Asterisk phone swissvoice ip10s

Asterisk and the Swissvoice IP10s phone

This article covers the Swissvoice ip10s (MGCP version; Swissvoice also offers H.323 and SIP): MGCP phone with graphical display (no backlight).

Latest SIP firmware: 1.0.1b4 (Aug 2006)
Latest MGCP firmware: 1.0.5b8 (Aug 2006)
Latest H.323 firmware: 1.1.0b1 (Aug 2006)

Note that Innovaphone is now marketing this re-branded device as IP110. Swissvoice was in financial trouble recently and dropped all VoIP activties and in particular the French company belonging to the Swissvoice group that was producing the ip10 phones.

It appears that the MGCP channel supports the following service codes:

  1. - blind transfer
FLASH - consultative transfer (buggy!)
  • 67 - Calling Number Delivery Blocking
  • 70 - Cancel Call Waiting
  • 72 - Call Forwarding Activation
  • 73 - Call Forwarding Deactivation
  • 78 - Do Not Disturb Activation
  • 79 - Do Not Disturb Deactivation
  • 8 - Call pick-up


  • read the handbook draft offered by digium to learn about the three different types of .conf files. Be aware that mgcp.conf does not work the same way as sip.conf... so place your line= statement at the end...
  • the only way I found to set the "phone name" was to change the user's display settings "idle text": that idle text IS the phone name it seems. That'll matter for any TFTP files that you might plan to load at boot.
  • 3-way-calling with this phone will not work. See bug 881.
  • mgcp phones should not be dynamically ip managed cause mgcp relies on the ip address for identification, as does asterisk in mgcp.conf

Screen logo format:

The supported format is BMP (black & white) , size 128x40.


  • does anyone have access to the "MGCP XML" guide or any description of the other configuration files that can be loaded via TFTP or FTP?

Sample entry for mgcp.conf:

; note that transfer=yes requires threewaycalling
; transfer=yes permits FLASH transfers, assign FLASH to the key F1!
singlepath=yes ; only one RTP stream is supported by the ip10s
; we have too much trouble with the ip10s and callwaiting
; so we better turn this feature off (sniff sniff)
; not sure what nat=yes actually does for MGCP devices
; we prefer canreinvite=no so that ASTERISK can do codec translation
; the ip10s doesn't provide GSM, so we cannot talk directly to X-Lite
; using the GSM codec
; not sure why anyone would want to use inband dtmf ...
callerid = Sven Svoboda
; if you configure the F4 key with the "voicemail" function then
; this key will light up in the event of vm box 5444 having msgs
; note, however, that you'll have to lift and replace the handset
; in order to update the blinking status of the F4 button
line => aaln/1

Sample extensions.conf entry:

exten => 1000,1,Dial(MGCP/aaln/1@,20,rt)

Sample voicemail context for extensions.conf:


exten => #,1,GotoIf($[${CALLERIDNUM} = 101]?11:2)
exten => #,2,GotoIf($[${CALLERIDNUM} = 102]?15:3)
exten => #,3,GotoIf($[${CALLERIDNUM} = 103]?19:40)
exten => #,11,Playback(vm-login)
exten => #,12,SayDigits(${SWISS1VM})
exten => #,13,VoiceMailMain2(${SWISS1VM})
exten => #,14,Hangup
exten => #,15,Playback(vm-login)
exten => #,16,SayDigits(${SWISS2VM})
exten => #,17,VoiceMailMain2(${SWISS2VM})
exten => #,18,Hangup
exten => #,19,Playback(vm-login)
exten => #,20,SayDigits(${SWISS3VM})
exten => #,21,VoiceMailMain2(${SWISS3VM})
exten => #,22,Hangup
exten => #,40,VoiceMailMain2()
exten => #,41,HangUp

include => vm-mgcp

Telnet access

username: target
password: password

In the telnet session to the VxWorks CLI you have two modes: application level prompt or operating system prompt. To make changes to your configuration you need to be on the application level prompt!

Some commands:
dbgcmd: go to application prompt
exit: leave application prompt
logout : close telnet session
show * : display the current *
set * : change *
commit * : commit the changes to *

where * can be configure, telnet, telbook, utilities, feature, .... type only show if you'd like to see a list of all available options.
If you have made changes this way, you still need to

activate : activate the new setting on your phone
commit : write the new setting to flash memory.

SIP: BusyLamp (BLF) and call pick-up

You needed to go into telnet mode, to activate the busy lamp, with the hint option (remember to set call-limit=1 in sip.conf). Moreover, if you wanted to pick up the phone call, then you needed also to add another telnet command to handle this pickup! See the firmware release notes of Swissvoice for details on the telnet commands.

