MVP410ST and 810ST with Asterisk

How to use MultiTech's MultiVoIP MVP-810(ST) with Asterisk


The MultiTech MVP-410ST and MVP-810ST are part of the MultiVOIP family.
It provides you with an VoIP gateway from SIP & H323 to and from BRI-interfaces.
It has T.38 support on H.323 enabling you to receive faxes reliable and has a fallback system so you are able to configure a fail-over system for calls coming in over the BRI-channels.

Configuring this unit can be done using H.323 and SIP. For now I will only explain the configuration using SIP in combination with T.38 for receiving faxes.
A SIP-Proxy would not be possible anyway since Asterisk is only a SIP-Registrar NOT a SIP-Proxy.
However you are able to configure the MultiVoIP & Asterisk in such a manner that it will work.
If you feel up to it you are free to add a document explaining how to configure this unit with the OH323 Gatekeeper, but for now I haven’t felt the need…
I would advise you to upgrade the unit to the latest release available. Many bugs have been reported through time and most of them have been solved in newer releases.

Configuring Asterisk SIP.CONF
First of all you need an SIP-account at which the MultiVoIP unit is able to register itself in order to place calls outbound. For receiving calls we use the same account but we set the security very low. This off course introduces a security risk and therefore it’s best to use totally different context for the MultiVoIP unit. Although in this example I do allow guest calls I use the default context to play a “Go Away� message to any unauthorized user.
The SIP-Account created can be used incoming as well as outgoing.If you do apply a secret for this account, Asterisk will try to authenticate every call using the callers CLI which would mean you need an account for every number in this world…..
To solve this, use an extra setting called "insecure=very"
Doing this will only allow this SIP-Account to call the numbers listed in the context AND not the default context.

My SIP.CONF file thus looks like:

;
; SIP Configuration for Asterisk
;

[general]
context=default ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpirey=3600 ; Max length of incoming registration we allow
disallow=all ; First disallow all codecs
allow=alaw ; Allow alaw codec in default context
allow=ulaw ; Allow ulaw codec in default context
allow=gsm ; Allow gsm codec in default context
musicclass=default ; Sets the default music on hold class for all SIP calls
language=nl ; Default language setting for all users/peers
rtptimeout=180 ; Terminate call if 60 seconds of no RTP activity
useragent=Asterisk ; Allows you to change the user agent string

[MultiVoIP]
type=peer
username=MultiVoIP
secret=secret
context=sip-incoming
host=dynamic
nat=no
insecure=very
canreinvite=no
qualify=no
disallow=all
allow=alaw

[User1]
type=friend
context=internal
username=User1
callerid="User 1" <100>
secret=hissecret
qualify=yes
nat=yes
host=dynamic
canreinvite=no
language=nl
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[User2]
type=friend
context=internal
username=User2
callerid="User 2�<101>
secret=hissecret
qualify=yes
nat=yes
host=dynamic
canreinvite=no
language=nl
disallow=all
allow=alaw
allow=ulaw
allow=gsm



Configuring Asterisk EXTENSIONS.CONF
This extensions.conf file of Asterisk is quite simple, but includes all the contexts mentioned before.
To call out to the PSTN it simply uses the same “MultiVoIP� SIP-Account created in sip.conf
Nothing special is done to change CLI or things like that.
The incoming DID’s in this setup are 555-123456 for ext. 100 and 555-123457 for ext. 101



[general]
static=yes
writeprotect=no


[default] ; People who are not authorised go to this context
exten => _[0-z].,1,Answer
exten => _[0-z].,2,Wait(1)
exten => _[0-z].,3,Playback(privacy-stop-calling-not-welcome) ; Available in Asterisk-addon-sounds
exten => _[0-z].,4,Wait(3)
exten => _[0-z].,5,Hangup


[internal] ; Internal callers get into this context
include => outgoing ; Include context outgoing within this context

exten => 100,1,Dial(SIP/User1)
exten => 101,1,Dial(SIP/User2)

exten => t,1,Congestion
exten => i,1,Congestion
exten => t,1,Congestion
exten => T,1,Congestion


