G.729 calls to IVR/VM/etc

Discussion in '3CX Phone System - General' started by carolinainnovative, Jun 25, 2010.

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  1. carolinainnovative

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    In an effort to reduce bandwidth usage, I've set all of my extensions to use G.729. My VOIP provider supports it, phones all support it - but for some reason, when I restrict G.711u, I can't connect to voicemail, ivr or anything that the pbx does itself.

    Is there a way to get the IVR/VM/etc to use G.729? All of my extensions are remote.

    Running 3cx V9 Beta 2

    Thanks guys!
     
  2. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    no, vbox and ivr works as well in free edition. due to this codec 729 is not present in the license and it would fail for many reasons as well.

    IVR,Vbox,Queues are only work in G711
    Load for the system is the other aspect.
    Cant be changed by the user.
     
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  3. carolinainnovative

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    I don't see why you can't let G.729 work on those things for LICENSED users. I understand free - thats the same reason 3cxPhone doesn't have h.264 or G.729 - they cost money - but LICENSED users may WANT G.729 throughout their system... processor overhead bedamned...

    Just a thought...
     
  4. hgcomp

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    I neeeeeeeeed g729 to remote extensionssssssssssssssssss... !!!
    Thanks
     
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  5. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    1. Tick "PBX delivers Audio" for that EXT or Provider
    2. Change/Add in PhoneSystem.ini the beginning as shown, rest leave at it is (make Backup of file)
    Code:
    [General]
    AppPath=C:\Program Files\3CX PhoneSystem\
    WebServer=""
    FirstRun=False
    pbxSLNIC=127.0.0.1
    LogLevel=2
    ;Msusetickcount=1
    [b]externalcodecs=G729 PCMU PCMA GSM
    localcodecs=PCMU PCMA G729 GSM[/b]
    
    3. Configure your phone to prefer G729 and configure PCMU as fall back

    Thx
     
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  6. carolinainnovative

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    Stefan,

    I did that and it seems my IVR is still seems to insist on using G711u as a codec.

    see log. Doesn't mention the codec, but basically I get the infamous "no rtp packets received"

    Code:
    
    22:59:01.017  [MS105000] C:11.2: No RTP packets were received:remoteAddr=127.0.0.1:40626,extAddr=0.0.0.0:0,localAddr=127.0.0.1:7028
    22:58:59.704  [MS105000] C:11.1: No RTP packets were received:remoteAddr=70.63.254.171:9006,extAddr=0.0.0.0:0,localAddr=64.x.x.x:9012
    22:58:59.654  [CM503008]: Call(11): Call is terminated
    22:58:59.624  Session 587 of leg C:11.1 is confirmed
    22:58:59.509  [CM503007]: Call(11): Device joined: sip:999@127.0.0.1:40600;rinstance=2aae7af5f5a56d71
    22:58:59.500  [CM503007]: Call(11): Device joined: sip:101@70.63.254.171:5060
    22:58:59.497  [MS210005] C:11.1:Answer provided. Connection(proxy mode):64.128.46.61:9012(9013)
    22:58:59.490  [MS210001] C:11.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40626(40627)
    22:58:59.488  Remote SDP is set for legC:11.2
    22:58:59.481  [CM505001]: Ext.999: 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:999@127.0.0.1:5060]
    22:58:59.480  [CM503002]: Call(11): Alerting sip:999@127.0.0.1:40600;rinstance=2aae7af5f5a56d71
    22:58:59.323  [CM503025]: Call(11): Calling Ext:Ext.999@[Dev:sip:999@127.0.0.1:40600;rinstance=2aae7af5f5a56d71]
    22:58:59.320  [MS210004] C:11.2:Offer provided. Connection(proxy mode): 127.0.0.1:7028(7029)
    22:58:59.289  [CM503004]: Call(11): Route 1: Ext:Ext.999@[Dev:sip:999@127.0.0.1:40600;rinstance=2aae7af5f5a56d71]
    22:58:59.287  [CM503010]: Making route(s) to <sip:999@mysipdomain.net>
    22:58:59.287  [MS210000] C:11.1:Offer received. RTP connection: 70.63.254.171:9006(9007)
    22:58:59.282  Remote SDP is set for legC:11.1
    22:58:59.273  [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [RTP300-3.1.22] PBX contact: [sip:101@sip.mysipdomain.net:5060]
    22:58:59.267  [CM503001]: Call(11): Incoming call from Ext.101 to <sip:999@mysipdomain.net>
    22:58:59.258  [CM500002]: Info on incoming INVITE:
      INVITE sip:999@sccltd.net SIP/2.0
      Via: SIP/2.0/UDP 70.63.254.171:5060;branch=z9hG4bK-46cbcab4
      Max-Forwards: 70
      Contact: "Chavous Camp"<sip:101@70.63.254.171:5060>
      To: <sip:999@mysipdomain.net>
      From: "Chavous Camp"<sip:101@sccltd.net>;tag=76598d11392a4b64o0
      Call-ID: 6bb76c55-6e6254f0@10.254.0.3
      CSeq: 102 INVITE
      Expires: 240
      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
      Proxy-Authorization: Digest username="ccamp",realm="3CXPhoneSystem",nonce="414d535c02617f7383:683de171586b11f7d7bfd781bd352094",uri="sip:999@mysipdomain.net",algorithm=MD5,response="aabf35d03cad079f0063ce409d79d651"
      Supported: x-sipura, replaces
      User-Agent: RTP300-3.1.22
      Content-Length: 0
      Remote-Party-ID: Chavous Camp <sip:101@mysipdomain.net>;screen=yes;party=calling
    
     
  7. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    sure the IVR will answer in PCMU ALWAYS
    But due to bound to media server the media server should convert now to G729 and then send it to an EXT or Provider
     
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  8. carolinainnovative

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    ideally, couldn't you guys get the ivr to use g.729 "natively" ? I understand it would be a paid-only feature and would require some work to handle the prompt sets (ie transcoding on the fly or re-encoding them as a whole and then storing an additional set for each codec), but seriously - I don't particularly want to use "pbx delivers audio" just to get G.729 to work properly.
     
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