# Asterisk 1.4 calling voip providers

**URL:** <https://community.asterisk.org/t/asterisk-1-4-calling-voip-providers/13459>\
**Category:** Asterisk Support\
**Created:** [March 30, 2007, 1:00am UTC](https://community.asterisk.org/t/asterisk-1-4-calling-voip-providers/13459 "2007-03-30T01:00:31Z")\
**Posts on this page:** 2\
**Page:** 1

<div class="post-metadata">

**Author:** ![cenetadmin1](https://avatars.discourse-cdn.com/v4/letter/c/f4b2a3/32.png) [@cenetadmin1](https://community.asterisk.org/u/cenetadmin1)\
**Post date:** [March 30, 2007, 1:00am UTC](https://community.asterisk.org/t/asterisk-1-4-calling-voip-providers/13459/1 "2007-03-30T01:00:31Z")

</div>

I have installed 1.4.2 and when I attempt to call using a voip provider sip, I can not hear the other person or answering machine. An IAX2 provider appears to work

What have I done wrong in my configuration.

Here are the messages from the console

```
-- Executing [710015732041111@numberplan-custom-1:1] Dial("SIP/2001-082e57d8", "SIP/trunk_1/0015732041111") in new stack
-- Called trunk_1/0015732041111
-- Call on SIP/trunk_1-082e6d58 left from hold
-- SIP/trunk_1-082e6d58 is making progress passing it to SIP/2001-082e57d8
-- Call on SIP/trunk_1-082e6d58 left from hold
-- SIP/trunk_1-082e6d58 answered SIP/2001-082e57d8
-- Native bridging SIP/2001-082e57d8 and SIP/trunk_1-082e6d58

```

How do I setup a stun server, since the asterisk box is behind a firewall.

Thanks

Eric

---

<div class="post-metadata">

**Author:** ![zmanea](https://avatars.discourse-cdn.com/v4/letter/z/8dc957/32.png) [@zmanea](https://community.asterisk.org/u/zmanea)\
**Post date:** [April 1, 2007, 4:34pm UTC](https://community.asterisk.org/t/asterisk-1-4-calling-voip-providers/13459/2 "2007-04-01T16:34:11Z")

</div>

Stun should not be required if your asterisk box is using a static IP. Make sure to use externip in your sip.conf and forward the following ports to your asterisk box.

UDP 5060 - 5061  
UDP 10000 - 20000