Interop List
From FreeSWITCH Wiki
This is where we need to list all devices that have worked successfully with FreeSWITCH.
Please make sure that you have the following criteria in your phone review:
- Registration: work/do not work
- Caller Id: work/do not work
- Call in/out: work/do not work
- Call waiting: work/do not work
- Transfer calls: work/do not work
- Park calls: work/do not work
Contents |
Asterisk
Please refer to: Connecting Freeswitch And Asterisk
D-Link Products
D-Link DIV-140
In general not very function, with little technical support. Have never been able to actually dial out on the box, and there does not seem to be any way to have it automatically dial a call on a PSTN line. The documentation is quite terrible.
D-Link DPH-540
Works fairly well.
- Registration: works
- Caller Id: No Caller Id name
- Call in/out: work
- Call waiting: work
- Transfer calls: unknown
- Park calls: unknown
Sipura Products
Sipura SPA-2000
Works very well. Ensure your RTP packet size is set to 0.020 under the SIP -> RTP Parameters tab, as the Sipura doesn't advertise it's ptime capabilities correctly.
Sipura SPA-2002
Works very well as a client side UA.
- Registration: works
- Caller Id: works
- Call in/out: work
- Call waiting: work
- Transfer calls: works
- Park calls: works
Has the same ptime issues as SPA-2000.
Linksys Products
Notes: Most Linksys products has RTP Packet size set to 0.30 by default. Ensure your RTP packet size is set to 0.020 under the SIP -> RTP Parameters tab, as some devices doesn't advertise it's capabilities correctly. In firmware versions 5.2.x, ptime option is advertised correctly (tested with 942s and 962s).
On phones series SPA9xx paging works via "<action application="set" data="sip_h_Call-Info=<sip:$${domain}>;answer-after=0"/>" before calling the phone.
Linksys PAP2-NA
- Registration: Works
- Caller Id: Works
- Call in/out: Works
- Call waiting: Works
- Transfer calls: Works
- Park calls: Not Tested
Linksys PAP2 v2
Unlocked with CYT-Unlocker, PAP2 V2 has no STUN ability!
- Registration: Works
- Caller Id: Works
- Call in/out: Works
- Call waiting: Works
- Transfer calls: Not Tested
- Park calls: Not Tested
Linksys PAP2T
- Registration: Works
- Caller Id: Unknown
- Call in/out: Works
- Call waiting: Unknown
- Transfer calls: Unknown
- Park calls: Not Unknown
Linksys SPA8000
- Registration: works
- Caller Id: unknown
- Call in/out: works
- Call waiting: unknown
- Transfer calls: unknown
- Park calls: unknown
Linksys SPA941
Note: Has the RTP Packet Size issue.
Linksys SPA942
- Registration: Works
- Caller Id: Works
- Call in/out: Works
- Call waiting: Works
- Transfer calls: Works both from phone or FS
- Park calls: Works
Note: Has the RTP Packet Size issue.
Linksys SPA962
Works very well. It will advertise the ptime correctly. No need to toy with RTP Packet Size.
Linksys SPA932
Sidecar for SPA962 with 16 programmable buttons. Since firmware version 5.2.8SC, presence is supported (Server-Type option set to "RFC3265_4235"). BLF, hold and intercept of ringing extensions work.
Cisco Products
Cisco has SIP images for both old and new generation of their phones. Some firmwares can be found here.
Cisco 7940G
- Registration: Works (possible NAT issues)
- Caller Id: Works
- Call in/out: Works
- Call waiting: Works
- Transfer calls: Works (possible att. transfer issue)
- Park calls: Works
Cisco UC520
The UC520 is a fairly flexible PBX geared for small businesses; and the SIP stack it uses seems to be reasonably
compatible with FreeSWITCH out of the box.
- Registration: Works
- Caller Id: Works
- Call in/out: Works
- Call waiting: Works
- Transfer calls: N/A?
- Park calls: N/A?
- To call into this box from FreeSWITCH, it just blindly accepts SIP to the pre-determined 3 digit extensions.
- To use FreeSWITCH as a gateway for this box:
First, setup a SIP UA entry (assumes your FS box is 10.50.0.50):
sip-ua authentication username uc500 password 7 encryptedgoo retry invite 2 retry register 10 timers connect 100 registrar ipv4:10.50.0.50 expires 3600 sip-server ipv4:10.50.0.50 !
Then you have to setup a dialplan ("dial-peer") entry, this one requires you dial '9' before your outbound number:
dial-peer voice 1900 voip description apex SIP test destination-pattern 9[2-9]...... session protocol sipv2 session target ipv4:10.50.0.50 codec g711ulaw no vad !
SNOM Products
SNOM 190, 300, 320, 360, 370
Works fine. Multiple line appearances, call waiting, transfer, etc.
