[asterisk-users] Desperately need help with Asterisk setup

[Mojo with Horan & Company, LLC]

I agree, seems odd you didn't have a [peername] section for your 
softphone in your sip.conf.

aren't 404 errors a likely symptom of this? :)

Mojo
Steve Totaro wrote:
> Pete,
>
> You are connecting via a SIP softphone correct?  Where is that in your sip.conf?
>
> On Mon, Mar 17, 2008 at 11:42 AM, Pete Kay <petedao at gmail.com> wrote:
>   
>> Hi,
>>
>> My sip.conf has the allow=gsm as shown in the following:
>>
>>
>> [general]
>> port = 5060
>> bindaddr = 0.0.0.0
>> context = others
>>
>> register =>outraspace:password at voipuser.org/outraspace
>>  nat=yes
>> externip=58.251.75.251
>>
>> localnet=192.168.1.0/255.255.255.0
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>>  qualify=yes
>>
>> All the sound files are in /var/lib/asterisk/sounds instead.  Is it correct?
>>
>> I have tried both Wengo and xlite, but same result.
>>
>> I can't figure out what caused the 404 error.  Any idea?
>>
>>
>> Thank you so much for your help.
>>
>> Pete
>>
>>
>>
>> On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin Hoffmeister
>> <anselm at hoffmeister-online.de> wrote:
>>
>>     
>>> Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay:
>>>       
>>>> Hi,
>>>>
>>>>         
>>>> Here is the SIP debug output for the playback test.  Thank you so much
>>>> for your help.
>>>>         
>>> Hi Pete,
>>>
>>>
>>>       
>>>> <------------>
>>>> [Mar 18 05:33:08]     -- Executing [333 at my-phones:1]
>>>> Answer("SIP/2000-081e0738", "") in new stack
>>>> [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028
>>>> [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP
>>>> [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP
>>>> [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP
>>>>         
>>> I do not see "gsm" here. Any reason not to allow that codec? Or did I
>>> miss something? You wrote you enabled it, so it should be here IMO.
>>>
>>>
>>>       
>>>> <--- Transmitting (NAT) to 192.168.1.102:5060 --->
>>>> SIP/2.0 404 Not Found
>>>> Via: SIP/2.0/UDP
>>>>
>>>>         
>> 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102;rport=5060
>>     
>>>> From: 2001 <sip:2001 at 192.168.1.101>;tag=2612560371
>>>> To: <sip:ping at 192.168.1.101>;tag=as0ca1ddb0
>>>> Call-ID: 2808830214 at 192.168.1.102
>>>> CSeq: 20 OPTIONS
>>>> User-Agent: Asterisk PBX
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>> Supported: replaces
>>>> Accept: application/sdp
>>>> Content-Length: 0
>>>>         
>>> "404" does not sound good. Please, look which sound files exist on your
>>> system (e.g. what does
>>>        find /usr/share/asterisk -file "vm-goodbye*"
>>> say?)
>>>
>>> Another point: Which client do you use, is it Wengo or is it Xlite? Or
>>> both? In that case: Any differences?
>>>
>>>
>>>
>>>
>>> BR
>>> Anselm
>>>
>>>
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>       
>> _______________________________________________
>>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>>  asterisk-users mailing list
>>  To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>     
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>