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isaac

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Posts posted by isaac

  1. Pressing 5 (with time/date envelope) when listening to VM causes CS410 to Crash

    in Embedded

    Posted

    Ouch. The white edition does not have the features that you need to run version 3.

     

    Check the /etc/network/interfaces file. Make sure that there is no "eth2".

     

    No eth2:

     

    #This file was automatically generated by the IP PBX.

    auto lo

    iface lo inet loopback

     

    auto eth0

    iface eth0 inet static

    address 70.x.x.x

    network 70.x.x.x

    netmask 255.255.255.192

    broadcast 70.x.x.x

    gateway 70.x.x.x

     

    auto eth1

    iface eth1 inet static

    address 1.1.1.1

    network 1.1.1.0

    netmask 255.255.255.0

    broadcast 1.1.1.255

     

    Log File shows:

     

     

    Logfile

    Clear or Reload the log.

    [1] 2009/01/05 16:31:15: Starting up version 2.1.0.2115

    [8] 2009/01/05 16:31:15: Route: eth0 462a4400 ffffffc0

    [8] 2009/01/05 16:31:15: Route: eth1 01010100 ffffff00

    [8] 2009/01/05 16:31:15: Default Route uses 70.x.x.x

    [7] 2009/01/05 16:31:15: Found time zones AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET CST CAT IST GMT

    [1] 2009/01/05 16:31:15: Working Directory is /pbx

    [5] 2009/01/05 16:31:17: Starting threads

    [7] 2009/01/05 16:31:17: UDP: Opening socket

    [0] 2009/01/05 16:31:17: UDP: bind() to port 5060 failed

    [0] 2009/01/05 16:31:17: FATAL: Could not open UDP port 5060 for SIP

    [7] 2009/01/05 16:31:17: Opening TCP socket on port 5060

    [7] 2009/01/05 16:31:17: Opening TCP socket on port 5061

     

    Not sure why???

  2. Pressing 5 (with time/date envelope) when listening to VM causes CS410 to Crash

    in Embedded

    Posted

    Oh, maybe you can upgrade to 3.1.2.3120. I would suggest you make a backup first (log in to the system and tar the /pbx directory), so that if anything should go wrong you can always move back to that version.

    3.1.2.3120 created many problems on inbound and outbound calling. It seems like dialplan now automatically strips 1 from NA numbers and I cannot get it to send numbers with the +1. In addition, inbound calls are not playing autoattendant file.

     

     

    I restored from TAR and now NO phones can register nor can I make any calls to PBX. Please help!

     

    I have 1st gen white CS410.

  5. 2.1.0 RC2 image available

    in New Features/Versions

    Posted

    I know the DTMF is being sent because when we place a call outside of PBXnsIP the receiving end does detect the tones however tones to the IVR within PBXnSIP are undetected. Jitter is a non issue since this is on a LAN with minimal traffic.

     

     

     

    That is really weired. I could hear the DTMF in the audio stream, it did not sound too bad.

     

    Of course, jitter in the DTMF tones is very bad for the detection (that is why they invented out of band!), but I guess that does not seem to be the problem here.

     

    [shrugging shoulders]

  7. 2.1.0 RC2 image available

    in New Features/Versions

    Posted

    Carrier supports Out Of Band DTMF for G729 but not for G711. Carrier require inband DTMF for G711.

     

    Inband detection fails. Transcoding works.

     

     

    Okay.

     

    I still don't understand why G.729 would improve that situation... DTMF was not featured in this codec (it must be out of band then).

     

    Are we talking about DTMF detection or DTMF transcoding here? Detection should be okay or better than previois versions, transcoding requires that the PBX falls back to RTP disassembly, which it should do automatically (there should be a log message about that).

  9. 2.1.0 RC2 image available

    in New Features/Versions

    Posted

    I have no choice but to use inband unless we can get G729 support! Carrier can only support G711/inband.

     

    Inband was working on previous versions so something changed on newer version.

     

    Inband: If you don't have to, then don't use inband. Especially on the embedded system, it cost a lot of performance to analyze the audio streams.

     

    G729 pass through. Well, it might be possible to pass G729 through, but the big problem is what happens if someone hits the hold button or parks the call. The PBX then is not able to render audio... Therefore, G729 is still still a problem.

     

    G722 is possible there, but you must edit the pbx.xml file and add the codec to the preference list (codec number is 9). We did some tests with the codec, tests show that it works-however it really has limited value as the preferred PSTN termination is FXO and that is quite the opposite of wideband audio.

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