# Basic call on Asterisk **URL:** \ **Category:** Asterisk Support\ **Created:** [September 13, 2013, 11:49pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906 "2013-09-13T23:49:51Z")\ **Posts on this page:** 8\ **Page:** 1
**Author:** ![eng\_asa](https://avatars.discourse-cdn.com/v4/letter/e/54ee81/32.png) [@eng\_asa](https://community.asterisk.org/u/eng_asa)\ **Post date:** [September 13, 2013, 11:49pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/1 "2013-09-13T23:49:51Z")
Hello, I have just installed Asterisk on my ubuntu server and trying to get it running I have defined two accounts in sip.conf and now want to do a basic successful call using X-lite sip phone. The registration is fine, the ringing is OK and the call in general looks normal but still no voice can be heard and lead the call to be disconnected after!.. checking the logs chan\_sip.c:3641 retrans\_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) I have a very simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=/255.255.255.0 [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal What could be the cause for this error? One more question, I can see in the configuration an option to add a registrar using the data of SIP service provider, but what if I want my server to be the service provider… should I add this part in my configuration and why should we need another SIP provider? Thanks. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 14, 2013, 2:12pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/2 "2013-09-14T14:12:22Z")
Asterisk is probably not receiving the ACK to the 200 OK for the incoming call. A SIP trace would confirm this. That tends to suggest a firewall or NAT problem. Note that you have localnet without any NAT settings (nat= is not one). If you really need local net, you also need one of externip; externhost, or stunaddr. ---
**Author:** ![eng\_asa](https://avatars.discourse-cdn.com/v4/letter/e/54ee81/32.png) [@eng\_asa](https://community.asterisk.org/u/eng_asa)\ **Post date:** [September 14, 2013, 2:57pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/3 "2013-09-14T14:57:39Z")
Thanks for your reply. So what can I do to overcome the NAT issue? My target is to have some running internal calls, and I thought adding a localnet with the machine IP would do that! I have added externip to my configuration and still doesn’t work? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 14, 2013, 3:33pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/4 "2013-09-14T15:33:35Z")
You only need localnet if you have also defined a public address and the the local network is not the one directly connected to the local interface. Fixing NAT (and firewall) problems needs a level of detail of your configuration that you haven’t provided. Normally, if you provide enough information, the problem should be obvous to you. ---
**Author:** ![eng\_asa](https://avatars.discourse-cdn.com/v4/letter/e/54ee81/32.png) [@eng\_asa](https://community.asterisk.org/u/eng_asa)\ **Post date:** [September 14, 2013, 10:45pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/5 "2013-09-14T22:45:07Z")
I have taken a network dump capture to my call, and I can see invite,OK and ACK Indeed the connection is established and the ringing done but still the voice packet can’t be reached some how! What are the other configuration I can look at to find the issue? The captured packets for my call converted to text can be found here [fileconvoy.com/dfl.php?id=g9 … 0c4a1b48ba](http://www.fileconvoy.com/dfl.php?id=g9e3530bcc01162be99937113465bc570c4a1b48ba) Thanks. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 15, 2013, 10:04am UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/6 "2013-09-15T10:04:13Z")
There are no ACKs to the 200 OKs. What looks funny is that both X-Lite and Asterisk have the same public IP address, I would expect them both to have a private address. I suspect the NAT configuration on the X-Lite is broken, and that there are also issues with the router. ---
**Author:** ![eng\_asa](https://avatars.discourse-cdn.com/v4/letter/e/54ee81/32.png) [@eng\_asa](https://community.asterisk.org/u/eng_asa)\ **Post date:** [September 15, 2013, 12:07pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/7 "2013-09-15T12:07:19Z")
The Asterisk server is on a different network than the xlite client which is installed on my laptop, so I am using the public IP in both to connect them together. I tried xlite from different laptops and networks, do you really think the problem is in the client site not the server? Is there any other configuration files I can check on the server side? Thanks. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 15, 2013, 12:12pm UTC](https://community.asterisk.org/t/basic-call-on-asterisk/43906/8 "2013-09-15T12:12:31Z")
They shouldn’t have the same IP address in that case. You need to work out which IP addresess they should have and fix the one that it is using the wrong one.