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Grandstream

Grandstream Products​

Grandstream BudgeTone 100​

  • Registration: works
  • Caller Id: works
  • Call in/out: works
  • Call waiting: not tested
  • Transfer calls: not tested
  • Park calls: not tested

Grandstream BudgeTone 200​

Codec selection is broken as well as G722. It will "negotiate" the 7th codec in it's preferences. But it works with PCM u-law and a-law, may work with others.

Grandstream GXP1405​

  • Registration: Works
  • Caller Id: Works
  • Call in/out: Works
  • Call waiting: Works
  • Transfer calls: Works
  • Park calls: Works
  • MWI (Message Waiting Indicator): Works

Grandstream GXP-2000​

  • UDP Registration: works
  • TCP Registration: not tested
  • Caller Id: works
  • Call in/out: works
  • G722: works
  • Call Forwarding: works
  • Transfer calls: works - Requires Symmetric routing Enabled
  • Hold calls: works - Requires Symmetric Routing Enabled
  • Call waiting: not tested
  • Park calls: not tested
  • SIP Password: Funky - Not displayed, if you can't get it to register, Reset to Factory defaults and re-enter password
  • Annoying Defaults: Symmetric Routing off, Speaker enabled on disconnect, Non-US DST string, check for update on reboot enabled

Grandstream GXP-2020​

GXP-2020, SRTP works too. G722 is broken. TLS is not working (as on any Grandstream GXP series phone)

GrandStream GXV3140​

  • Registration: works
  • Caller Id: works
  • Call in/out: works
  • Call waiting: works
  • Transfer calls: work
  • Park calls: work via transfer
  • MWI: works
  • SRTP: works
  • TLS: works

GrandStream GXV3140​

  • Registration: works
  • Caller Id: works
  • Call in/out: works
  • Call waiting: works
  • Transfer calls: work
  • Park calls: not tested
  • MWI: not tested
  • SRTP: not tested
  • TLS: not tested
  • Video: work

Grandstream GXW 40xx​

  • GXW4024, GXW4008, GXW4004: Provides FXO ports
    • Registrations work
    • call in and out work
    • Call transfer via touch tones in demo works (Needs configuration tweaking to enable)

Grandstream GXW 410x​

  • GXW4104, GXW4108: Provides FXO ports
    • calls in and out work

Grandstream HandyTone-486​

  • Registration: works
  • Caller Id: works
  • Call in/out: works
  • Call waiting: works
  • Transfer calls: not tested
  • Park calls: not tested
  • MWI: works

Grandstream HandyTone 488​

FXS side of things appears to work well. I haven't tested caller ID, nor the FXO side of the device.

Caller ID works on FXS port (FreeSWITCH 1.0) from SIP originated calls. Also, had issues with upgraded firmware -- reset configuration (***99) and all was well again. I also have not tested FXO port. Jpeterson275 23:19, 2 December 2008 (PST)

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 HandyTone 502​

July 2009: Firmware: 1.0.1.35

According to homepage "SIP over TCP/TLS". But it doesn´t work.

Summary:

  • Registration: works with SIP over UDP and TCP.
  • Caller Id: not tested
  • Call in/out: Works
  • Call waiting: not tested
  • Transfer calls: not tested
  • Park calls: not tested
  • SRTP not tested
  • TLS doesn´t register, see below
  • DTMF not tested
  • FAX over G.711 Works
  • FAX over T.38 Works

Talking about TLS:

In fact there are fields where you can enter your certificate, the private key and the password for the private key. It doesn't register with TLS, but only UDP and TCP. I tried with Pangolin, where TLS works without any problem (importing the CA certifacte of FS). FreeSWITCH tells me:

recv 627 bytes from tls/[10.10.2.206]:2048 at 09:45:02.070075:
------------------------------------------------------------------------
REGISTER sip:domain.de:5061 SIP/2.0
Via: SIP/2.0/TLS 10.10.2.206:5062;branch=z9hG4bK1786789569;rport;alias
Route: <sip:domain.de:5061;lr>
From: <sip:1001@domain.de:5061>;tag=2027020172
To: <sip:1001@domain.de:5061>
Call-ID: 824582011-5062-1@10.10.2.206
CSeq: 2000 REGISTER
Contact: <sip:1001@10.10.2.206:5062;transport=tls>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B821E4582>"
Max-Forwards: 70
User-Agent: Grandstream HT-502  V1.1C 1.0.1.35
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

------------------------------------------------------------------------
send 627 bytes to tls/[10.10.2.206]:2048 at 09:45:02.078878:
------------------------------------------------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 10.10.2.206:5062;branch=z9hG4bK1786789569;rport=2048;alias
From: <sip:1001@domain.de:5061>;tag=2027020172
To: <sip:1001@domain.de:5061>;tag=gD6H4BF5v4rXN
Call-ID: 824582011-5062-1@10.10.2.206
CSeq: 2000 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="domain.de", nonce="a8a5dca0-7836-11de-b5f7-7f0e22d79464", algorithm=MD5, qop="auth"
Content-Length: 0

Grandstream HandyTone 503​

June 2009: This device is supposed to support TLS + SRTP - only ATA that is known to have it (?)

