[Asterisk-Users] Re: Connecting to provider

[VoIP Newbie]

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 <voip.newbie at gmail.com> 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/9991234567 at sip_proxy") in new stack
>    -- Called 9991234567 at sip_proxy
> 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
>