[sip-incoming]
exten => 555123456,1,Goto(internal,100,1)
exten => 555123457,1,Goto(internal,101,1)
exten => t,1,Congestion
exten => i,1,Congestion
exten => t,1,Congestion
exten => T,1,Congestion


[outgoing]
Exten _0X.,Dial(SIP/MultiVoIP/${EXTEN})



Configuring the MultiVoIP unit itself
To configure the MultiVoIP unit you first need to make a serial connection using the supplied (Windows) software in order to set the IP-address, after this is done you can use the Web-interface of the MultiVoiP unit to do any other configurations. Although I would advise you to use the MultiVoIP software client itself since it is more stable. My unit had a default password on delivery which was hard to figure out since it wasn’t listed anywhere I looked, it turned out to be username: “admin� and password: “00000� (5 zeros)
Normally there should not be a password protection active.

The configuration itself is quite simple although some things should be kept in mind.
One smart thing to do is to follow the steps from the upper left menu-tree from top to bottom so you won’t be forgetting anything.

Setting the IP-address:
Just configure the IP-settings as you wish.
You are allowed to use DHCP although I would favour using a Static IP-address to make management easier.
It’s also possible to use DNS if you are using this on your Asterisk server as well, otherwise it’s of no real use.

Configuring Voice/Fax
Don't change much on this tab, just select the right codec and your done.
Just don’t forget to use the button Copy Channel, to copy your new settings to all voice-channels.
Off course you are free to fool around with the settings although I would do this after I got the unit to work..

Configuring ISDN BRI
Now this tab is somewhat more interesting…
You need to configure this correctly or else it would seem that the unit isn’t doing anything at all.

First off all, you would want to set the Layer 1 interface mode…
This is will set the ISDN-ports in TE or NT mode…
(TE is for connecting to a telco and NT if you are connecting ISDN-phones)

Second, you would want to set the correct Switch information.
Depending on which country you are using the system this system the setting may vary.
Choose a country that uses the same type of ISDN lines as you do.
Then select the Operator, actually this is the protocol used to signal over ISDN…
for Europe this is almost always ETSI but if your connecting to another PBX you might want to try ECMA_QSIG.

The TEI-assignment is next and is set to Automatic by default, this is fine if your using a ISDN2 line with MSN-numbers but if your using DID-numbers your telco will most likely expect you to be using TEI-value 0. In that case set the TEI-value as shown above. If you are using MSN numbers, list all these numbers in the MSN-Details box using only the last digits that differ between these numbers, most likely you will be entering the MSN number without the areacode.
Next, you could configure all the other parameters.. but I found out that this is of no use when your using the unit as a SIP-gateway for Asterisk. So play around with them after you got the system to work.
!!! DON’T FORGET TO COPY TO ALL OTHER CHANNELS !!!!

Configuring SNMP
You free do do so… but you don’t need to, so I’ll advise you to read the manual on this.

Configuring Regional
This is simple…. Just select the region closed to you which uses the same type of signalling.
If you would like to change it then choose “Custom� and edit the tones..

Configuring SMTP
This unit can be configured to email error-traps the system administrator if you would like.
Since it is of no further use right now I’ll skip this part, your could read the manual for more info..

Configuring LOGS
This section configures the log settings.
Since it is of no further use right now I’ll skip this part, your could read the manual for more info..

Configuring Supplementary Services
This section enables you to set specific H.323 signalling settings..
Since we are not using H.323 I’ll skip this part, your could read the manual for more info..

Configuring Advanced / Packetization time
Asterisk uses a 20mS packet size for the G711 codec by default so you'll need to change this accordingly…

Configure the Phonebook / SIP Proxy
This is an important part, the phonebook in these units decide where to put an incoming or outgoing call.
It consists out of three separate parts: “The Phonebook configuration�, “The Outbound Phonebook� and the “Inbound Phone Book�. Settings made in “The Phonebook configuration� are used by the inbound and outbound phonebooks to forward calls to other systems. Therefore it’s very important to correctly configure “The Phonebook configuration�.

