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Sangoma A101D PSTN Gateway Unsupported Media

Discussion in '3CX Phone System - General' started by spike711, Sep 5, 2010.

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  1. spike711

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    Sangoma A101D PSTN Gateway Unsupported Media

    I have setup 3CX with Sangoma A101D card for fractal PRI. Incoming calls work fine but calls from certain devices will not call out. I receive "Unsupported Media" or other error.
    3CX Softphone = "Unsupported Media"
    Grandstream 2020 = "488 Not Acceptable Try different vocoder"
    ATA-186 (v3.2) = Works Perfectly
    Sorry, I don't have any other devices to check.
    All of them recieve inbound calls via DID perfectl.

    Thx,
    Spike
     
  2. spike711

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    Additional error Information from Netborder (Sangoma A101D Driver) & Software:

    Highest Error Level reached : YELLOW
    Last Error Received : call-id=1283664472-687500-4827-128 No routing rule matches the attributes of the incoming call. The call will be rejected. Please review your rules if it is unexpected. Detailed Call Information: sip.in.requestUri=sip:407XX249XX@127.0.0.1:5066 sip.in.requestUri.canonical=sip:407XX249XX@127.0.0.1:5066 sip.in.from.uri=sip:407XX985XX@127.0.0.1:5066 sip.in.from.uri.canonical=sip:407XX985XX@127.0.0.1:5066 sip.in.to.uri=sip:407XX249XX@127.0.0.1:5066 sip.in.to.uri.canonical=sip:407XX249XX@127.0.0.1:5066 sip.in.header.Via=SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=z9hG4bK-d8754z-78196668595c0e1f-1---d8754z- sip.in.header.CSeq=1 INVITE sip.in.header.Call-ID=ZTliZDE4MTYzODA2ZGMxZGRiNWMzNjFlZTg0NzBkZTU. sip.in.header.Content-Length=401 sip.in.header.Contact= media.rtp.stream0=sendrecv media.rtp.stream1=inactive transfer=false
     
  3. spike711

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    Here are the routing rules that were generated by the 3CX gateway wizard:

    <?xml version="1.0" encoding="utf-8"?>

    <routing_rules>



    <!-- SIP OUT TO sip:localhost:5060 -->

    <rule name="default_sip_out" outbound_interface="sip" qvalue="0.001">

    <condition param="transfer" expr="false"/>

    <condition param="pstn.in.channelName" expr=".*"/>

    <condition param="pstn.in.ani" expr="(.*)"/>

    <condition param="pstn.in.dnis" expr="([0-9]*)"/>

    <condition param="pstn.in.callerName" expr="([0-9a-zA-Z]+)"/>

    <out_leg name="" media_type="sendrecv">

    <!-- To modify the target SIP destination, just change the value below -->

    <param name="sip.out.requestUri" expr="sip:10004@127.0.0.1:5060"/>

    <param name="sip.out.from.uri" expr="sip:%0@GW_HOST_IP:GW_SIP_PORT"/>

    <param name="sip.out.from.displayName" expr="%2"/>

    <param name="sip.out.to.uri" expr="sip:%1@127.0.0.1:5060"/>

    <param name="sip.out.to.displayName" expr="%1"/>

    <param name="sip.out.transport" expr="udp"/>

    </out_leg>

    </rule>



    <!-- PSTN OUT -->

    <rule name="default_pstn_out" outbound_interface="pstn" qvalue="0.001">

    <condition param="transfer" expr="false"/>

    <condition param="sip.in.requestUri.canonical" expr="sip:([0-9]+)@GW_HOST_IP:GW_SIP_PORT"/>

    <condition param="sip.in.from.displayName" expr="([0-9]+)"/>
    <out_leg name="" media_type="sendrecv">

    <param name="pstn.out.phoneNumber" expr="%0"/>

    <param name="pstn.out.deviceGroup" expr="default"/>

    <param name="pstn.out.cpa.enable" expr="false"/>

    <param name="pstn.out.ani" expr="%1"/>

    </out_leg>

    </rule>



    <rule name="redirect_to_sip" outbound_interface="sip" qvalue="0.1">

    <condition param="sip.in.redirect.Contact" expr="(.*)"/>

    <out_leg name="" media_type="sendrecv">

    <param name="sip.out.redirect.Contact" expr="%0"/>

    </out_leg>

    </rule>



    <rule name="redirect_to_pstn" outbound_interface="pstn" qvalue="0.2">

    <condition param="sip.in.redirect.Contact" expr="^Contact:.*sip:([0-9]+)@GW_HOST_IP:GW_SIP_PORT"/>