Taken from the SIP firmware notes:

Ticket AFV 01184: Telnet Busy Lamp Appearance command

MXP> set functionkey <ID[F1..F4|0..9]> FREE_NUMBER "<Name>" "<Number>" <BusyLamp[on/off]>
MXP> activate

Telnet example 1 (SIP)

set sip transport udp
set sip udp_listen_port 5060
set sip registrar_proxy on 5060
set sip outbound_proxy on 5060
set sip auto_register on
set sip register_timeout 3600
set sip refer_timeout 1800

set tcid 0 sip endpoint_name 701
set tcid 0 sip auth on 701 XXXX

set idle_screen free_idle "ext701 AEP"

set utilities sntp_srv_ipaddr 82.211.81.XXX
set utilities sntp_mode request
set utilities sntp_date_format 0
set utilities sntp_time_zone 12
set utilities tone_table EU


Telnet example 2 (MGCP)

MXP>show service_state
+----------------------------------------------+---+| + STATES | I | 1 | 1 | 1 | 1 | 1 | 1 | 2 | 2 | 2 | 2 | 2 | 2 |
| + | D | R | D | C | A | B | H | R | C | D | C | A | B |
| + | L | I | I | A | C | U | O | I | W | I | A | C | U |
| + | E | N | A | L | T | S | L | N | T | A | L | T | S |
| FEATURES + | | G | L | L | I | Y | D | G | | L | L | I | Y |
+----------------------------------------------+---+| VOICEMAIL | | | | | | | | | | | | | |
| CR | | | | | | | | | | | | | |
| CTB | | | | | | | | | | | | | |
| CH | | | | | | | | | | | | | |
| CE | | | | | | | | | | | | | |
| CWA | | | | | | | | | | | | | |
| FLASH | | | | | | | | | | | | | |
| Transfer | | | | | | | | | | | | | |
| Operator | | | | | | | | | | | | | |

How to use an external phone book

IP10S phone supports access to Cisco Phone Book but not all functionalities. The IP10 uses his own interface to access to the Phone Book. If you want to connect to your remote phone book, you have to do the following actions:

First, copy the URL under Search by name in a Web browser, for example:

You are going to have a XML file display in the Web Browser, like this one:

<Title>Directory Search</Title>
<Prompt>Enter search criteria</Prompt>
<DisplayName>First Name</DisplayName>
<DisplayName>Last Name</DisplayName>

Copy the information from the URL line

The easiest way to set the path in your phone is by the Web interface (but it could also be done by Telnet).
Connect to your phone web server. Login and password are normally: admin
Select Configure common phonebook. In Select phone book to use chose the value: Remote Then click on submit. File the box below with the URL you get previously:

The IP address and the port number of the Phonebook server can be manually entered or synchronised with the Call Agent. In our case, IP address:, Port number: 8080,Path:

Then click on submit. If you return to your phone and select the common phone book, it is normally connected to the remote one now. You can search by a name or if you put nothing and press on OK, it will return you the entire content of the remote phone book.

More information about Cisco Phonebook management To manage remote phone book, you need a server. It could be one you developed by yourself or one include in your proxy server (not all of them include this feature). It must follow the Cisco implementation; you can have more information here:
Check for "Cisco IP Phone Services Application Development Notes with Cisco CallManager 3.1."

News (Oct. 2005)

It appears that SIP firmware 1.0.0 is now available

News (December 2004)

by Paolo Marini

While looking into newsgroups and searching for usage of IP10S phone, SIP version, with asterisk, some users report the phone can register but not call nor answer to a terminated call.
I have got 2 SIP and 2 MGCP IP10S phones, upgraded them to build 7 (SIP) and used them with asterisk succesfully.

The important thing to do after the upgrade is to restart the phone while keeping the 1-4-7 keys pressed, in order to restore the factory defaults in the configuration data, then configure the phone and use it. Without doing this procedure, I was myself stuck with the phone registering but not answering to calls.

The phone provides 2 lines, call transfer and switch between the 2 calls.

It seems that profile saving from the WEB interface is not working.

Wishlist for SIP version:

  • Have the profile saving/restore working from the WEB interface
  • Have the configuration data documented in order to provision the phone by tftp/dhcp
  • Have the password for telnet login
  • Have documentation for the command that can be issued by telnet

News (October 2004)

New release of SIP firmware from Swissvoice I've made this available temporarily here - version 0.0.1build 7 - anyone
care to offer hosting for this stuff email me michael at Note, there was a small permissions issue on the server this is hosted on at 9:30am EST 10/04/04,
it has been fixed and the files are now available.

News (June 2004)

by Florian Overkamp
SIP firmware is currently being tested, there are a few issues that need to be resolved.

For your MGCP phone: configip10.cfg can be altered to add services:

set features new 1 "Transfer" NOINFO NOCONF TRUE NOSEQ
set features new 2 "Operator" NOINFO NOCONF FALSE <extension of your secretary>

And then:

set service_state IDLE NEW 2
set service_state ONE_ACTIVE_LINE NEW 1

This will add two services:
In idle state: An operator button that speeddials your secretary (who can then connect you through)
In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved).

See also

Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones

Created by: jpiterak, Last modification: Wed 19 of Jan, 2011 (20:59 UTC) by JustRumours
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