200-b7804d20", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/200-b7804d20", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/200-b7804d20' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-b7804d20'
localhost*CLI>


My voip service provider only gives me these parameters so i can use in ipsoftphone or hardphones:

User Name: My user name

Password: My Password

Proxy: 66.50.240.46

Proxy Port: 5060

Domain Name: phonemaxpr.com

And finally my Voip provider says check this option in the settings: "There is a firewall / NAT between my PC and this proxy".


Any suggestions to make my Asterisk server work????

Thanks!!
Last edited by jcalderin on Wed Dec 30, 2009 10:09 pm, edited 2 times in total.
jcalderin
Newsterisk
 
Posts: 2
Joined: Wed Dec 30, 2009 7:51 pm

Re: How to connect Asterisk to a Voip Provider? Phonemaxpr

Postby jcalderin » Wed Dec 30, 2009 10:00 pm

Name/username Host Dyn Nat ACL Port Status
phonemaxpr/myuser xx.xx.xxx.xx N 5060 OK (79 ms)
200/200 xxx.xxx.x.xxx D N A 32904 OK (105 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
jcalderin
Newsterisk
 
Posts: 2
Joined: Wed Dec 30, 2009 7:51 pm


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