    <out_leg name="" media_type="sendrecv">

    <param name="pstn.out.phoneNumber" expr="%0"/>

    <param name="pstn.out.deviceGroup" expr="default"/>

    </out_leg>

    </rule>



    <!-- SIP to SIP transfer, results in a SIP bridged transfer on the gateway if

    transfer target is the gateway and no other transfer rule matches -->

    <rule name="sip_to_sip_transfer" outbound_interface="sip" qvalue="0.001">

    <condition param="transfer" expr="true"/>

    <condition param="sip.in.referTo" expr="(?U)(.*)(?:;|\?|$)"/>

    <condition param="sip.in.from.uri" expr="(.*)"/>

    <out_leg name="" media_type="sendrecv">

    <param name="sip.out.requestUri" expr="%0"/>

    <!-- echoing "from" fields of the REFER request -->

    <param name="sip.out.from.uri" expr="%1"/>

    </out_leg>

    </rule>



    <!-- PSTN (CTBus/SCBus/Internal audio switch) bridge -->

    <rule name="default_pstn_bridge_transfer" outbound_interface="pstn" qvalue="0.01">

    <condition param="transfer" expr="true"/>

    <condition param="sip.in.referTo.canonical" expr="sip:([0-9]+)@GW_HOST_IP:GW_SIP_PORT"/>

    <out_leg name="" media_type="sendrecv">

    <param name="pstn.out.phoneNumber" expr="%0"/>

    <param name="pstn.out.deviceGroup" expr="default"/>

    <param name="pstn.out.transfer.type" expr="2channel-bridged"/>

    <param name="pstn.out.transfer.supervision" expr="connect"/>

    <param name="pstn.out.transfer.timeoutMs" expr="30000"/>

    <param name="paraxip.gw.ringTimeoutMs" expr="0"/>

    <param name="paraxip.gw.connectionTimeoutMs" expr="0"/>

    </out_leg>

    </rule>



    </routing_rules>
     
  4. spike711

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    Here is the server log for calls from 3 different devices. One works the other 2 fail:
    Call WORKS on ATA-186:
    17:21:21.515 [CM503008]: Call(79): Call is terminated
    17:21:13.312 [CM503002]: Call(79): Alerting sip:10004@127.0.0.1:5066
    17:21:11.578 [CM503025]: Call(79): Calling Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:21:11.562 [CM503004]: Call(79): Route 1: Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:21:11.546 [CM503010]: Making route(s) to <sip:407xxxx973@10.0.0.157;user=phone>
    17:21:11.531 [CM505001]: Ext.108: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco ATA 186 v3.2.0 atasip (041111A)] PBX contact: [sip:108@10.0.0.157:5060]
    17:21:11.531 [CM503001]: Call(79): Incoming call from Ext.108 to <sip:407xxxx973@10.0.0.157;user=phone>

    Call Does NOT work Softphone:
    17:20:06.984 [CM503016]: Call(78): Attempt to reach <sip:407xxxx973@10.0.0.157:5060> failed. Reason: Not Acceptable HereReason Unknown
    17:20:06.984 [CM503003]: Call(78): Call to sip:407xxxx973@127.0.0.1:5066 has failed; Cause: 488 Not Acceptable Here; from IP:127.0.0.1:5066
    17:20:06.937 [CM503025]: Call(78): Calling Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:20:06.906 [CM503004]: Call(78): Route 1: Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:20:06.890 [CM503010]: Making route(s) to <sip:407xxxx973@10.0.0.157:5060>
    17:20:06.890 [CM505001]: Ext.102: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13679.0] PBX contact: [sip:102@10.0.0.157:5060]
    17:20:06.875 [CM503001]: Call(78): Incoming call from Ext.102 to <sip:407xxxx973@10.0.0.157:5060>

    Call Does NOT work Grandstream 2020:
    17:28:22.359 [CM503016]: Call(80): Attempt to reach <sip:407xxxx973@10.0.0.157> failed. Reason: Not Acceptable HereReason Unknown
    17:28:22.359 [CM503003]: Call(80): Call to sip:407xxxx973@127.0.0.1:5066 has failed; Cause: 488 Not Acceptable Here; from IP:127.0.0.1:5066
    17:28:22.296 [CM503025]: Call(80): Calling Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:28:22.281 [CM503004]: Call(80): Route 1: Unknown:407xxxx973@(Ln.10004@CenturyTelPRI)@[Dev:sip:10004@127.0.0.1:5066]
    17:28:22.250 [CM503010]: Making route(s) to <sip:407xxxx973@10.0.0.157>
    17:28:22.250 [CM505001]: Ext.110: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.4.3] PBX contact: [sip:110@10.0.0.157:5060]
    17:28:22.234 [CM503001]: Call(80): Incoming call from Ext.110 to <sip:407xxxx973@10.0.0.157>
     
  5. spike711

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    I ordered full version Netborder Express, uninstalled the trial version and installed the paid version. Now outbound is working on all phone types including caller id output which was not working before. But now inbound is not working! I'm getting fast busy which is what happens when a DID is not setup. I only had acces to the live PRI for about an hour so I didn't get to try many things. Any ideas would be appreciated.
    Thx,
    Spike
     
  6. brentrichburg

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    I contacted Sangoma Support. They made some changes to the routing-rules.xml and it started working. The outbound caller id is not sending the extension caller id, but I have not had time to send them the log files yet.

    Kudos to Sangoma. :!:
     
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