Can't dial out on PSTN via Grandstream HT-503 Gateway

Discussion in '3CX Phone System - General' started by mcbsys, Oct 8, 2008.

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  1. mcbsys

    mcbsys New Member

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    Hi,

    I'm setting up an HT-503 as a gateway for 3CX. I have firmware version 1.0.0.15. The FXO port is plugged in to an analog phone line, and the FSX port connects to an analog phone. The FXS side is configured as extension 10 on port 5060. The FXO side is configured as gateway 10000 on port 5062.

    1. I can receive calls okay: inbound FXO to 3CX to the FXS extension works.

    However I cannot place outbound calls (FXS or SIP to 3CX to FXO). For example, from a softphone, I dial a local phone number that belongs to me (one that is not connected to the HT-503). I hear ringing in the softphone, but the call is not actually going out to the PSTN--the receiving phone is not ringing. It’s like the HT-503 is “ringing” but not sending the call along to the PSTN. (In a sniffer, I can even see the SIP packet "Status: 180 Ringing" being sent from the HT-503 to 3CX, and 3CX passing that along to the softphone client.)

    The output at the end of this message shows the outbound call progress as reported by 3CX.

    How can I use the HT-503 to place outbound calls?

    2. Also, another (related?) problem: when I dial *00 from a phone connected to the HT-503 FXS port, I get a busy signal. This is supposed to pass me through to the PSTN line, bypassing 3CX.

    Thanks for any tips!

    Mark

    Time Function Message

    15:31:02.810 Call::Terminate [CM503008]: Call(14): Call is terminated

    15:30:38.667 Line::printEndpointInfo [CM505002]: Gateway:[Business] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 V1.1B 1.0.0.15] Transport: [sip:10.5.3.120:5060]

    15:30:38.667 CallCtrl::eek:nAnsweredCall [CM503002]: Call(14): Alerting sip:10000@10.5.3.121:5062

    15:30:38.467 MediaServerReporting::SetRemoteParty [MS210002] C:14.2:Offer provided. Connection(transcoding mode): 10.5.3.120:7052(7053)

    15:30:38.447 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(14): Calling: PSTNline:1234567@(Ln.10000@Business)@[Dev:sip:10000@10.5.3.121:5062]

    15:30:38.427 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:1234567@10.5.3.120]

    15:30:38.427 MediaServerReporting::SetRemoteParty [MS210000] C:14.1:Offer received. RTP connection: 10.5.3.18:49198(49199)

    15:30:38.427 CallLeg::setRemoteSdp Remote SDP is set for legC:14.1

    15:30:38.417 Extension::printEndpointInfo [CM505001]: Ext.50: Device info: Device Identified: [Man: SJ Labs;Mod: SJ Phone;Rev: 1.65] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [SJphone/1.65.377a (SJ Labs)] Transport: [sip:10.5.3.120:5060]

    15:30:38.417 CallCtrl::eek:nIncomingCall [CM503001]: Call(14): Incoming call from Ext.50 to [sip:1234567@10.5.3.120]

    15:30:38.397 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:

    INVITE sip:1234567@10.5.3.120 SIP/2.0
    Via: SIP/2.0/UDP 10.5.3.18;branch=z9hG4bK0a6e78120000011f48ed348e00003d8c000000db;rport=5060
    Max-Forwards: 70
    Contact: [sip:50@10.5.3.18]
    To: [sip:1234567@10.5.3.120]
    From: "unknown"[sip:50@10.5.3.120];tag=4cd1e9403c0
    Call-ID: 2160C86271CA4699BD4ED7D40A4D44600x0a6e7812
    CSeq: 2 INVITE
    Proxy-Authorization: Digest username="50",realm="3CXPhoneSystem",nonce="12867978637:14b67894db030c7bf1fe55179006fdfb",uri="sip:1234567@10.5.3.120",response="7520bfaa12172df4307e2103ffb9fcd2",algorithm=MD5
    Supported: replaces, norefersub, timer
    User-Agent: SJphone/1.65.377a (SJ Labs)
    Content-Length: 0
     
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  2. galal202

    galal202 New Member

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    I have this device and it works fine with 3CX except caller id
    so I think if you reset it to factory set and reconfig it again crefuly (use HT488 guide)
    it will work nice with you
     
  3. mcbsys

    mcbsys New Member

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    Thanks I may try that.

    One thing that I forgot to mention (and I don't think this was in the 488 guide) was that on the Basic Settings tab of the HT-503, I set "Unconditional Call Forward to VOIP" to forward to 10000 on the 3CX machine, port 5060. This is how I got the incoming PSTN calls to work. However, I haven't gotten the outbound calls to work yet.

    Mark
     
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  4. mcbsys

    mcbsys New Member

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    Okay I wiped the HT503 and started from scratch. Identical behavior: inbound PSTN calls work, but outbound PSTN calls don't.

    Dialing *00 from an FXS-attached phone still gives me a busy signal.

    Oddly, if I put an analog phone on the same jack as the FXO port, then call that number from an outside line, the phone keeps ringing even AFTER 3CX has stared processing the call.

    I'm beginning to wonder if the HT503 doesn't know how to properly take my line off hook.

    Mark
     
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  5. galal202

    galal202 New Member

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    disable wait for dial tone
     
  6. mcbsys

    mcbsys New Member

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    Gala, thanks for the suggestion. Unfortunately that doesn't help.

    I turned on SYSLOG and found an FXO_BAT_DROPPED event was happening even when I just tried dialing *00.

    That led me to this post:

    http://forums.grandstream.com/node/484

    and from that I got the idea of increasing the "Current Disconnect Threshold (ms)" value.

    Once I changed "Current Disconnect Threshold (ms)" from 100 to 200, I was able to use *00 on the FXS port to get to the PSTN, and I was able to dial out on the PSTN line from 3CX!

    BTW the SYSLOG still shows FXO_BAT_DROPPED even when *00 works, so I have no idea what that means, but at least it led me to a solution!

    Mark
     
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  7. galal202

    galal202 New Member

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    you're welcome
    mine works with 100 - the default
    where you are from?
    did cid works with you ?
     
  8. mcbsys

    mcbsys New Member

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    I'm in San Diego, California, about four blocks away from a Central Office. You would think this would be a high-quality line!

    Well on line 1, I have an alarm system cut-out installed before the HT-503.

    On line 2, I have DSL and a DSL filter before the HT-503.

    For whatever reason, both lines require 200 ms disconnect time.

    I see what you mean about the Caller ID not working. I just see the ID of the 3CX virtual extension for the gateway (10000).
     
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  9. creativeit

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    Re: Can't dial out on PSTN via "Unsupported IP Phone" HT-503

    Heya,

    I'm trying to get my HT503 to work for incoming calls. I'm a little new to VOIP and PBX, but I am at least somewhat technical as I do systems integration development (software dev). Any chance you could assist me set it up?

    PS: I'm in South Africa and I believe I have the correct cid settings based on info I found on another forum...

    Looking forward to hearing from you.

    Dave
     
  10. mcbsys

    mcbsys New Member

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    Re: Can't dial out on PSTN via "Unsupported IP Phone" HT-503

    Dave, not sure you are addressing me but I returned the HT503 when I couldn't get it to work right. That was over six years ago, so maybe they have cleared up some things in the firmware. You may have better luck starting a new thread.
     
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