Tested and working SIP provider configurations

Trying to get some SIP accounts working correctly in Asterisk & FreePBX can take hours of trial and error.
In the hope of saving others some wasted time, I'm starting a list of settings for various providers.

Please post tested & fully working setup info only.

When submitting, please include all information:
General settings,
Outgoing dial rules (if anything special needed),
Outgoing Settings section,
Incoming Settings section,
Registration string format.
Any requirements for incoming routes (DID etc.)

Don't forget to remove your real username, password & DID number when posting.


This list has been rearranged to put providers in alphabetical order. If you add a new one (or if a provider changes their name) please put their information in the proper position in alphabetical order. Use only one exclamation point before the provider name, or it will not be indexed in the "Quick Links to Provider Configurations" list below. Some of these configurations were provided by jpe-nyc in the #freepbx channel.

NOTE: If you have any problems receiving incoming calls or routing them properly using inbound routes, see How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail)

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Allo.com

NOTE: It appears that Allo.com will cease operations effective August 1, 2007. However, there is a note in this configuration explaining the need to place context names in a particular alphabetical order, which may prove useful to others who are having problems. These are extracted from the the Allo.com Trixbox configuration page (pdf file) - we assume that these settings were tested prior to being posted on the site.

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OUTGOING SETTINGS

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Trunk Name: allo_out
PEER Details:
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=inband
fromdomain=allo.com
fromuser=<allousername>
host=<alloproxy>
secret=<allopassword>
type=peer
username=<allousername>

NOTE: If you prefer to use the g729 codec (not recommended because the documentation states that "some system only support 711, thus defining your trunk as 729 will not let you call those systems") then replace the above "allow" and "dtmfmode" lines as follows:

allow=g729
dtmfmode=rfc2833

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INCOMING SETTINGS

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User Context: in_allo
USER Details:
context=from-trunk
host=<alloproxy>
type=peer

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Registration String: <allousername>:<allopassword>@<alloproxy>/<allousername>

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NOTE (also from the Allo.com documentation): Make sure to use in_allo for user context. Basically you have to make sure that the Trunk_Name comes before the User_Context in alpha order. Trixbox writes the sip config files following alpha order, in return asterisk will chose the last type=peer entry of the sip config when a call comes from host=X. e.g. Trunk_Name : allo_out, User Context : allo_in is BAD; Trunk_name : allo_out, User_Context : in_allo is GOOD

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free-voip.eu

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OUTGOING SETTINGS

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Trunk Name: freevoip_eu
context=from-pstn
host=sip.free-voip.eu
nat=yes
type=peer
fromuser=sipnr
secret=sippasswd
fromdomain=sip.free-voip.eu

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INCOMING SETTINGS

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USER Context: freevoip_eu-in
context=from-trunk
nat=yes
fromuser=sipnr
secret=sippasswd
fromdomain=sip.free-voip.eu
type=user

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Registration String: sipnr:sippasswd@sip.free-voip.eu

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FWD (Free World Dialup)

NOTE: These detaiils are for setting up a SIP trunk. Although FWD claims to support IAX trunks, the reality is that most IAX trunks have apparently stopped working, while SIP trunks continue to work just fine. Therefore we are only giving instructions for setting up a SIP trunk at this time.

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OUTGOING DIAL RULES

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NOTE: FWD can complete toll-free calls to several countries, but requires * in front of the country code on such calls, so you will probably want to add it here, as shown above. Note that the above assumes that international dialing prefix codes are not used or have already been stripped - if that's not the case, you'll need to modify the above entries appropriately. For example, in those parts of North America where 011 is the international dialing prefix, you may want to use a syntax such as this:

(in other words, adding 011| to the start of the international entries) for all non-domestic toll-free numbers.

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OUTGOING SETTINGS

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Trunk Name: fwd
PEER Details:
allow=ulaw
context=from-trunk
disallow=all
host=fwd.pulver.com
qualify=no
secret=xxxxxxxxxx
type=peer
username=xxxxxxxxxxx (your FWD number)

NOTE: FWD doesn't send proper DID information, which may cause incoming calls to fail. To fix that problem, try this: instead of setting the context= to from-trunk, set it to custom-from-fwd and then add the following lines to the end of /etc/asterisk/extensions_custom.conf:

See How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail) for a discussion of why you can't just create an inbound route for FWD without going through some extra steps.

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INCOMING SETTINGS NOT USED

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Registration String: number:secret@fwd.pulver.com (Be sure to use your FWD number, not your FWD account user name!)

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les.net

NOTE: this provider recommends a somewhat simpler configuration than what is shown below - see their pages at:
http://les.net/config_example_freepbx_trunk.php
http://les.net/config_example_freepbx_inbound_route.php
Try their suggested settings first with one exception: DON'T use from-trunk in the USER Context field, use something descriptive instead, such as les-in (but DO use from-trunk in the other places where it is shown). If you have problems, THEN and ONLY then try modifying them to what is shown below (note the lines in red are NOT part of the provider's recommended configuration):

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OUTGOING SETTINGS

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Trunk Name: lesnet
PEER Details:
context=from-trunk
host=did.voip.les.net
nat=yes
qualify=yes
type=peer

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INCOMING SETTINGS

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USER Context: les-in
USER Details:
context=from-trunk
nat=yes
secret=xxxxxxxx
type=user

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Registration String: account:password@did.voip.les.net/account

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Peopleline.net (formerly Evolve)

OUTGOING SETTINGS

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fromuser=604XXXXXXX
username=604XXXXXXX
host=gs.ucantalk.net
nat=yes
outboundproxy=op.ucantalk.net
outboundproxyport=5062
qualify=yes
secret=xxxxxxxx
type=peer

