Outgoing Calls with Asterisk Problem


#1

Hi!

I have just started using Asterisk. I have setup Asterisk using Asterisk@Home installation on a machine with 1 internal and 1 external IP addresses. As mentioned in some tutorials I have configured Asterisk with Asterisk Management Portal (AMP). I have a VOIP account from a service provider called WebTel and even Vonage. I want users in the LAN to dial US numbers using this account. Asterisk seems to be working fine in the LAN but we are not able to make any outgoing calls. I have tried connecting directly to the WebTel/Vonage servers without using Asterisk (over our gateway with NAT enabled) and was able to do that. So I am sure its not the problems with the accounts. Attached is my WebTel and Vonage Configuration and The log I have got from WebTel. Can any one pls help me decipher the problem??

Configuration

[200]
username=200
type=friend
secret=passwd
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="First Name Last Name" <200>

[vonage]
username=16302425663
type=peer
secret=passwd
host=sphone.vopr.vonage.net

[webtel]
username=9990030285
type=peer
secret=passwd
qualify=yes
insecure=very
host=webtel IP
fromuser=9990030285
fromdomain=WebTel IP
dtmfmode=inband
disallow=all
callerid=9990030285
authuser=9990030285
allow=ulaw

Log
The following is the log that I got when I dialed a number thru WebTel Account

Sep 2 17:13:58 DEBUG[2042]: Setting NAT on RTP to 0
Sep 2 17:13:58 DEBUG[2042]: Stopping retransmission on 'C2D07F7E-A83C-4841-9D8B-3D592856F53D@192.168.0.143' of Response 9865: Found
Sep 2 17:13:58 DEBUG[2042]: Setting NAT on RTP to 0
Sep 2 17:13:58 DEBUG[2042]: Check for res for 200
Sep 2 17:13:58 DEBUG[2042]: Call from user '200' is 1 out of 0
Sep 2 17:13:58 DEBUG[2042]: build_route: Contact hop: 
Sep 2 17:13:58 VERBOSE[2042]: -- Executing Macro("SIP/200-71fa", "dialout-trunk|2|18002232250|") in new stack
Sep 2 17:13:58 DEBUG[2042]: Expression is '1'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "1?3:2)") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-dialout-trunk,s,3)
Sep 2 17:13:58 VERBOSE[2042]: -- Executing Macro("SIP/200-71fa", "record-enable|200|OUT") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "0 > 0?2:4") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-record-enable,s,4)
Sep 2 17:13:58 DEBUG[2042]: Expression is '1'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "1?5:8") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-record-enable,s,5)
Sep 2 17:13:58 VERBOSE[2042]: -- Executing DBget("SIP/200-71fa", "RecEnable=RECORD-OUT/200") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- DBget: varname=RecEnable, family=RECORD-OUT, key=200
Sep 2 17:13:58 DEBUG[2042]: Unable to find key '200' in family 'RECORD-OUT'
Sep 2 17:13:58 VERBOSE[2042]: -- DBget: Value not found in database.
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetVar("SIP/200-71fa", "CALLFILENAME=OUT200-20050902-171358-1125695638.8") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing Goto("SIP/200-71fa", "s|14") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-record-enable,s,14)
Sep 2 17:13:58 DEBUG[2042]: Expression is '0'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "0?15:99") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-record-enable,s,99)
Sep 2 17:13:58 VERBOSE[2042]: -- Executing NoOp("SIP/200-71fa", "NO RECORDING NEEDED") in new stack
Sep 2 17:13:58 DEBUG[2042]: Expression is '1'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "1?7") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Goto (macro-dialout-trunk,s,7)
Sep 2 17:13:58 DEBUG[2042]: Expression is '0'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "0?9") in new stack
Sep 2 17:13:58 DEBUG[2042]: Not taking any branch
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetCallerID("SIP/200-71fa", "9990030285") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetGroup("SIP/200-71fa", "OUT_2") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing CheckGroup("SIP/200-71fa", "1") in new stack
Sep 2 17:13:58 DEBUG[2042]: Avoiding initial deadlock for 'SIP/200-71fa'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetVar("SIP/200-71fa", "DIAL_NUMBER=18002232250") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetVar("SIP/200-71fa", "DIAL_TRUNK=2") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing AGI("SIP/200-71fa", "fixlocalprefix") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Sep 2 17:13:58 VERBOSE[2042]: -- AGI Script fixlocalprefix completed, returning 0
Sep 2 17:13:58 VERBOSE[2042]: -- Executing SetVar("SIP/200-71fa", "OUTNUM=18002232250") in new stack
Sep 2 17:13:58 VERBOSE[2042]: -- Executing Cut("SIP/200-71fa", "custom=OUT_2|:|1") in new stack
Sep 2 17:13:58 DEBUG[2042]: Expression is '0'
Sep 2 17:13:58 VERBOSE[2042]: -- Executing GotoIf("SIP/200-71fa", "0?19") in new stack
Sep 2 17:13:58 DEBUG[2042]: Not taking any branch
Sep 2 17:13:58 VERBOSE[2042]: -- Executing Dial("SIP/200-71fa", "SIP/webtel/18002232250") in new stack
Sep 2 17:13:58 DEBUG[2042]: Setting NAT on RTP to 0
Sep 2 17:13:58 DEBUG[2042]: Outgoing Call for 18002232250
Sep 2 17:13:58 DEBUG[2042]: 18002232250 is not a local user
Sep 2 17:13:58 VERBOSE[2042]: -- Called webtel/18002232250
Sep 2 17:13:59 DEBUG[2042]: Acked pending invite 102
Sep 2 17:13:59 DEBUG[2042]: Stopping retransmission on '03f9306e59668dd57eb891f3476c1f35@203.145.190.170' of Request 102: Found
Sep 2 17:13:59 DEBUG[2042]: Acked pending invite 103
Sep 2 17:13:59 DEBUG[2042]: Stopping retransmission on '03f9306e59668dd57eb891f3476c1f35@203.145.190.170' of Request 103: Found
Sep 2 17:13:59 DEBUG[2042]: Acked pending invite 104
Sep 2 17:13:59 DEBUG[2042]: Stopping retransmission on '03f9306e59668dd57eb891f3476c1f35@203.145.190.170' of Request 104: Found
Sep 2 17:13:59 NOTICE[2042]: Failed to authenticate on INVITE to '"9990030285" ;tag=as7117921e'
Sep 2 17:14:03 DEBUG[2042]: Auto destroying call '2416af8144fd7fb64f122cb2149f7e1b@203.145.190.170'
Sep 2 17:14:03 DEBUG[2042]: Auto destroying call '2416af8144fd7fb64f122cb2149f7e1b@203.145.190.170'

I think its able to find the server and register with it…but its not able to dial the call…any ideas why? Thanks in advance!!


#2

Looks like it dialed here Sep 2 17:13:58 VERBOSE[2042]: – Called webtel/18002232250

Then Failed here
Sep 2 17:13:59 NOTICE[2042]: Failed to authenticate on INVITE to '“9990030285” ;tag=as7117921e’
Sep 2 17:14:03 DEBUG[2042]: Auto destroying call '2416af8144fd7fb64f122cb2149f7e1b@203.145.190.170’
Sep 2 17:14:03 DEBUG[2042]: Auto destroying call ‘2416af8144fd7fb64f122cb2149f7e1b@203.145.190.170’

Sorry I cant be more helpful I do not know much about sip