Configuring a Mitel

Discussion in '3CX Phone System - General' started by gsaccountants, Apr 6, 2010.

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  1. gsaccountants

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    Edit by MFM: Please avoid using colors and bold where not necessary also please tag your logs in code and /code tags.

    hi there,

    thanks for taking the time to read.

    I have a mitel 5235 sip phones running v7 firmware that I am trying to configure as external extensions on my home network. now I know that Mitel was not one of your supported but I have managed to get 3 extensions working in the office and have these 2 phones for working at home. If it is possible to get it working properly then I would be most grateful otherwise it looks like we will be picking up a Mitel2000 system instead. My issue is one way audio on the external extension. They can hear me, but I cannot hear them.

    I have opened all ports and run the firewall checker which passes successfully on the pbx server.

    i have taken opened all ports on my Netgear router at home, after trying various combinations of port openings 5060, 5090, 7000-7100 (i changed the end of local port numbering on my pbx server), 9000-9050 and the stun 3489.

    This is a log of me trying to call the mailbox (i changed the default from 999 to 500 as 999 is our police and emergency number). The remote extension number, authorisation, logon etc is all 201. The pbx is on SBS2003 at 87.244.76.5 with ports forwarded to internal IP address 192.168.1.6. The IP address of the phone at home is 192.168.1.70


    Code:
    10:38:46.246  [CM503008]: Call(17): Call is terminated
    10:38:37.184  Session 12387 of leg C:17.1 is confirmed
    10:38:36.903  [CM503007]: Call(17): Device joined: sip:500@127.0.0.1:40600;rinstance=e7977758436fcb1d
    10:38:36.903  [CM503007]: Call(17): Device joined: sip:201@82.112.132.230:5060;transport=UDP
    10:38:36.887  [MS210003] C:17.1:Answer provided. Connection(transcoding mode[unsecure]):87.244.76.5:9030(9031)
    10:38:36.887  [MS210001] C:17.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40636(40637)
    10:38:36.887  Remote SDP is set for legC:17.2
    10:38:36.887  [CM505001]: Ext.500: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:500@127.0.0.1:5060]
    10:38:36.887  [CM503002]: Call(17): Alerting sip:500@127.0.0.1:40600;rinstance=e7977758436fcb1d
    10:38:36.731  [CM503025]: Call(17): Calling Ext:Ext.500@[Dev:sip:500@127.0.0.1:40600;rinstance=e7977758436fcb1d]
    10:38:36.731  [MS210002] C:17.2:Offer provided. Connection(transcoding mode): 127.0.0.1:7040(7041)
    10:38:36.684  [CM503004]: Call(17): Route 1: Ext:Ext.500@[Dev:sip:500@127.0.0.1:40600;rinstance=e7977758436fcb1d]
    10:38:36.684  [CM503010]: Making route(s) to <sip:500@87.244.76.5>
    10:38:36.684  [MS210000] C:17.1:Offer received. RTP connection: 192.168.1.70:9018(9019)
    10:38:36.684  Remote SDP is set for legC:17.1
    10:38:36.668  [CM505001]: Ext.201: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mitel-5235-SIP-Phone 07.02.00.04 08000F2980DB] PBX contact: [sip:201@87.244.76.5:5060]
    10:38:36.668  [CM503001]: Call(17): Incoming call from Ext.201 to <sip:500@87.244.76.5>
    10:38:36.668  [CM500002]: Info on incoming INVITE:
      INVITE sip:500@87.244.76.5 SIP/2.0
      Via: SIP/2.0/UDP 82.112.132.230:5060;rport=5060;branch=z9hG4bKf2d05b2
      Max-Forwards: 70
      Contact: "Alun Griffiths - home"<sip:201@82.112.132.230:5060;transport=UDP>
      To: <sip:500@87.244.76.5>
      From: "Alun Griffiths - home"<sip:201@87.244.76.5>;tag=4bbb0f27-34c-e4a11c2
      Call-ID: f270000-69f64ff4@87.244.76.5
      CSeq: 1758813030 INVITE
      Subject: sip phone call
      Session-Expires: 1800
      Min-SE: 90
      Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, NOTIFY, PRACK, UPDATE
      Proxy-Authorization: Digest username="201",realm="3CXPhoneSystem",nonce="414d535c01cb921c56:3508cfed63e85c4a37ec03d7e4ac63c1",uri="sip:500@87.244.76.5",response="81b4dd193874b1ddfb69590b28ddc4fc",opaque="",algorithm=MD5
      Supported: timer, 100rel, replaces
      User-Agent: Mitel-5235-SIP-Phone 07.02.00.04 08000F2980DB
      Allow-Events: talk, hold, conference
      Content-Length: 0

