[asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP

[Rizwan Hisham]

1.8 supports static peers along with realtime peers. I have tested.

On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho <
rjcarvalho.lists at gmail.com> wrote:

> Thanks Faisal, in fact I made a test that confirmed that in realtime
> asterisk doesn’t supported static peers, like you told me.
> Do you know if newer versions of asterisk, like 1.8, have this issue
> already solved?
>
> Regards,
> Ricardo.
>
>
>
>
> On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif <faisal at vopium.com> wrote:
>
>> I have played a lot on this issue with asterisk config but in realtime it
>> doesn’t supported static peers with version 1.6.2.14.
>>
>>
>>
>> *From:* Ricardo Carvalho [mailto:rjcarvalho.lists at gmail.com]
>> *Sent:* Wednesday, February 16, 2011 10:21 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Cc:* Faisal Hanif
>> *Subject:* Re: [asterisk-users] trunk not working if I register a phone
>> at the same IP as the trunk peer's IP
>>
>>
>>
>> Isn't this a limitation that can be surpassed with some configuration that
>> I'm lacking in my sip.conf or extensions.conf of my asterisk?
>>
>>
>>
>> Ricardo.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif <faisal at vopium.com> wrote:
>>
>> Well a quick n easy fix for you is you can configure you call sending
>> peers to use username & secret in INVITE. As far as I know it possible in
>> almost all CISCO, Avaya and all other standard Gateway and SBCs which
>> follows full SIP RFCs.
>>
>>
>>
>> If you can’t do it then you need to use curl as realtime engine instead of
>> MySQL. It will call a URL for each SIP request which you can handle with
>> flexibility in your CGI scripts with apache. But be careful as per my
>> experience asterisk 1.6 with curl as realtime engine can handle a max of 120
>> registration in parallel if registration refresh time is 120 seconds.
>>
>>
>>
>> Faisal Hanif
>>
>>
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Ricardo Carvalho
>> *Sent:* Wednesday, February 16, 2011 9:41 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] trunk not working if I register a phone at
>> the same IP as the trunk peer's IP
>>
>>
>>
>> How should I configure my asterisk server so that I can receive calls from
>> an unregistered peer from whom I also receive registrations of sip phones?
>>
>>
>>
>> I'm asking you this, because with my actual configuration, when I register
>> a contact from that peer's IP, no more inbound calls are accepted from that
>> peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication
>> Required", I assume because they don't carry the registered contact
>> registration!!!
>>
>> My SIP contacts have type=friend and all inbound calls not coming from my
>> registered phones fall in the default context without authentication, so
>> that someone in the Internet be able to call freely through the Internet
>> anyone in my server's dial plan.
>>
>>
>>
>> Some ideas?
>>
>>
>>
>> Regards,
>>
>> Ricardo Carvalho.
>>
>>
>> --
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>>
>
>
> --
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-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 3333 6767 26
E: rizwanhasham at gmail.com
W: www.axvoice.com
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