NOTE for releases 5.08 or higher:
In newer releases "The Phonebook configuration" does no longer exist, instead there is an extra option under the configuration tab named "signalling", under this tab you will find the "SIP-Proxy" tab that lists the same items

For use with Asterisk you need to enable the SIP-Proxy although Asterisk is NOT a SIP-proxy
In newer releases you are able to enter DNS-names but older releases only support IP-addresses.
The username and password used are the set in SIP.CONF.

Next you need to configure rules for the inbound and outbound phonebook.
Important to know is that this OUTBOUND phonebook is used to decide what to do with incoming calls from the ISDN-ports and that the INBOUND phonebook is used to decide what to do with incoming calls from the Asterisk-server. This might not be exactly what your expecting but is just as it is…


Configuring the OUTBOUND phonebook
This is quite simple.. Just click on Add Entry and create a default entry..
Enter mark the option "Use SIP-proxy" and give it a description.
Note that you need to set the Transport protocol to UDP, this is easily forgotten!!


Configuring the INBOUND Phonebook
The last thing you need to configure is the inbound phonebook in order to route calls coming from Asterisk out to the ISDN-ports (telco/PSTN). This also is very simple.
Just add a default entry and for channel number select "Hunting"
And enter a description like "PSTN"


Save and Reboot
Now the last step is to save and reboot the MultiVoIP unit:

Note that this needs to be done after EVERY change you make to the unit!



If you have any more question please use the comments at this WiKi page or better contact MultiTech at:




Configuring the Fax over IP with T.38

(This part still needs some extra info....)

If you like you can use the T.38 capabilities of the MultiVoIP unit to reliable receive Faxes.

Simply add the phone number that you would like to forward to another MVP analog unit as it is being received by the unit.
Then enter the IP-address of this unit (not asterisk, since we are bypassing asterisk.)
Enter a description and select the SIP protocol type.
Now just configure your terminal to dial out using the IP-address of the MultiVoIP unit.
And don’t forget to enable the Fax capability on the Voice channels of the MultiVoIP unit.

Also note that the T.38 protocol is very open to changes by different manufactures and therefore might not work in combination with different manufacturers T.38 FXS terminals. I would therefore advise you to use the Multitech’s MVP130 VoIP to Analog converter.


How to use MultiTech's MultiVoIP MVP-810(ST) with Asterisk


The MultiTech MVP-410ST and MVP-810ST are part of the MultiVOIP family.
It provides you with an VoIP gateway from SIP & H323 to and from BRI-interfaces.
It has T.38 support on H.323 enabling you to receive faxes reliable and has a fallback system so you are able to configure a fail-over system for calls coming in over the BRI-channels.

Configuring this unit can be done using H.323 and SIP. For now I will only explain the configuration using SIP in combination with T.38 for receiving faxes.
A SIP-Proxy would not be possible anyway since Asterisk is only a SIP-Registrar NOT a SIP-Proxy.
However you are able to configure the MultiVoIP & Asterisk in such a manner that it will work.
If you feel up to it you are free to add a document explaining how to configure this unit with the OH323 Gatekeeper, but for now I haven’t felt the need…
I would advise you to upgrade the unit to the latest release available. Many bugs have been reported through time and most of them have been solved in newer releases.

Configuring Asterisk SIP.CONF
First of all you need an SIP-account at which the MultiVoIP unit is able to register itself in order to place calls outbound. For receiving calls we use the same account but we set the security very low. This off course introduces a security risk and therefore it’s best to use totally different context for the MultiVoIP unit. Although in this example I do allow guest calls I use the default context to play a “Go Away� message to any unauthorized user.
The SIP-Account created can be used incoming as well as outgoing.If you do apply a secret for this account, Asterisk will try to authenticate every call using the callers CLI which would mean you need an account for every number in this world…..
To solve this, use an extra setting called "insecure=very"
Doing this will only allow this SIP-Account to call the numbers listed in the context AND not the default context.