    This is the log from the home router/firewall when making the call.

    Code:
    Tue, 2010-04-06 09:38:37 - UDP Packet - Source:192.168.1.70,9018 Destination:87.244.76.5,9030 - [3cx rtp rule match]
    Tue, 2010-04-06 09:38:37 - UDP Packet - Source:192.168.1.70,5060 Destination:87.244.76.5,5060 - [3cx rule match]
    Tue, 2010-04-06 09:38:37 - UDP Packet - Source:192.168.1.70,9018 Destination:87.244.76.5,9030 - [3cx rtp rule match]
    Tue, 2010-04-06 09:38:40 - UDP Packet - Source:192.168.1.70,5060 Destination:87.244.76.5,5060 - [3cx rule match]
    Tue, 2010-04-06 09:38:40 - UDP Packet - Source:192.168.1.70,9018 Destination:87.244.76.5,9030 - [3cx rtp rule match]
    Tue, 2010-04-06 09:38:46 - UDP Packet - Source:192.168.1.70,5060 Destination:87.244.76.5,5060 - [3cx rule match]
    Tue, 2010-04-06 09:39:20 - UDP Packet - Source:192.168.1.70,5060 Destination:87.244.76.5,5060 - [3cx rule match]
    Tue, 2010-04-06 09:44:21 - UDP Packet - Source:192.168.1.70,5060 Destination:87.244.76.5,5060 - [3cx rule match]
    
    So it looks like I am not recieving the packet on port 9019????

