[Asterisk-Users] Help Me - SIP Phones ( No Voice) !!!!

[Steve Maroney]

Also make sure there isn't any packet filtering enabled on the BSD box as
well.

Thank you,
Steve Maroney

On Wed, 1 Sep 2004, Jefferson Carvalho wrote:

> Hello list,
>
> I've posted my problem on BSD list and i still have the
> problem.
> The remote side receives the call , but there's no voice
> on the call.
> I tried everything about possible NAT problems ..
> but ther're on same net.
>
> My platform:
>
> FreeBSD 5.2.1-Release
> Asterisk 1.0-RC2
> soft phones : X-Lite
>
>  >>>>
> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20") in new stack
> -- Called 1262
> -- SIP/1262-c597 is ringing
> -- SIP/1262-c597 answered SIP/1260-a7ae
> -- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> - Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> Sep  1 14:53:17 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
> retries exceeded on call
> DB93109A-FC24-11D8-B3A5-005004803F0B at 172.20.1.133 for seqno 11288
> (Non-critical Response)
> *
>  >>>>> My sip.conf
>
> *[1260]
> type=friend
> username=1260
> secret=jeff
> context=sip
> qualify=300
> mailbox=1260
> callerid="Jefferson Carvalho" <1260>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
> [1262]
> type=friend
> context=sip
> username=1262
> secret=1262
> qualify=300
> callerid="Ialle" <1262>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
>
> *>> My extensions.conf
> *
> [general]
>
> static=yes
> writeprotect=no
>
> [globals]
>
> CONSOLE => Console/dsp
> IAXINFO => guest
> TRUNK => Zap/g2
> TRUNKMSD => 1
>
> [sip]
>
> exten => 1260,1,Dial(SIP/1260,20)
> exten => 1261,1,Dial(SIP/1261,20)
> exten => 1262,1,Dial(SIP/1262,20)
>
> Best Regards,
>
> -Jefferson Carvalho
>  IT Analist
>  Credishop S/A
>  Teresina-PI-Brasil
>  5586-94321901
>
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