My SIP.CONF file thus looks like:

;
; SIP Configuration for Asterisk
;

[general]
context=default ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpirey=3600 ; Max length of incoming registration we allow
disallow=all ; First disallow all codecs
allow=alaw ; Allow alaw codec in default context
allow=ulaw ; Allow ulaw codec in default context
allow=gsm ; Allow gsm codec in default context
musicclass=default ; Sets the default music on hold class for all SIP calls
language=nl ; Default language setting for all users/peers
rtptimeout=180 ; Terminate call if 60 seconds of no RTP activity
useragent=Asterisk ; Allows you to change the user agent string

[MultiVoIP]
type=peer
username=MultiVoIP
secret=secret
context=sip-incoming
host=dynamic
nat=no
insecure=very
canreinvite=no
qualify=no
disallow=all
allow=alaw

[User1]
type=friend
context=internal
username=User1
callerid="User 1" <100>
secret=hissecret
qualify=yes
nat=yes
host=dynamic
canreinvite=no
language=nl
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[User2]
type=friend
context=internal
username=User2
callerid="User 2�<101>
secret=hissecret
qualify=yes
nat=yes
host=dynamic
canreinvite=no
language=nl
disallow=all
allow=alaw
allow=ulaw
allow=gsm



Configuring Asterisk EXTENSIONS.CONF
This extensions.conf file of Asterisk is quite simple, but includes all the contexts mentioned before.
To call out to the PSTN it simply uses the same “MultiVoIP� SIP-Account created in sip.conf
Nothing special is done to change CLI or things like that.
The incoming DID’s in this setup are 555-123456 for ext. 100 and 555-123457 for ext. 101



[general]
static=yes
writeprotect=no


[default] ; People who are not authorised go to this context
exten => _[0-z].,1,Answer
exten => _[0-z].,2,Wait(1)
exten => _[0-z].,3,Playback(privacy-stop-calling-not-welcome) ; Available in Asterisk-addon-sounds
exten => _[0-z].,4,Wait(3)
exten => _[0-z].,5,Hangup


[internal] ; Internal callers get into this context
include => outgoing ; Include context outgoing within this context

exten => 100,1,Dial(SIP/User1)
exten => 101,1,Dial(SIP/User2)

exten => t,1,Congestion
exten => i,1,Congestion
exten => t,1,Congestion
exten => T,1,Congestion


[sip-incoming]
exten => 555123456,1,Goto(internal,100,1)
exten => 555123457,1,Goto(internal,101,1)
exten => t,1,Congestion
exten => i,1,Congestion
exten => t,1,Congestion
exten => T,1,Congestion


[outgoing]
Exten _0X.,Dial(SIP/MultiVoIP/${EXTEN})



Configuring the MultiVoIP unit itself
To configure the MultiVoIP unit you first need to make a serial connection using the supplied (Windows) software in order to set the IP-address, after this is done you can use the Web-interface of the MultiVoiP unit to do any other configurations. Although I would advise you to use the MultiVoIP software client itself since it is more stable. My unit had a default password on delivery which was hard to figure out since it wasn’t listed anywhere I looked, it turned out to be username: “admin� and password: “00000� (5 zeros)
Normally there should not be a password protection active.

The configuration itself is quite simple although some things should be kept in mind.
One smart thing to do is to follow the steps from the upper left menu-tree from top to bottom so you won’t be forgetting anything.

Setting the IP-address:
Just configure the IP-settings as you wish.
You are allowed to use DHCP although I would favour using a Static IP-address to make management easier.
It’s also possible to use DNS if you are using this on your Asterisk server as well, otherwise it’s of no real use.

Configuring Voice/Fax
Don't change much on this tab, just select the right codec and your done.
Just don’t forget to use the button Copy Channel, to copy your new settings to all voice-channels.
Off course you are free to fool around with the settings although I would do this after I got the unit to work..

Configuring ISDN BRI
Now this tab is somewhat more interesting…
You need to configure this correctly or else it would seem that the unit isn’t doing anything at all.

First off all, you would want to set the Layer 1 interface mode…
This is will set the ISDN-ports in TE or NT mode…
(TE is for connecting to a telco and NT if you are connecting ISDN-phones)

Second, you would want to set the correct Switch information.
Depending on which country you are using the system this system the setting may vary.
Choose a country that uses the same type of ISDN lines as you do.
Then select the Operator, actually this is the protocol used to signal over ISDN…
for Europe this is almost always ETSI but if your connecting to another PBX you might want to try ECMA_QSIG.