    This is the configuration on the Mitel 5235.
    Code:
    <Parameter Model="5235">
       <dhcpenable>0</dhcpenable>
       <tftp_config>0</tftp_config>
       <pppoe_enable>0</pppoe_enable>
       <tftp_task_enable>1</tftp_task_enable>
       <boot_version>02.03.02.02</boot_version>
       <image_version>R7.2.07.02.00.04</image_version>
       <tftp_upgrade>0</tftp_upgrade>
       <http_upgrade>0</http_upgrade>
       <outbound_state>0</outbound_state>
       <local_sip_port>5060</local_sip_port>
       <tls_port>5061</tls_port>
       <tos>0</tos>
       <e802_priority>-1</e802_priority>
       <vlan_id>-1</vlan_id>
       <host_name>3cx</host_name>
       <domain>plato.gsaccountants.com</domain>
       <addr_type>0</addr_type>
       <hot_line>0</hot_line>
       <hot_address>operator@example.com</hot_address>
       <hot_addr_type>0</hot_addr_type>
       <tls_private_url></tls_private_url>
       <tls_certificate_url></tls_certificate_url>
       <tls_ca_cert_url></tls_ca_cert_url>
       <tls_root_cert_url></tls_root_cert_url>
       <tls_certificate></tls_certificate>
       <tls_ca_cert></tls_ca_cert>
       <tls_root_cert></tls_root_cert>
       <poundkeydial>1</poundkeydial>
       <dialtonekey>12</dialtonekey>
       <htmlpuseraccess>1</htmlpuseraccess>
       <remote_reboot>1</remote_reboot>
       <checkpeercert>0</checkpeercert>
       <sipkeepalive>1</sipkeepalive>
       <rss_feed>'GS Accountants'</rss_feed>
       <blf_pickup>*98</blf_pickup>
       <host_ip>135.199.77.12</host_ip>
       <video_ip>135.199.77.12</video_ip>
       <sntp>pool.ntp.org</sntp>
       <time_zone>0</time_zone>
       <auth_method>1</auth_method>
       <register_expire>7200</register_expire>
       <session_timer>1800</session_timer>
       <emerg_number></emerg_number>
       <emerg_ip>0.0.0.0</emerg_ip>
       <emerg_port>5060</emerg_port>
       <audio_codec>5</audio_codec>
       <audio_pkt_size>20</audio_pkt_size>
       <video_codec>0</video_codec>
       <dtmf_type>0</dtmf_type>
       <dtmf_payload>101</dtmf_payload>
       <advisorymsg>0</advisorymsg>
       <reasons>0</reasons>
       <other_reason></other_reason>
       <do_not_disturb>0</do_not_disturb>
       <noans_fwd_mode>0</noans_fwd_mode>
       <try_ring_nums>10</try_ring_nums>
       <noans_fwd_addr></noans_fwd_addr>
       <beep_on_hold>1</beep_on_hold>
       <on_hold_alert>60</on_hold_alert>
       <system_mode>0</system_mode>
       <pppoe_login></pppoe_login>
       <pppoe_passwd>******</pppoe_passwd>
       <callCountIn>10</callCountIn>
       <callCountOut>35</callCountOut>
       <discovery>0</discovery>
       <pbIndex>0</pbIndex>
       <adminId>admin</adminId>
       <admin_dispname>Administrator</admin_dispname>
       <admin_passwd>923e325e16617477e457f6a468a2d6df</admin_passwd>
       <busy_fwd_mode>0</busy_fwd_mode>
       <busy_fwd_addr></busy_fwd_addr>
       <always_fwd_mode>0</always_fwd_mode>
       <always_fwd_addr></always_fwd_addr>
       <pcport>0</pcport>
       <lanport>0</lanport>
       <lcd>17</lcd>
       <lcd_brightness>5</lcd_brightness>
       <rdkw1></rdkw1>
       <rdringtype1>0</rdringtype1>
       <rdvmail1>0</rdvmail1>
       <rdblock1>0</rdblock1>
       <rdkw2></rdkw2>
       <rdringtype2>0</rdringtype2>
       <rdvmail2>0</rdvmail2>
       <rdblock2>0</rdblock2>
       <rdkw3></rdkw3>
       <rdringtype3>0</rdringtype3>
       <rdvmail3>0</rdvmail3>
       <rdblock3>0</rdblock3>
       <rdkw4></rdkw4>
       <rdringtype4>0</rdringtype4>
       <rdvmail4>0</rdvmail4>
       <rdblock4>0</rdblock4>
       <rdkw5></rdkw5>
       <rdringtype5>0</rdringtype5>
       <rdvmail5>0</rdvmail5>
       <rdblock5>0</rdblock5>
       <dtringtype1>0</dtringtype1>
       <dtringtype2>0</dtringtype2>
       <dtringtype3>0</dtringtype3>
       <dtringtype4>0</dtringtype4>
       <dtringtype5>0</dtringtype5>
       <dtringtype6>0</dtringtype6>
       <dtringtype7>0</dtringtype7>
       <dtringtype8>0</dtringtype8>
       <dtringtype9>0</dtringtype9>
       <dtringtype10>0</dtringtype10>
       <dtringtype11>0</dtringtype11>
       <dtringtype12>0</dtringtype12>
       <http_task_enable>1</http_task_enable>
       <https_task_enable>0</https_task_enable>
       <httpport>80</httpport>
       <httpsport>443</httpsport>
       <telnet_task_enable>1</telnet_task_enable>
       <voicemail_ringnum>4</voicemail_ringnum>
       <gruu_ctl>0</gruu_ctl>
       <proxyrequire_ctl>1</proxyrequire_ctl>
       <fwEnable>0</fwEnable>
       <fwWanurl></fwWanurl>
       <sym_udp>0</sym_udp>
       <stunip>stun.3cx.com</stunip>
       <fwWanDurl>87.244.76.5</fwWanDurl>
       <fwMode>0</fwMode>
       <start_port>9000</start_port>
       <end_port>9049</end_port>
       <multi_user_enable>0</multi_user_enable>
