Sorry, I just fixed it by myslef. It is an issue of incompatible
codec. I am wondering why option "t" in dial() is not able to make it
work.
Any advice??? Many Thanks.
On 5/6/05, VoIP Newbie <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I could register * to a provider. However, I failed to make outgoing
> calls through the provider. Please help and advise how to get it work.
>
> m2*CLI> sip show registry
> Host Username Refresh State
> sip_proxy:5060 abc 105 Registered
>
> m2*CLI> sip show peers
> Name/username Host Dyn Nat ACL Mask
> Port Status
> sip_proxy/abcxxxx 107.211.128.16 255.255.255.255 5060
> Unmonitored
> 2 sip peers [2 online , 0 offline]
>
> -- Executing Dial("SIP/12345678-00d4", "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> May 6 19:24:49 WARNING[4173]: channel.c:2173
> ast_channel_make_compatible: No path to translate from
> SIP/sip_proxy-1713(4) to SIP/12345678-00d4(256)
> -- SIP/sip_proxy-1713 is ringing
> -- SIP/sip_proxy-1713 answered SIP/12345678-00d4
> May 6 19:24:50 WARNING[4173]: channel.c:2173
> ast_channel_make_compatible: No path to translate from
> SIP/12345678-00d4(256) to SIP/sip_proxy-1713(4)
> May 6 19:24:50 WARNING[4173]: app_dial.c:1251 dial_exec_full: Had to
> drop call because I couldn't make SIP/12345678-00d4 compatible with
> SIP/sip_proxy-1713
> == Spawn extension (sip, 99912345678, 1) exited non-zero on
> 'SIP/12345678-00d4'
>
> [sip_proxy]
> type=peer
> secret=sfdsf
> username=abc
> fromuser=abc
> fromdomain=proxy.provider.net
> host=proxy.provider.net
> usereqphone=yes
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
