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

[tdensmore]

  For a quick and dirty view, from your asterisk box, do:

tcpdump host 192.168.34.1

and make a test call. For a pcap file you can read with wireshark, 
instead do

tcpdump host 192.168.34.1 -s1500 -w FILENAME.pcap

where FILENAME is whatever you think is meaningful.  This will show you 
what's being sent back and forth.



On 7/20/2010 9:36 AM, 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...
>>>
>>> -- 
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com 
>>> <http://www.api-digital.com>  --
>>>
>>>
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.aste...
>>>
>>>
>>> -- 
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>