The TEI-assignment is next and is set to Automatic by default, this is fine if your using a ISDN2 line with MSN-numbers but if your using DID-numbers your telco will most likely expect you to be using TEI-value 0. In that case set the TEI-value as shown above. If you are using MSN numbers, list all these numbers in the MSN-Details box using only the last digits that differ between these numbers, most likely you will be entering the MSN number without the areacode.
Next, you could configure all the other parameters.. but I found out that this is of no use when your using the unit as a SIP-gateway for Asterisk. So play around with them after you got the system to work.
!!! DON’T FORGET TO COPY TO ALL OTHER CHANNELS !!!!

Configuring SNMP
You free do do so… but you don’t need to, so I’ll advise you to read the manual on this.

Configuring Regional
This is simple…. Just select the region closed to you which uses the same type of signalling.
If you would like to change it then choose “Custom� and edit the tones..

Configuring SMTP
This unit can be configured to email error-traps the system administrator if you would like.
Since it is of no further use right now I’ll skip this part, your could read the manual for more info..

Configuring LOGS
This section configures the log settings.
Since it is of no further use right now I’ll skip this part, your could read the manual for more info..

Configuring Supplementary Services
This section enables you to set specific H.323 signalling settings..
Since we are not using H.323 I’ll skip this part, your could read the manual for more info..

Configuring Advanced / Packetization time
Asterisk uses a 20mS packet size for the G711 codec by default so you'll need to change this accordingly…

Configure the Phonebook / SIP Proxy
This is an important part, the phonebook in these units decide where to put an incoming or outgoing call.
It consists out of three separate parts: “The Phonebook configuration�, “The Outbound Phonebook� and the “Inbound Phone Book�. Settings made in “The Phonebook configuration� are used by the inbound and outbound phonebooks to forward calls to other systems. Therefore it’s very important to correctly configure “The Phonebook configuration�.

NOTE for releases 5.08 or higher:
In newer releases "The Phonebook configuration" does no longer exist, instead there is an extra option under the configuration tab named "signalling", under this tab you will find the "SIP-Proxy" tab that lists the same items

For use with Asterisk you need to enable the SIP-Proxy although Asterisk is NOT a SIP-proxy
In newer releases you are able to enter DNS-names but older releases only support IP-addresses.
The username and password used are the set in SIP.CONF.

Next you need to configure rules for the inbound and outbound phonebook.
Important to know is that this OUTBOUND phonebook is used to decide what to do with incoming calls from the ISDN-ports and that the INBOUND phonebook is used to decide what to do with incoming calls from the Asterisk-server. This might not be exactly what your expecting but is just as it is…


Configuring the OUTBOUND phonebook
This is quite simple.. Just click on Add Entry and create a default entry..
Enter mark the option "Use SIP-proxy" and give it a description.
Note that you need to set the Transport protocol to UDP, this is easily forgotten!!


Configuring the INBOUND Phonebook
The last thing you need to configure is the inbound phonebook in order to route calls coming from Asterisk out to the ISDN-ports (telco/PSTN). This also is very simple.
Just add a default entry and for channel number select "Hunting"
And enter a description like "PSTN"


Save and Reboot
Now the last step is to save and reboot the MultiVoIP unit:

Note that this needs to be done after EVERY change you make to the unit!



If you have any more question please use the comments at this WiKi page or better contact MultiTech at:




Configuring the Fax over IP with T.38

(This part still needs some extra info....)

If you like you can use the T.38 capabilities of the MultiVoIP unit to reliable receive Faxes.

Simply add the phone number that you would like to forward to another MVP analog unit as it is being received by the unit.
Then enter the IP-address of this unit (not asterisk, since we are bypassing asterisk.)
Enter a description and select the SIP protocol type.
Now just configure your terminal to dial out using the IP-address of the MultiVoIP unit.
And don’t forget to enable the Fax capability on the Voice channels of the MultiVoIP unit.

Also note that the T.38 protocol is very open to changes by different manufactures and therefore might not work in combination with different manufacturers T.38 FXS terminals. I would therefore advise you to use the Multitech’s MVP130 VoIP to Analog converter.


Created by: ramonpeek, Last modification: Thu 10 of Aug, 2006 (16:22 UTC)
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