       <upgrade>0</upgrade>
       <bksrvtm>3</bksrvtm>
       <ntfcfg>0</ntfcfg>
       <lancode>en_GB</lancode>
       <tonecode>US</tonecode>
       <dsmode>1</dsmode>
       <dsmonth>3</dsmonth>
       <dsweek>2</dsweek>
       <dsday>1</dsday>
       <dsemonth>11</dsemonth>
       <dseweek>1</dseweek>
       <dseday>1</dseday>
       <ds_transition_time>2</ds_transition_time>
       <flashVer>201</flashVer>
       <http_download></http_download>
       <tftp>192.168.1.11</tftp>
       <downloadtype>1</downloadtype>
       <dialpl></dialpl>
       <gtEnable>0</gtEnable>
       <dtimer>3</dtimer>
       <autoanswer>0</autoanswer>
       <ringPitch>0</ringPitch>
       <keysys_enable>0</keysys_enable>
       <snmp>0</snmp>
       <srtp>0</srtp>
       <pkDescription>
          <Key Line="25" Fea="6" Des="Line  1" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="26" Fea="7" Des="Line 2" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="27" Fea="8" Des="Line 3" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="28" Fea="9" Des="Line 4" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="29" Fea="10" Des="Line 5" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="30" Fea="11" Des="Line 6" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="31" Fea="12" Des="Line 7" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="32" Fea="13" Des="Line 8" Addr="" Addr2="" Mode="1" Mode2="1" UserID="201"></Key>
          <Key Line="33" Fea="2" Des="Call logs" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="34" Fea="3" Des="Advisory            " Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="35" Fea="4" Des="Headset             " Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="36" Fea="19" Des="Weather" Addr="http://weather.yahooapis.com/forecastrss?p=CAXX0343" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="37" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="38" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="39" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="40" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="41" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="42" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="43" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="44" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="45" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="46" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="47" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
          <Key Line="48" Fea="0" Des="" Addr="" Addr2="" Mode="1" Mode2="1" UserID=""></Key>
       </pkDescription>
       <webdialurl></webdialurl>
       <cw_tone>1</cw_tone>
       <missedcallsctl>1</missedcallsctl>
       <callforwardctl>1</callforwardctl>
       <lcdbacklightctl>1</lcdbacklightctl>
       <time_format>1</time_format>
       <csta_enable>0</csta_enable>
       <csta_passwd>******</csta_passwd>
       <cfg_poll_timer>1440</cfg_poll_timer>
       <reboot_phone>1</reboot_phone>
       <firmware_timer>1440</firmware_timer>
       <firmware_abs_timer_hr>23</firmware_abs_timer_hr>
       <firmware_abs_timer_min>59</firmware_abs_timer_min>
       <firmware_abs_enable>1</firmware_abs_enable>
       <installer_passcode></installer_passcode>
       <user_passwd>5d41402abc4b2a76b9719d911017c592</user_passwd>
       <web_logo1>&lt;img src="http://edocs.mitel.com/DB/SIPhelp2_files/mitel_logo.gif" alt="Mitel" width="143" height="43" hspace="0" align="left" /&gt;</web_logo1>
       <sip_mode>sip</sip_mode>
       <voicemail_key>87.244.76.5</voicemail_key>
       <html_enable>1</html_enable>
       <html_filename></html_filename>
       <facDef>90</facDef>
       <user_list>
          <User State="1" ID="201" DispName="Alun Griffiths     - home" Pwd="201" AuthName="201" Realm="" RegSvr="87.244.76.5" RegPort="5060" RegScheme="2" ProxySvr="87.244.76.5" ProxyPort="5060" ProxyScheme="2" VMSvr="87.244.76.5" VMPort="5060" VMScheme="2" OutSvr="87.244.76.5" OutPort="5060" OutCtr="1" Ring="1" Line="0" EventSvr="" EventPort="5060" EventScheme="2" NatMode="1" NatType="stun" NatIp="1" BlfGroup=""></User>
       </user_list>
       <ipadr>192.168.1.70</ipadr>
       <ipgateway>192.168.1.1</ipgateway>
       <ipmask>255.255.255.0</ipmask>
       <dhcpLease>0</dhcpLease>
       <dhcpT1>0</dhcpT1>
       <dhcpT2>0</dhcpT2>
       <dhcpSrv>0.0.0.0</dhcpSrv>
       <ipdns>192.168.1.1</ipdns>
       <ipscddns>212.9.0.135</ipscddns>
       <cfg_version>R7.2</cfg_version>
       <answered_calls>******</answered_calls>
       <missed_calls>******</missed_calls>
       <made_calls>******</made_calls>
       </Parameter>
    Any help or ideas gratefully received! I have spent all the Easter weekend on this so hope there is just something simple.
     