- Attended Transfers
- Blind Tranfers
- CMC Code
- Record Button (Set record template on sip profile)
- TLS and SRTP
Snom370 Configuration
This is required on the user to force TLS to work proper with the SNOM with version 7 firmware.
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/>
Grandstream Products
Grandstream GXW 4004
Registrations work, call in-call out work
Grandstream HandyTone 488
FXS side of things appears to work well. I haven't tested caller ID, nor the FXO side of the device.
Surprisingly good ULaw sound over wan (cable modem).
Also, surprisingly effective NAT support.
DTMF tones seem to be failing when connected to the IVR module (but working when forwarded through to the free 800 # service).
Grandstream GXP-2020
GXP-2020, SRTP works too. G722 is broken.
Aastra Products
Aastra 55i
SRTP in Preferred mode doesn't work. G722 on this phone sounds choppy. TLS not tested.
Aastra 57i
TLS not working
Polycom Products
Polycom IP 431
basic registration and calling in and out works, haven't tested any other functionality yet
Polycom IP 320
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: works
- Transfer calls: work
- TLS: not tested
- SRTP: not tested
Polycom IP 501
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: works
- Transfer calls: work
- TLS: works (must install custom CA cert from the phone and force it to use the cafile.pem)
- SRTP: works.
Polycom IP 550/650
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: works
- Transfer calls: work
- TLS: works (must install custom CA cert from the phone and force it to use the cafile.pem)
- SRTP: works.
AVM Products
FRITZ!Box Fon WLAN 7050
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: not tested
- Transfer calls: not tested
- TLS: not tested (maybe not even supported, have to look it up)
- SRTP: not tested
FRITZ!Box Fon WLAN 7270
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: not tested
- Transfer calls: not tested
- TLS: not tested
- SRTP: not tested
Conference rooms do not work because the phone is obviously using 30ms RTP packets but it is not advertising it. This results in some very weird sound-effects for the other people in the room. We will file a bug at AVM for that.
Microsoft
Microsoft Exchange 2007 Unified Messaging Server
Please refer to: Exchange 2007 UM
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: not tested
- Transfer calls: works
- TLS: not tested
- SRTP: not tested
Mitel devices
Mitel 3300 ur3
Using a SIP Trunk, works well calling Mitel phones over SIP or Minet, voicemail, etc..
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: not tested
- Transfer calls: not tested
- TLS: not tested
- SRTP: not tested
Other Devices
Multipoint MultiVoip Media Gateway
This is a dual PRI to SIP gateway intended to be used to provide a SIP gateway to a legacy PBX with PRI. I have tested it as a SIP to PRI gateway between FreeSwitch and a CopperCom CSX, and Lucent 5ESS switch. While the web UI has some of the labels turned about when used in this configuration it functions fine and passes calls in and out properly. DNIS, CallerID, and RDNIS are passed from PRI to SIP properly.
Patton M-ATA-1 Micro Analog Telephone Adapter
Works fine. Brief documentation, but functional.
Mobilephones
Nokia N95
- Registration: works
- Caller Id: works
- Call in/out: works
- Call waiting: works
- Transfer calls: works
- Park calls: unknown
Nokia E60
- Registration: works
- Caller Id: unknown
- Call in/out: work
- Call waiting: work
- Transfer calls: works
- Park calls: unknown
All Nokia phones support STUN. This is configurable through XML provisioning files or the SIP VoIP Settings tool
Skype (using SippySkype)
Please refer to SippySkype Skype Adapter
- Registration: doesn't work
- Caller Id: Works
- Call in/out: works
- Call waiting: not tested
- Transfer calls: not tested
- Park calls: not tested
Softphones
Twinkle 0.8
- Registration: works
- Caller Id: unknown
- Call in/out: work
- Call waiting: unknown
- Transfer calls: works
- Park calls: unknown
Bria 2.2
- Registration: works
- Caller Id: works
- Call in/out: work
- Call waiting: works
- Transfer calls: works
- Park calls: unknown
CounterPath X-Lite Softphone
X-Lite release 1105d build stamp 99999 Registrations, Calling in and out works
3CX Phone 1.17
- Registration: works
- Caller Id: unknown
- Call in/out: work
- Call waiting: unknown
- Transfer calls: unknown
- Park calls: unknown
- NAT Traversal: unknown (Initial Failure)
Express Talk 2.06
http://www.nch.com.au/index.html
- Registration: works
- Caller Id: unknown
- Call in/out: work
- Call waiting: unknown
- Transfer calls: unknown
- Park calls: unknown
- NAT Traversal: unknown (Initial Failure)
Zoiper 2.14
- Registration: works
- Caller Id: unknown
- Call in/out: work
- Call waiting: unknown
- Transfer calls: unknown
- Park calls: unknown
- NAT Traversal: unknown
PBX Softwares
CallButler
- Registration: works
- Caller Id: works
- Call in/out: work
- Call waiting: unknown
- Transfer calls: unknown
- Park calls: unknown
