[asterisk-users] Calls in ulaw, not gsm as desired

Salvatore Giudice Salvatore.Giudice at VoIPSecurityTraining.com
Tue May 1 14:23:32 MST 2007


How did you set it to 5053?

 

Can you post your sip.conf? You should remove the passwords and ip
addresses, etc.

 

Usually, it's just an allow and a disallow statement inserted into each
inbound and outbound channel definition.

 

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Schall
Sent: Tuesday, May 01, 2007 4:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired

 

That's what I did though. So my sip.conf file no longer has any "allows" in
it. Instead, it should be relying on the realtime settings for that.
However, even though I told it to only use 5053, it still is using ulaw.

Rob

Salvatore Giudice wrote: 

Yeah that is fine. You don't need to do any more than that.

 

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Schall
Sent: Tuesday, May 01, 2007 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired

 

I was in the asterisk console and I typed "reload". Is this not enough to
reload the sip.conf file?

Rob

Andreas Sikkema wrote: 

However, even once I reloaded the extensions, its still only 
using ulaw.
    

 
You didn't reload the sip config? Maybe that's your problem?
 
  

 

 



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