Cant get calls to go through my SIP trunk

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Cant get calls to go through my SIP trunk

Postby computergroove » Thu Apr 24, 2008 7:43 am

When I use one of the agent machines to make an outbound call I get an error. I am using putty to get the output. I used the vi editor to view the screenlog.0 and I used copy and paste through putty to get this:

My CLI Output:

Apr 23 10:20:54 NOTICE[2447]: chan_sip.c:5508 sip_reg_timeout: -- Registration for '3174@adamstelecom.ath.cx' timed out, trying again (Attempt #2581)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
Apr 23 10:21:08 NOTICE[2447]: chan_sip.c:5508 sip_reg_timeout: -- Registration for '3174@adamstelecom.ath.cx' timed out, trying again (Attempt #2582)
Apr 23 10:21:14 NOTICE[2447]: chan_sip.c:5508 sip_reg_timeout: -- Registration for '3174@adamstelecom.ath.cx' timed out, trying again (Attempt #2582)
Apr 23 10:21:14 NOTICE[2447]: chan_sip.c:5508 sip_reg_timeout: -- Registration for '3174@adamstelecom.ath.cx' timed out, trying again (Attempt #2582)
Apr 23 10:21:14 NOTICE[2447]: chan_sip.c:5508 sip_reg_timeout: -- Registration for '3174@adamstelecom.ath.cx' timed out, trying again (Attempt #2582)

My Sip.conf File:

[general]
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)
context = default
disallow=all
allow=g729 ; Yes I have registered the g729 codec
allow=ulaw
allow=alaw

register => 1234:secret@adamstelecom.ath.cx ; I can dial 4 calls at once
register => 1234:secret@adamstelecom.ath.cx ; with this SIP provider but
register => 1234:secret@adamstelecom.ath.cx ; they only gave me 1
register => 1234:secret@adamstelecom.ath.cx ; phone number

[authentication]
[adamstelecom]
type=peer
host=adamstelecom.ath.cx
username=1234
secret=secret
fromuser=1234
fromdomain=adamstelecom.ath.cx
context=default
insecure=very
canreinvite=no
disallow=all
allow=g729
dtmfmode=rfc2833

[1001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1001
secret=1234
host=dynamic
dtmfmode=rfc2833
canreinvite=no

[1002]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1002
secret=1234
host=dynamic
dtmfmode=rfc2833
canreinvite=no

[1003]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1003
secret=1234
host=dynamic
dtmfmode=rfc2833
canreinvite=no

[1004]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1004
secret=1234
host=dynamic
dtmfmode=rfc2833
canreinvite=no

Obviously I am using SIP. I read somewhere that SIP uses a different port for each number being used. I only have port 5060 forwarded to my server. Do I need to add a port range or do I need to add :5060 after my register => line in my SIP.conf? I havent done much with the extensions.conf so I am guessing that I need to change something in that file to make this work. Also I am unsure if I am going about using 1 SIP provider account for 4 concurrent calls right in the register => line in sip.conf. Please advise.
computergroove
 
Posts: 59
Joined: Sat Dec 29, 2007 5:24 pm

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