[asterisk-users] Call not going through and failing because "never answered"

[Gareth Blades]

If you add qualify=yes to the setting in sip.conf it will send a sip 
message to the peer every 60 seconds to check if it is alive.
If you try to make a call while the peer is not alive it will fail 
immediatly rather than the caller hearing silence while your box waits 
for a reply timeout.

Andy Beak wrote:
> Hi,
> 
> No that is the correct address.  I know it is an internal IP.
> 
> We have our machine hosted in racks at our SIP providers data center.
> 
> They've patched a new port to our cabinet and linked that to a gateway 
> (172.28.20.105).
> 
> As long as we use that gateway (and the IP address they assigned to us) 
> our traffic will reach their SBC.
> 
> I've confirmed that traceroute follows the path it is supposed to:
> 
> traceroute to 192.168.34.1 (192.168.34.1), 30 hops max, 60 byte packets
>  1  192.168.0.1 (192.168.0.1)  0.656 ms  0.562 ms  0.501 ms
>  2  172.28.20.105 (172.28.20.105)  1.211 ms  1.209 ms  1.196 ms
>  3  192.168.34.5 (192.168.34.5)  23.270 ms  23.269 ms  23.328 ms
>  4  * * *
>  5  * * *
>  6  * * *^C
> 
> Is there a way to test in Asterisk if it is able to reach a particular 
> IP address?  Do you think that there is a routing problem here?
> 
> Thanks,
>  Andy
> 
> 
> 
> 
> On 20/07/2010 04:58 PM, Zeeshan Zakaria wrote:
>>
>> This "host=192.168.34.1" is where you'll put your provider's IP 
>> address. Currently you are using some local address which is not your 
>> provider's IP address. Where did you get it from? Call your providrr 
>> and ask them the IP address of the server where you'll be sending your 
>> calls.
>>
>> Zeeshan A Zakaria
>>
>> -- 
>> www.ilovetovoip.com <http://www.ilovetovoip.com>
>>
>>> On 2010-07-20 10:27 AM, "Andy Beak" <andrewb at cellsmart.co.za 
>>> <mailto:andrewb at cellsmart.co.za>> wrote:
>>>
>>> Hi,
>>>
>>> I set my list to subscribe to digest and I can't see how to reply to 
>>> your reply without starting a new thread.
>>>
>>> There is no need for SIP username and password because the provider 
>>> authenticates me on my IP address.
>>>
>>> I thought that "host=192.168.34.1" would be the sip provider IP address.
>>>
>>> At this point I don't need to accept incoming calls or place 
>>> VOIP-to-VOIP.  All I need to do is connect to their PBX to place a 
>>> call to a cellphone.
>>>
>>> I reread all the documentation I could find and couldn't see where 
>>> else in sip.conf I should set the provider IP.
>>>
>>> Thanks for your reply,
>>>  Andy
>>>
>>>
>>>
>>> > In your sip.conf, there is no mention of your sip provider's IP 
>>> address, username and secret (pa...
>>>
>>> www.ilovetovoip.com <http://www.ilovetovoip.com> 
>>> <http://www.ilovetovoip.com>
>>>
>>>
>>>
>>> > On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at xxxxxxxxxxxxxxx 
>>> <mailto:andrewb at xxxxxxxxxxxxxxx <mailto:andrewb at xxxxxxxxxxxxxxx>>> wr...
>>>
>>> -- 
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>