[asterisk-bugs] [JIRA] Commented: (ASTERISK-20501) Distorted Audio if you Dial multiple numbers at once (DAHDI)

Sven Hirschmueller (JIRA) noreply at issues.asterisk.org
Mon Oct 8 06:44:27 CDT 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20501?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=198104#comment-198104 ] 

Sven Hirschmueller commented on ASTERISK-20501:
-----------------------------------------------

So, i attached a log file riped of a testing rig.

Phone with number 21 calls extension 13.
13 is following extension:

13 => {
  Ringing();
  Dial(dahdi/g9/11&dahdi/g10/20);
}

If i pick up via either phone 11 or phone 20 the audio is distorted.

Phone 11 is SPAN: 7 (Port 4 of the ISDN Card)
Phone 20 is SPAN: 8 (Port 5 of the ISDN Card)
Phone 21 is SPAN: 9 (Port 6 of the ISDN Card)

Setup is : 
SPAN 1 => 16 Port Analogue Card
SPAN 2,3 => E1
SPAN 4-11 => 8 Port S0 Card

> Distorted Audio if you Dial multiple numbers at once (DAHDI)
> ------------------------------------------------------------
>
>                 Key: ASTERISK-20501
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20501
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_dahdi
>    Affects Versions: 1.8.15.0
>         Environment: Debian 6.0, Kernel 26.39.4
>            Reporter: Sven Hirschmueller
>            Assignee: Sven Hirschmueller
>         Attachments: fiallog.log
>
>
> If you dial multiple numbers(channels) at once via the dial app a dahdi call will have distorted audio if it is tunneled in hardware and not in software.
> e.g. You use following ael extension.
> 1000 => {
>   Dial(dahdi/g10/1000&sip/1001);
> };
> If you dial 1000 and you pick up the call via isdn phone 1000 and your caller phone is also a isdn phone on the same BRI-Card the audio will be distorted.
> If you pick up the phone via the sip phone the audio is ok.
> If you don't use multi dial statements the call is also ok and finaly the call is ok if you add parameters to the dial statement that prevent a route of the audio data via the chips as asterisk need to interpret them. (e.g. parameters t or T)
> so following statements are working
> 1000 => {
>   Dial(dahdi/g10/1000);
> };
> also this will work
> 1000 => {
>   Dial(dahdi/g10/1000&sip/1001,,tT);
> };
> i guess the problem is that the bridged call need to be masquaraded and in the end the hardware isn't told correctly witch b-channel to be tunneled to witch b-channel.
> tested with a openvox b800p, sorry i can't test it with our didium te405p e1 cards as i miss the end point for e1.

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