  2. mfm

    mfm Active Member

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    Hi,

    Considering STUN is switched on, and you are still getting one way audio, I recommend you run a wireshark capture to see what IP addresses are being passed. I also advise you try using the SIP proxy manager and advise what your outcome is with that.
     
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  3. gsaccountants

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    Thanks Mark.

    I did try the sip proxy manager.

    i have run the wireshark thing.
    I have absolutely no idea what to do with it now!
    what should i be looking at?

    Thanks,
    al.
     
  4. leejor

    leejor Well-Known Member

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    Is each set programmed with a unique port number? I've found that is usually isn't necessary to forward ports at the "remote" end. I have found (at times) one way audio problems when more than one device is behind a remote router, I have assumed that it is an issue with the particular make/model of router.
     
  5. gsaccountants

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    Hi Lee,
    Thanks for chipping in!
    by each set I suppose you mean each phone? I only have one phone online. I am trying to set up two but the other will be used at a different remote location and I only have one on at a time - i only have one PoE connector at home so can only ever program one phone at a time.
    Al.
     
  6. SY

    SY Well-Known Member
    3CX Support

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    Hint.
    For some reason, remote phone specifies its own local IP/port instead of external IP/port of its network. Strange, the phone does it for SIP related information (see "contact:" header of incoming invite, there is 82.112.132.230:5060) but doesn't do it for RTP/SDP part...
     
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  7. gsaccountants

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    thanks stepan. i will keep tweeking! i just think i have got to the point where i have been staring at this for so long if it was right in front of me now i wouldn't see it!
     
  8. leejor

    leejor Well-Known Member

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    I've set up some Mitel 5220's with the latest SIP firmware, and they work fine (for most functions) remotely. I don't recall too many STUN options, it's been a number of months since I set up the phones. Have you tried running the set behind a different router? Are you using DHCP or a fixed IP on the set?
     
  9. gsaccountants

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    Hi lee,
    i have tried both dhcp and fixed but only have a Netgear DG834G to try and work with.
    al.
     
  10. leejor

    leejor Well-Known Member

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    Have you tried turning off STUN and running it with a public IP? No router.
     
  11. gsaccountants

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    hi, not i haven't - i'm not sure how to do that. would love some advice as i'm down another week!
     
  12. leejor

    leejor Well-Known Member

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    At the remote end, if the router is plugged into a modem (Looks like your Router is an all-in-one, so you'd have to get your hands on another modem). Turn off STUN, on the set, unplug the router from the modem and plug in the set. This should allow the set to get a public IP from the ISP. If everything works, audio both ways, then the router at that end might be causing problems. At that point, If it were me, I would borrow another router, to try, to see if it made a difference. Some routers have been known to have problems with SIP or require some settings to be changed to properly accommodate SIP.

    Here is someone else with the same issue, same router...
    http://www.getvoipsolution.com/thread-2042-1-1.html

    This one seen to think it's a firmware issue...
    http://forum1.netgear.com/showthread.php?t=20531

    Here is the set-up of another VoIP service on your model modem...
    http://www.jj-telecoms.com/Dogandbone/DogandBoneFAQ%27s.htm

    Good luck.
     
  13. gsaccountants

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    purely for effort that has to be worth a beer!
    many thanks. I shall give it another weekend then burn the router!
     
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