[asterisk-ss7] chan_ss7 one way adio

Antoine Megalla aatef at rocketmail.com
Tue Feb 24 07:49:05 CST 2009


This one way audio problem is caused by a little bug in the free_cic routine in l4isup.c
The free_cic function neglects to reset the DTFM sending flag , and this in turn causes any new call on the CIC that used DTMFs to have one audio.
This is due to the fact the chan_ss7 stops audio transmission when there are DTMFs to be handled, so not resetting the DTMF flag causes one way audio on subsequent calls on the same CICs.

To correct the error just edit l4isup.c and add the following line to the free_cic function
pvt->sending_dtmf = 0;

So your function should look like this:

static void free_cic(struct ss7_chan* pvt)
  pvt->state = ST_IDLE;
  pvt->hangupcause = 0;
  pvt->dohangup = 0;
  pvt->has_inband_ind = 0;
+  pvt->sending_dtmf = 0;
  pvt->owner = NULL;


Antoine Megalla.

----- Original Message ----
From: marek cervenka <cervajs at fpf.slu.cz>
To: asterisk-ss7 at lists.digium.com
Sent: Tuesday, February 24, 2009 2:46:21 PM
Subject: Re: [asterisk-ss7] chan_ss7 one way adio

> Hello,
> I`m using chan_ss7-1.0.10  with Digium TE410 card . Very often is one way
> audio where the called party can`t hear the callee, I `ve seen this problem
> was also reported by others, is there any fix available ?  

use chan_ss7 1.1 http://www.dicea.dk/download/chan_ss7-1.1.tar.gz

Marek Cervenka


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