Summary:

  • Registration: FXS works with SIP over UDP, TCP. Works 2% of time with TLS.
  • Caller Id: FXS works with SIP over UDP
  • Call in/out: FXS works with SIP over UDP (call drops sometimes - see below). Does NOT work with TCP or TLS
  • Call waiting: not tested
  • Transfer calls: not tested
  • Park calls: not tested
  • SRTP FXS works
  • TLS FXS registration did not work 98% of time (without setup change), but can never make/receive calls
  • DTMF FXS works with FS IVR module

Details:

What works:

FXS SIP over UDP, RTP & SRTP, DTMF with IVR also found to work fine (with DTMF order set in devices as: RFC, INFO, in audio). Caller id also worked fine. Audio was quite clear with default codec order.

STUN, device behind NAT can make/receive calls and audio fine with FS (that also happened to be behind its own NAT (with UPnP support and using FS autonat feature))

What doesn't work:

FXS with SIP over TCP or TLS - registration works. However nothing else works - no calls can be made or received with SIP over TCP or TLS. Exact same setup eyebeam and snom work, so problem is not with FS.

Calls get dropped sometimes in between due to problem with HT 503 (not FS) sometimes. For a valid call and call id, HT 503 returns "SIP/2.0 481 Call Leg/Transaction Does Not Exist"

Please update wiki with any known results/fix for TCP/TLS and call drops. Also results with FXO.

January 2011 - Requires the latest HT503 firmware update to work properly. FXS works properly and FXO functions properly but registers by default as an extension not as a gateway. HT503 FXO default setting need the "Preferred DTMF method" setting set to "RFC2833" to connect. Also there are two ways to get the FXO to send calls: the first (not recommended) requires a dialplan that calls the fxo extension to get the dial tone and then the dialplan needs to dial the number. Able to make it work by using the following lines in the dial plan.

      <action application="queue_dtmf" data="W$1@100"/>
      <action application="bridge" data="user/1003"/>
      <action application="answer"/>

A better way to make the FXO connection is to create a gateway for the FXO but do not request registration (never was able to get it to register) using the 5062 port- example -

   <gateway name="ht503-1">
     <param name="realm" value="192.168.0.82:5062"/>
     <param name="proxy" value="192.168.0.82:5062"/>
     <param name="expire-seconds" value="800"/>
     <param name="register" value="false"/>
     <param name="retry-seconds" value="60"/>
     <param name="context" value="public"/>
   </gateway>

Then when you make a call you can create a dialplan like the one below, it sends local calls to the first HT503 that answers.

  <extension name="Local 7 digit calls" >
  <condition field="destination_number" expression="^(\d{7})$">
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge" data="sofia/gateway/ht503-2/$1@192.168.0.85:5062"/>
      <action application="bridge" data="sofia/gateway/ht503-1/$1@192.168.0.82:5062"/>
  </condition>
  </extension>

Using for Loudspeaker Paging/Overhead Paging The HT503 can be used as a gateway to a traditional PBX amplifier if the amplifier can connect to a CO or FX0 port (one that can provide line or battery power to the tip and ring) such as a Valcom V-200A.

  1. Configure a user for the HT503 to register to
  2. Configure the HT503
    1. SIP Server and login information
    2. Wait for Dial Tone (P3303) = No
    3. Stage Method (P3304) = 1
    4. Unconditional Call forward to PSTN (P796) = #
  3. If your amplifier has an RJ-11 port, connect a cable from the "Line" port on the HT503 to the RJ-11 port on the amplifier. If not, cut the end off an RJ-11 cable and expose enough of the red (tip) and green (ring) wires to connect to the tip and ring inputs on the amplifier.
  4. Make sure that line or battery power on the amplifier is "turned on"
  5. Attach the speaker
  6. Power all devices
  7. Create a dialplan, conference, etc. entry that connects a call to the device you defined
  8. Your voice should be heard over the speaker

Check out the Intercom page for dialplan ideas.