cant hear caller after few secs, they hear us

Discussion in 'General' started by bfalzarano, Jun 29, 2009.

  1. bfalzarano

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    For some inbound calls we cannot hear the caller after a few seconds however they can still hear us.

    We are using a new Sangoma AFT-102 dual E1/T1 card and we tried also just using a new Sangoma AFT-200 FXO/FXS card. The same thing occurs for some inbound calls on the fxo. After a few seconds we cannot hear the caller anymore but the caller can still hear us.

    One way voice.

    I read a post by XORCOM suggesting a dahdi update is required, but not sure if this is applicable.
    yum update dahdi kernel-module-dahdi

    Here is asterisk module and module dahdi versions:
    [root@elastix asterisk]# rpm -qa | grep asterisk
    asterisk-perl-0.10-1
    elastix-asterisk-sounds-1.2.2-2
    asterisk-1.4.24.1-1
    asterisk-devel-1.4.24.1-1
    asterisk-sounds-fr-1.4.24.1-1
    asterisk-addons-1.4.7-4
    asterisk-sounds-es-1.4.24.1-1
    [root@elastix asterisk]# rpm -qa |grep module-dahdi
    kernel-module-dahdi-2.1.0.4-9_2.6.18_92.1.22.el5
    kernel-module-dahdi-devel-2.1.0.4-9
    kernel-module-dahdi-xen-2.1.0.4-9_2.6.18_92.1.22.el5
    [root@elastix asterisk]#


    How can I begin to troubleshoot why this is occurring?

    Thank you,
    bfalzarano
     
  2. bfalzarano

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    To Add:

    I setup a voice vlan separate from the data vlan and turned off verbose and debug logging in asterisk.
    We are using Cisco 7940 phones and the problem is still occurring. I provisioned an Aastra 57i and have been testing with that phone and the problem has not occurred so far. I will continue testing tomorrow.

    Fonality is all over this issue with their sales reps calling me to migrate to their TrixBox system. Their comments were that Elastix uses the newer and more unstable versions of Asterisk. Whereas their system uses more stable (albeit older) versions of Asterisk, but they back-port the new features into their system.

    In either case I am continuing to troubleshoot this issue by trying to determine if this problem is related to the Cisco phones or the Elastix system. I have not seen any replies to my post so I wanted to provide a post that shows how I am troubleshooting this issue so far in case anyone else is having a similar problem. If anyone has some comments on this issue please reply to my post.
     
  3. bfalzarano

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    Everyone's feedback is overwhelming : )

    Update 2:

    Problem appears to be related to the Cisco phones. I will be deploying different model phones tomorrow to confirm the problem is with the Cisco 7940 phones. I also modified rtp.conf start and end ports from 16384 to 32766. This matches the rtp media ports specified in the SipDefault.cnf file. Not sure if this will help. Asterisk search on one-way audio also suggests setting the canreinvite to no.

    en cierro bocas no entra moscas
     
  4. virusbcn

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    Sorry if you use a softphone do you have the same problem ???
    I like more the Aastra phones, 0 problems and 100x100 satisfaction ;)
     
  5. bfalzarano

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    We have not had the same one way voice problem with softphones and the Aastra phone has also been working fine. Last night I changed the \\etc\\asterisk\\rtp.conf rtp begin and end range to 16384 and 32766 to match the Cisco phone rtp range and so far this morning no problems have been encountered. It is still too early for me to tell if the problem is entirely cleared up.

    I very much agree with you that the Aastra phones out of the box are much easier to provision and configure and work very well. Thank you your reply.
     

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