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INCOMING SETTINGS NOT USED

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Registration String: 604XXXXXXX:xxxxxxxx@gs.ucantalk.net

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SellVOIP

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OUTGOING SETTINGS

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Trunk Name: sellvoip
PEER Details:
allow=ulaw
disallow=all
host=seaiax1.sellvoip.net
qualify=yes
secret=xxxxxxxxxx
type=peer
username=xxxxxxxxx

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INCOMING SETTINGS

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USER Context: username
context=from-pstn
secret=xxxxxxx
type=user

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Registration String: username:secret@seaiax1.sellvoip.net

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Sip Number

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OUTGOING SETTINGS

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Trunk Name: sipnumber
PEER Details:
allow=ulaw&alaw
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=sipnumber.net
fromuser=xxxxxxxxx
host=sipnumber.net&outbound.freedigits.net
insecure=very
nat=no
qualify=yes
secret=xxxxxxxxxx
type=peer
username=xxxxxxxxxx

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INCOMING SETTINGS

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USER Context: username
USER Details:
context=from-pstn
insecure=very
nat=no
secret=xxxxxxx
type=user
username=xxxxxxxxxx

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Registration String: username:secret@sipnumber.net/username

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Stanaphone

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OUTGOING SETTINGS

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Trunk Name: stanaphone
PEER Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
fromdomain=sip.stanaphone.com
fromuser=xxxxxxxxxx
host=sip.stanaphone.com
insecure=very
qualify=yes
secret=xxxxxxxxxx
type=friend
username=xxxxxxxxx

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INCOMING SETTINGS NOT USED

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Registration String: username:secret@sip.stanaphone.com/did

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SunRocket

NOTE: Although SunRocket is now defunct (and we have no idea whether these settings will work with the SunRocket heir apparent, Teleblend), this configuration shows an example of adding a useragent= line to sip.conf to emulate a particular VoIP adapter, which might be useful in the future with certain other providers. These settings are from the VoIP experiences blog, which has the following note: "In order to get FreePBX working with SunRocket one needs to set up the trunk correctly. Here are the settings I have working at the moment. This is assuming you have the Innomedia Gizmo. These may work for users with other ATAs, but I can't say for sure."

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Edit sip.conf and add or modify the useragent line to:
useragent=InnoMedia SIP MTA6328-2Re v3.0.77

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OUTGOING SETTINGS

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Trunk Name: sunrocket
PEER Details:
allow=ulaw
authuser=<AuthID>
disallow=all
dtmfmode=auto
fromdomain=sunrocket.com
fromuser=<UserID>
host=67.133.234.125
insecure=invite
nat=1
outboundproxy=67.133.234.125
secret=<Passwd>
type=friend
username=<AuthID>

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INCOMING SETTINGS NOT USED

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Registration String: <UserID>:<Passwd>:<AuthID>@sunrocket.com/<UserID>

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NOTE (also from the blog): In the above settings:
<AuthID> is the AuthID from the Innomedia (basically extended account number)
<UserID> is essentially your 10 digit SunRocket telephone number
<Passwd> is the Passwd from the Innomedia

That's all the required settings. The difficulty may be though that if your <Passwd> is like mine, it contains equal signs (=). The currently released version of FreePBX (2.2) has a bug that prevents PEER Details from having values that contain equal signs. See trac bug report for the details and the fix.

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Viatalk

OUTGOING SETTINGS

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Trunk Name: viatalk
PEER Details:
allow=ulaw
authuser=xxxxxxxxxxx (your 11 digit viatalk number)
context=from-trunk
disallow=all
dtmf=auto
dtmfmode=inband
fromdomain=chicago-1.vtnoc.net
fromuser=xxxxxxxxxxx (your 11 digit viatalk number)
host=chicago-1.vtnoc.net
insecure=very
qualify=yes
secret=xxxxxxxxxx
type=peer
username=xxxxxxxxxxx (your 11 digit viatalk number)

NOTE: Use your nearest ViaTalk server as selected in the ViaTalk control panel for fromdomain and host settings (Chicago is shown in this example)

NOTE: If you have problems with touch tones not being able to control your IVR on incoming ViaTalk calls, or you find you can't create an inbound route that works, try this: instead of setting the context= to from-trunk, set it to custom-from-viatalk and then add the following lines to the end of /etc/asterisk/extensions_custom.conf:

Where you see your-viatalk-number replace that with your 11 digit ViaTalk number. Now inbound touch tones may work more reliably, and also you'll be able to create a specific inbound route for ViaTalk using the 11 digit number. See How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail) for a discussion of why you can't just create an inbound route for ViaTalk without going through some extra steps.

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INCOMING SETTINGS NOT USED

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Registration String: YOUR 11 DIGIT NUMBER:YOUR SIP PASSWORD@YOUR VT PROXY/YOUR 11 DIGIT NUMBER

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Voxalot

OUTGOING SETTINGS

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Trunk Name: voxalot
PEER Details:
allow=ulaw
context=from-trunk
disallow=all
host=XX.voxalot.com (replace XX with au or eu or us - your selected server)
secret=xxxxxxxxxx
type=peer
username=xxxxxxxxxxx (your Voxalot number)

NOTE: Voxalot doesn't send proper DID information, which may cause incoming calls to fail. To fix that problem, try this: instead of setting the context= to from-trunk, set it to custom-from-voxalot and then add the following lines to the end of /etc/asterisk/extensions_custom.conf:

Replace xxxxxx with your Voxalot number. See How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail) for a discussion of why you can't just create an inbound route for Voxalot without going through some extra steps.

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INCOMING SETTINGS NOT USED

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Registration String: number:secret@XX.voxalot.com (replace XX with au or eu or us - use the server you have selected in your Voxalot account settings, both here and in the outgoing settings.)

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