Configuration of outbound Calls in sip.conf

[sdf123]

Hi,

I am using asterisk 1.8 and I have to configure outbound call using sip account. I have to configure seaprate IPs for voice and Signalling.

voice IP is 10.XXX.XX.142 and signalling IP is 10.XXX.XX.150

I have make configuration is sip.conf like this:
[general]
context=default ; Default context for incoming calls
media_address=10.XXX.XX.142
externaddr=10.XXX.XX.150
nat=force_rport,comedia

And output of route -n is
Destination Gateway Genmask Flags Metric Ref Use Iface
10.XXX.XX.180 10.XXX.XX.150 255.255.255.255 UGH 0 0 0 eth1
0.0.0.0 10.XXX.XX.142 0.0.0.0 UG 0 0 0 eth0

But when I trying to test an outbound call using originate command its not working.Pls guide where I am making mistake.
originate SIP/10.XXX.XX.142/03430XXXXXX application playback vm-goodbye
originate SIP/10.XXX.XX.150/03430XXXXXX application playback vm-goodbye

Logs are below:
== Using SIP RTP CoS mark 5
[Jul 2 14:14:38] WARNING[23561]: chan_sip.c:3821 retrans_pkt: Retransmission timeout reached on transmission 2fb3ad7d558c2d85619519e3651a4ff7@10.XXX.XX.137:5060 for seqno 102 (Critical Request) -- See ........ ... nsmissions
Packet timed out after 32000ms with no response


Many Thanks.

[sdf123]

Pls guide.

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) WHY do you believe you have to do this? I'd love to see the instructions that told you that you should do so.

4) In most cases, this is handled automatically by the Carrier and your system need only accept the inbound traffic from the "unexpected" IP address, which generally means allowing that traffic through your firewall. No other changes are usually needed as asterisk is (by default) configured to negotiate the audio channel with the remote account.

5) Vicidial is Dialer software. You appear to have either a simple misunderstanding of how to connect to a telco OR a seriously technical configuration issue entirely unrelated to Vicidial.

If the latter is the case, then you should be on an Asterisk forum or a forum somehow related to the type of connection you are trying to make. Note that Asterisk forums have a significantly larger base than Vicidial (ie: more help!). Once you get Asterisk to generate that call, Vicidial can handle the call and use it for dialing purposes.

Happy Hunting!