hi all,
   my apologies for bothering you all with my previous post...already found
the problem....forgot to add my gateway to the domain and specify its name
server.


these commands were the ones i neglected to include in the router

*ip domain name ourcompany.com*

*ip name-server 172.16.10.254*

seems making my brain work after long periods of inactivity is not as easy
as i thought -_-




On Thu, Apr 15, 2010 at 3:41 PM, ronald teng <ronaldt...@gmail.com> wrote:

> Reconfigured my router/gateway w/ different ip(s) and setup gre vpn config
> and now im back to having problems with receiving incoming calls. Same
> symptoms, i can call the AA from an outside pstn line but when i ask it to
> transfer me to a user extension, it just stays silent for a minute then goes
> busy. This time around though, im getting a 503 service unavailable error
> instead of the 401 in the debug. Outgoing calls still working. I've attached
> my cisco gateway config and results of the debug. I hope someone can shed
> some light on this. thnx.
>
>
>
>
> On Mon, Apr 5, 2010 at 7:51 PM, Jean-Hugues Royer <jhro...@joher.com>wrote:
>
>>  Hi,
>>
>> Glad I could help.
>>
>> Now you got it is working I advise you to take some time when you can to
>> investigate why the Cisco was unable by itself (without a configured route)
>> to decide where to send the INVITE, you probably have an underlying problem
>> that you masked by this quick fix that might reappear in other conditions.
>>
>> Regards.
>>
>>
>> ronald teng wrote:
>>
>> Hi Jean,
>>     Thnx for highlighting to me why it's not working...i got it working
>> now....im soooo happy i feel like crying.
>> It was an issue w/ one of my dialpeers on cisco.
>>
>> my previous config was:
>>
>> dial-peer voice 10 voip
>>  description ***  SIP trunk from router to SipX ***
>>  *destination-pattern 100*
>>  voice-class codec 1
>>  session protocol sipv2
>>  session target ipv4:10.9.20.254
>>  dtmf-relay rtp-nte
>>  no vad
>>
>> it seems this is the only dialpeer i have pointing from the cisco gateway
>> to the sipX machine....so when we ask the gateway to take us to other
>> extensions (e.g. 210)  it doesn't know where to send it.
>>  I corrected it to "destination-pattern ..." and everything's now up and
>> running. Will be trying to feature sets from here on out. Thnx so much again
>> for your and everyone elses input. Couldn't have fixed this w/o ya'll.
>>
>> Regards,
>> Ron
>>
>>
>>
>>
>>
>>
>>
>> On Sat, Apr 3, 2010 at 8:02 PM, Jean-Hugues Royer <jhro...@joher.com>wrote:
>>
>>> Hi,
>>>
>>> I will assume that you meant still not working.
>>>
>>> From the cisco logs the only thing in which was weird at first sight was
>>> that you had IPs in every SIP URI (request/from/to/contact) _but_ in the
>>> REFER Refer-To field where you had "2...@alsterph.lan" <2...@alsterph.lan>
>>> .
>>>
>>> Since the cisco accepts the REFER (202 Accepted) but he doesn't generate
>>> an INVITE to "2...@alsterph.lan" <2...@alsterph.lan> in response and issue
>>> a "503 Service Unavailable", the most obvious answer is that he does support
>>> REFER but he is unable to decide where to send the INVITE to
>>> "2...@alsterph.lan" <2...@alsterph.lan>.
>>>
>>> I have very basic cisco knowledge, but the answer probably lies in how
>>> the cisco internally decides where to send an INVITE to
>>> "2...@alsterph.lan" <2...@alsterph.lan>.
>>>
>>> Normally if he has no pre-configured way to know it, he should issue a
>>> DNS SRV request for "alsterph.lan", from this DNS SRV request he will get
>>> the transport (tcp/udp), the host and port where he has to send his INVITE
>>> packet.
>>>
>>> So what I offered you was to first check/create a DNS SRV for
>>> "aslterph.lan",  or in last resort to use an IP instead of a host/domain in
>>> the "2...@alsterph.lan" <2...@alsterph.lan> transfer target which is
>>> probably the easiest (but not nicest) way to solve your problem.
>>>
>>> Regards.
>>>
>>>
>>> ronald teng wrote:
>>>
>>>> UPDATE: DNS has been fixed so i can now ping just
>>>> 'alsterph.lan'...incoming still now working (T_T)
>>>>
>>>> would i need to use the sip call transfer commands of cisco for this?
>>>> According to this link i posted before, i'll need to have this script named
>>>> " app_h450_transfer.tcl " in my flash. Any ideas as to where i can find a
>>>> copy of this?
>>>>
>>>>
>>>>
>>
>
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