Asterisk Upgrade
Asterisk users can now use our new media gateways. Our goal is to provide customers
with the highest quality network. This includes voice quality as well as redundancy.

We currently have new gateways in San Francisco, Tampa Bay, and Washington DC.
We are planning to add new gateways in New York, Boston, Chicago, Dallas, Los Angeles, Salt Lake City, and Seattle.

The new DNS entries are:
sfo.telasip.com
tpa.telasip.com
iad.telasip.com

We have also added the DNS Entries for the future gateways:
jfk.telasip.com
bos.telasip.com
ord.telasip.com
slc.telasip.com
lax.telasip.com
sea.telasip.com

To begin using the new gateways you can create an additional outbound peer with the
same settings you have now. Except the change the host and fromdomain setting to the DNS entry of the nearest gateway.

For Example Add:

[telasip-sfo]
type=peer
host=sfo.telasip.com
username=jdoe
secret=jdoepass
qualify=yes
fromuser=jdoe
fromdomain=sfo.telasip.com
canreinvite=no
sendrpid=yes
insecure=very
context=from-trunk
disallow=all
allow=ulaw
allow=gsm
;allow=g729

Once you are satisfied with the quality of the new gateway, you can register with the
new gateway, then request support to have your inbound number routed to that
gateway.

** New customers that are registering with gw3.sip.telasip.com have a slightly different setup. You can do the same as above but your DNS entries are as follows.
The new DNS entries are:

sip.sfo.telasip.com
sip.tpa.telasip.com
sip.iad.telasip.com

We have also added the DNS Entries for the future gateways:
sip.jfk.telasip.com
sip.bos.telasip.com
sip.ord.telasip.com
sip.slc.telasip.com
sip.lax.telasip.com
sip.sea.telasip.com

Configuring Asterisk Voip SIP.

Configuring Asterisk SIP Trunk for use with Telasip is very straight forward. However, configuring the SIP trunk is not enough to get it to work. With Asterisk there can be many things that cause calls to fail. Luckily Asterisk provides you with everything you need to debug the configuration.

When you signed up for service with Telasip you were sent an email with your account details. This email contained 3 important pieces of information.

[telasip-gw]
type=peer
host=gw3.sip.telasip.com
username=jdoe
secret=jdoepass
qualify=yes
fromuser=jdoe
fromdomain=gw3.sip.telasip.com
canreinvite=no
sendrpid=yes
insecure=very
context=from-trunk
disallow=all
allow=ulaw
allow=gsm
;allow=g729

Notice the insecure=very and context=from-trunk. These are actually the inbound
settings needed for Telasip. Typically we have found it is better to create one trunk
and put both the inbound and outbound settings in that one peer definition.

The reason for this is Asterisk will look up users by name, and then try peer by
IP address. Because of the way we route the incoming DID, Asterisk will not find
us by name. It will then search for peer by IP Address, this also creates an issue
because we originate and terminate using the same Servers / IP address.

So how are we doing? To see how we are doing login to Asterisk by typing
asterisk -r at the linux prompt. Issue the following 2 commands

CLI> sip show registry
Host                               Username     Refresh State
jfk.sip.telasip.com:5060   jdoe           585   Registered

CLI> sip show peers
telasip-gw/jdoe     4.79.19.56        5060     OK (66 ms)

This is all you need to configure the sip trunk. The sip trunk is configured properly.
So I have everything configured correctly, but it is still not working. Unfortunately
with asterisk tyou must also create Dial plan entires or extensions in extensions.conf.
You will need an inbound context and an outbound context. It is important to
keep them seperate. Because of the flexibility of asterisk we can only provide
one possible example.

[from-trunk]
exten=>_2403961450,1,Dial(SIP/201,30,r)

[from-internal]
exten=>_NXXNXXXXXX,1,Dial(SIP/telasip-gw/${EXTEN},,r)


What can I do next?

Can you place an outbound call?
Can you receive an inbound call?
Are the calls completing but you can not hear the caller?

Aasterisk provides you with everything you need to debug these issues. All you
need to do is log in to the server, and attach to asterisk with the command
asterisk -r. This will start the Command Line Interface (Console) for asterisk.
Once in asterisk a few simple commands will allow you to see what is happening.

sip debug - prints the sip messages on the console
set verbose 5 - prints the dial plan commands that are executing
sip show registry - shows the current registration status
sip show peers - shows the status of all peers

TrixBox users should become familiar with the asterisk command line. There is
a wealth of information at your disposal. we have provided some trouble shooting
guides below. Just click on the Link of the problem you are experiencing

Can not make Outbound Calls
If you have successfully registered with our servers then we can assume that the
NAT issues, DNS issues, and other network related issues are resolved.
The first thing to check is the sip peer is reachable. Do a sip show peers and verify the
status is not Unreachable.

The next thing to check is the Dial statement. Make sure you are sending 10 digits
and verify that the peer matches the sip.conf file. Example Dial statement should
look like this:

Dial("SIP/tealsip-gw/2403961450|30|r)

The next thing to do is verify that the following settings are set in the
outgoing peer:

username=jdoe
secret=jdoepass
fromuser=jdoe
fromdomain=gw3.sip.telasip.com

If the call rings but then disconnects when it is answered. You are most likely having
an issue with reinvites. You can disable reinvites by setting the codec to gsm.

The last thing you can do is:

CLI> sip debug CLI> set verbose 5 Place an outobund call

Email the output to support @ telasip.com

Getting One Way Audio
It is possibe to configure Asterisk for reinvites. But first lets verify the issue is
with reinvites, and not a firewall issue. To test reinvites set the codec to gsm.

TelaSip Peer:
disallow=all
allow=gsm

If the call works then the issue is with reinvites. You can email support or try the
following to configure asterisk for reinvites.


Telasip attempts to support re-invites. Re-invites allow the audio to be redirected
from your box to our providers gatewaey. Typically this provides the best overall
call quality. But Asterisk needs to be configured for re-invites.

To support re-invites Asterisk needs to know what it's public IP address is. There
are two mechanisms to configure Asterisk for reinvites. For users with static IP
addresses simply at the following in the sip.conf file in the general section.

externip=xxx.xxx.xxx.xxx               ; where xxx.xxx.xxx.xxx is the static IP address
localnet=xxx.xxx.xxx.xxx/255.255.255.0      ; where xxx.xxx.xxx.xxx is the internal subnet.


The other mechanism requires you to use Dynamin DNS.

externhost=<yourdynamicdns>
externrefresh=15
localnet=xxx.xxx.xxx.xxx/255.255.255.0

When using Dyanmic DNS make sure Asterisk is using an external DNS server, and the hostname resolves to the public IP address.

Asterisk Debugging Tips

Asterisk SIP Messages

Understanding the sip messaging in Asterisk is very straight forward. There are only
4 SIP dialogs used by Asterisk. They are REGISTER, INVITE, OPTIONS, and NOTIFY.

Register dialogs are used to perform registration requests between Asterisk and the
Proxy server. These are identifiable by the CSeg: REGISTER header in the SIP message.

INVITE dialogs are used to initiate a call through the system. The initial INVITE message
can be originated either by the Proxy Server (for incoming calls) or Asterisk (for outgoing
calls).

Options dialog is used by Asterisk as a ping or keep alive. This is usefull for keeping NAT
ports open.

Notify dialog is used for Message waiting indicator.