# Need Help With Outbound Calls

**URL:** <https://community.asterisk.org/t/need-help-with-outbound-calls/32251>\
**Category:** Asterisk Support\
**Created:** [August 14, 2010, 3:04am UTC](https://community.asterisk.org/t/need-help-with-outbound-calls/32251 "2010-08-14T03:04:33Z")\
**Posts on this page:** 2\
**Page:** 1

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**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\
**Post date:** [August 14, 2010, 3:04am UTC](https://community.asterisk.org/t/need-help-with-outbound-calls/32251/1 "2010-08-14T03:04:33Z")

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Frustration is starting to set in, so after digging through these forums and searching the internet for the past couple of hours, I decided to post here for some help.

I have installed and configured Asterisk 1.6. I was able to get a softphone as well as a PAP2 ATA to register its ports. In addition, I have one SIP trunk registered (Viatalk is the provider). At one point I was able to make outbound calls from any of the extensions, but I wasn’t able to receive inbound calls.

While working on my configurations, I seem to have fixed the inbound issue, but now I am not able to make outbound calls from any extension. I can’t seem to figure out what changed that caused it to stop working. My softphone gives me a message that says “Server is Unavailable”. This is what I am seeing on the Asterisk screen:

[quote] Using SIP RTP CoS mark 5  
– Executing [248xxxxxxx@default:1] Goto(“SIP/150-00000007”, “1248xxxxxxxx,1”) in new stack  
– Goto (default,1248xxxxxxx,1)  
– Executing [1248xxxxxxx@default:1] Dial(“SIP/150-00000007”, “SIP/1248xxxxxxx@viatalk,60,r”) in new stack  
== Using SIP RTP CoS mark 5  
== Everyone is busy/congested at this time (1:0/0/1)  
– Executing [1248xxxxxxx@default:2] PlayTones(“SIP/150-00000007”, “480+620/250,0/250”) in new stack  
– Executing [1248xxxxxxx@default:3] Congestion(“SIP/150-00000007”, “”) in new stack  
== Spawn extension (default, 1248xxxxxxx, 3) exited non-zero on ‘SIP/150-00000007’[/quote]

Here are my outbound rules from the extensions.conf file:

[quote][outgoing]  
exten =\> 911,1,Dial(SIP/911@${TRUNK},60,r)  
exten =\> 411,1,Dial(SIP/411@${TRUNK},60,r)  
exten =\> \*123,1,Dial(SIP/\*123@${TRUNK},60,r)

exten =\> \_NXXXXXX,1,Goto(1${NPX}${EXTEN},1) ; if dialing 7 digits, prepend 1 + Area Code  
exten =\> \_NXXNXXXXXX,1,Goto(1${EXTEN},1) ; if dialing 10 digits, prepend 1

exten =\> \_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@${TRUNK},60,r)  
exten =\> \_1NXXNXXXXXX,2,Playtones(480+620/250,0/250) ; play a fast busy (reorder) tone  
exten =\> \_1NXXNXXXXXX,3,Congestion

; For International dialing [Optional]  
exten =\> \_011X.,1,Dial(SIP/${EXTEN}@${TRUNK},60,r) [/quote]

I can’t seem to figure out where to go next. My outbound dialing rules came from Viatalk’s site, and as I mentioned, were working at one time. I didn’t touch any of the rules, so I don’t think they are the issue.

Any help would be greatly appreciated.

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<div class="post-metadata">

**Author:** ![rbreidenstein](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/rbreidenstein/32/477_2.png) [@rbreidenstein](https://community.asterisk.org/u/rbreidenstein)\
**Post date:** [August 14, 2010, 9:42pm UTC](https://community.asterisk.org/t/need-help-with-outbound-calls/32251/2 "2010-08-14T21:42:57Z")

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What does your SIP entry in sip.conf look like for [viatalk]?