[asterisk-users] 428 Loop Detected

[Marco Mouta]

It's so uncommon for me fxs and fxo cards and based on the reference
of sip.conf files and accounts i totally missed last paragraph where
it was mentioned only analogue lines and fxs (phone).

my appologies.

E1 and BRIs and sip trunks have been overloading my last month of work.

cheers,
--
Marco Mouta



2009/3/16 Steve Totaro <stotaro at totarotechnologies.com>:
> Again, if I am interpreting this correctly, he is not using SIP.  A
> four port card 2fxo/2fxs means to me that he is not using SIP at all.
>
> If by card, you mean some kind of SIP gateway, then I misunderstood
> and the problem, but seeing DAHDI channels leads me to believe that
> SIP is not required and actually causing your problems.
>
> SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this
> case)...  If you had a SIP device, it would be connected to the data
> network, not a phone line.  Can you just plug your phone into a
> regular landline jack and get dialtone?  If so, forget SIP for now.
>
> Comment out or delete all your sip.conf peers since you are not using SIP.
>
> Change your dialplan to not (Dial/SIP" but (Dial/DAHDI/1,10) and the
> correct channel to your FXS port that the phone is connected to.
>
> Thanks,
> Steve Totaro
>
> On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta <marco.mouta at gmail.com> wrote:
>> Hi,
>>
>> problem is that you are saying that phone in sip.conf is at the same
>> ip address of your asterisk box so you are dialing into a loop to your
>> self asterisk box
>>
>> [phone]
>> type=friend
>> context=phone1
>> secret=g00dpazzwerd
>> bindport=5060
>> host=192.168.1.106
>> dtmfmode=rfc2833
>>
>> what you need is:
>>
>> [phone]
>> type=friend
>> context=phone1
>> secret=g00dpazzwerd
>> dtmfmode=rfc2833
>> host=dynamic
>> ;configuring your codecs (i don't know what else you have configured,
>> just preventing audio for you)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>>
>>
>> Dial sip/phone is enough too..
>>
>> [from-pstn]
>> ;include => default
>> exten => s,1,Dial(SIP/phone,10)
>> exten => s,2,Voicemail(line)
>> exten => s,3,Hangup
>>
>>
>> hope it helps.
>>
>> don't forget to asterisk reload on cli.
>>
>> Looking forward to hearing from you.
>>
>> cheers
>>
>> --
>> Marco Mouta
>>
>>
>>
>> On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal <vadud3 at gmail.com> wrote:
>>> Hi I looked at few emails related to this subject. And still not sure
>>> how to solve the loop detect problem for my case
>>>
>>> iqbala at improvise:/etc/asterisk$ cat sip.conf
>>>
>>> [general]
>>> context=line1
>>>
>>> [phone]
>>> type=friend
>>> context=phone1
>>> secret=g00dpazzwerd
>>> bindport=5060
>>> host=192.168.1.106
>>> dtmfmode=rfc2833
>>>
>>> [line]
>>> type=friend
>>> context=line1
>>> secret=anothers33cret
>>> bindport=5061
>>> host=192.168.1.106
>>> dtmfmode=rfc2833
>>>
>>> iqbala at improvise:/etc/asterisk$ cat extensions.conf
>>> [default]
>>> exten => s,1,Answer
>>> exten => s,2,Wait(2)
>>> exten => s,3,Playback(tt-monkeys)
>>> exten => s,4,Hangup
>>>
>>> [from-internal]
>>> include => default
>>>
>>> [phone1]
>>>
>>> [from-pstn]
>>> ;include => default
>>> exten => s,1,Dial(SIP/phone at phone,10)
>>> exten => s,2,Voicemail(line)
>>> exten => s,3,Hangup
>>>
>>> [line1]
>>>
>>>
>>> So my home land line is going to the FXO port and my home phone is
>>> hanging off of FXS port.
>>>
>>> Here are the contexts for my fxo/fxs card
>>>
>>>
>>> improvise*CLI> dahdi show channels
>>>   Chan Extension  Context         Language   MOH Interpret
>>>  pseudo            default                    default
>>>      1            from-internal              default
>>>      2            from-internal              default
>>>      3            from-pstn                  default
>>>      4            from-pstn                  default
>>>
>>>
>>> I want to call from my cell and make my home phone ring and if I dont
>>> pickup in 10 secs I want the call
>>> go to my voicemail. But I am getting a loop detect. The debug output
>>> is attached.
>>>
>>> What am I doing wrong?
>>>
>>> --
>>> Asif Iqbal
>>> PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu
>>> A: Because it messes up the order in which people normally read text.
>>> Q: Why is top-posting such a bad thing?
>>>
>>>
>>
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>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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