asterisk users - Aug 2003 - X-Lite - No sound + chan_sip issue

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to
get iLBC to work and someone the other days couuld not get SPX working..

You will need to enable/disable the codecs in X-Lite..

If you also want to control the codecs that * uses then put the following in the
general section of your sip.conf

disallow=all
allow=alaw
allow=ulaw
allow=gsm

Hope that helps..
> Hi,
> 
> X-Lite logs into * with no problems. I dial "1000" and * plays
greeting, but
> i can not hear it.
> Tried many times with the same result.
> 
> After quite few tries * complains about:
> -------------------------------------------------
> WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries
> exceeded on call FBD1EC34-9FD3-4855-9DC1-97933EE0F5F3@192.168.1.104 for
> seqno 43 (Response)
> WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to
> determine sequence number from ''
> -------------------------------------------------
> 
> 
> 
> Has anyone had the same problem?
> 
> Senad
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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If I am understanding correctly your setup looks like this..

{Asterisk}--[NAT]--Internet--[NAT]--{X-Lite}

If this is correct then you are going to have major problems getting it to
work.. Your RTP traffic is going to get very confused..

You need to get Asterisk onto a Public IP address..

I have seen many try and get the double NAT setup to work but I haven't yet
heard of anyone getting it right..


> Hi,
> thanks for that.
> 
> after implementing yours and "wipeout's" suggestions (thank
you both),
> x-lite changed its default codecs to G711a. which is great... and a way
> forward.
> 
> but it still does not play sound when the "1000" is dialed.
> 
> my * is behind nat. and my test pc is as well.
> Here are my settings:
> 
> sip.conf
> [senad]
> type=friend
> secret=blah
> host=dynamic
> dtmfmode=inband
> 
> thanks
> 
> senad
> 
> 
> 
> 
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Nathan
> Littlepage
> Sent: 08 August 2003 19:50
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
> 
> 
> Change the allow=all in sip.conf to allow=alaw and see if that works.
> 
> > -----Original Message-----
> > From: asterisk-users-admin@lists.digium.com
> > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
> > Senad Jordanovic
> > Sent: Friday, August 08, 2003 1:14 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue
> >
> >
> > Hi,
> >
> > X-Lite logs into * with no problems. I dial "1000" and *
> > plays greeting, but
> > i can not hear it.
> > Tried many times with the same result.
> >
> > After quite few tries * complains about:
> > -------------------------------------------------
> > WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt):
> > Maximum retries
> > exceeded on call
> > FBD1EC34-9FD3-4855-9DC1-97933EE0F5F3@192.168.1.104 for
> > seqno 43 (Response)
> > WARNING[81926]: File chan_sip.c, Line 2002
> > (__transmit_response): Unable to
> > determine sequence number from ''
> > -------------------------------------------------
> >
> >
> >
> > Has anyone had the same problem?
> >
> > Senad
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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