16 Nov
2005
16 Nov
'05
2:45 p.m.
Does anyone have any advice for me on issue?
Thanks for your help!
Darren Nay
"Darren Nay" wrote:
Hey all,
I have a question regarding usrloc. I have run into a problem..
We have static routes sent to an asterisk server for all of our SIP
usernames. In addition our IAD’s will register with the same
username, so that calls coming into our switch for that username will
be routed to both the asterisk box and the SIP IAD. This way,
whichever endpoint (IAD or asterisk) answers the call first will take
the call.
For example.
root:/ # serctl ul show +18646404810
<sip:+18646404810@192.168.1.60>;q=1;expires=-1012151
<sip:+18646404810@192.168.1.157:5060>;q=;expires=403
192.168.1.60 is the asterisk server. This is a static route added by
serctl.
192.168.1.157 is my IAD which registers with the switch every 10 minutes.
So when calls are made to (864) 640-4810 then SER will send an INVITE
to both location.
I explained all of this just to explain now what my problem is, and
ask if anyone may know a possible solution.
Now, we also use asterisk to perform call fwd’ing functions. Asterisk
will answer the call and then originate another call out back to SER
to a new location. Now the problem! (finally!) This call fwd’ing
method works very well in most cases, except that if the call fwd’ing
is being sent to another location registered with SER then it will be
redirected back to asterisk again, albeit to a different URI, and
asterisk will kill the call because it thinks that it has looped
(which I guess it has… sort of).
So, I’m wondering if there is possibly a way to retrieve only the
usrloc locations that don’t contain the IP address 192.168.1.60 in the
contact URI? This way I can just check if the src_ip is 192.168.1.60
and if so then retrieve all the usrloc locations – without asterisk –
and the call will not be redirected back to asterisk.
Is this possible? Or if anyone has any other ideas that may help then
I am definitely open to suggestions.
Thanks for your help!!
Darren Nay
Ionosphere, Inc
VoIP Network Development
dnay(a)ionosphere.net <mailto:dnay@ionosphere.net>
17 Nov
17 Nov
4:39 p.m.
New subject: [Users] Usrloc question
One last try... :)
Does anyone out there have any advice for me on this issue? I'm stuck
and could really use some help. I've been trying to find a solution
myself for the past week or so and have been less than successful.
Thanks in advance!
Darren Nay
> Hey all,
>
>
>
> I have a question regarding usrloc. I have run into a problem..
>
>
>
> We have static routes sent to an asterisk server for all of our SIP
> usernames. In addition our IAD’s will register with the same
> username, so that calls coming into our switch for that username will
> be routed to both the asterisk box and the SIP IAD. This way,
> whichever endpoint (IAD or asterisk) answers the call first will take
> the call.
>
>
>
> For example.
>
>
>
> root:/ # serctl ul show +18646404810
>
> <sip:+18646404810@192.168.1.60>;q=1;expires=-1012151
>
> <sip:+18646404810@192.168.1.157:5060>;q=;expires=403
>
>
>
> 192.168.1.60 is the asterisk server. This is a static route added by
> serctl.
>
> 192.168.1.157 is my IAD which registers with the switch every 10 minutes.
>
> So when calls are made to (864) 640-4810 then SER will send an INVITE
> to both location.
>
>
>
> I explained all of this just to explain now what my problem is, and
> ask if anyone may know a possible solution.
>
>
>
> Now, we also use asterisk to perform call fwd’ing functions.
> Asterisk will answer the call and then originate another call out
> back to SER to a new location. Now the problem! (finally!) This
> call fwd’ing method works very well in most cases, except that if the
> call fwd’ing is being sent to another location registered with SER
> then it will be redirected back to asterisk again, albeit to a
> different URI, and asterisk will kill the call because it thinks that
> it has looped (which I guess it has… sort of).
>
>
>
> So, I’m wondering if there is possibly a way to retrieve only the
> usrloc locations that don’t contain the IP address 192.168.1.60 in
> the contact URI? This way I can just check if the src_ip is
> 192.168.1.60 and if so then retrieve all the usrloc locations –
> without asterisk – and the call will not be redirected back to asterisk.
>
>
>
> Is this possible? Or if anyone has any other ideas that may help
> then I am definitely open to suggestions.
>
>
>
> Thanks for your help!!
>
>
>
> Darren Nay
>
> Ionosphere, Inc
>
> VoIP Network Development
>
> dnay(a)ionosphere.net <mailto:dnay@ionosphere.net>
>
>
>
4:59 p.m.
I don't know if I understand your problem, but perhaps you can use two
separate location tables, one for asterisk and one for IADs.
Then, when a REGISTER comes, you can compare the source IP and store the
contacts in table location_ast. Other REGISTER requests would go to
location table.
When performing lookup, again, you can compare the source IP and use
either location_ast or location table respectively.
Jan.
On 17-11-2005 10:39, Darren Nay wrote:
One last try... :)
Does anyone out there have any advice for me on this issue? I'm stuck
and could really use some help. I've been trying to find a solution
myself for the past week or so and have been less than successful.
Thanks in advance!
Darren Nay
>>Hey all,
>>
>>
>>
>>I have a question regarding usrloc. I have run into a problem..
>>
>>
>>
>>We have static routes sent to an asterisk server for all of our SIP
>>usernames. In addition our IAD?s will register with the same
>>username, so that calls coming into our switch for that username will
>>be routed to both the asterisk box and the SIP IAD. This way,
>>whichever endpoint (IAD or asterisk) answers the call first will take
>>the call.
>>
>>
>>
>>For example.
>>
>>
>>
>>root:/ # serctl ul show +18646404810
>>
>><sip:+18646404810@192.168.1.60>;q=1;expires=-1012151
>>
>><sip:+18646404810@192.168.1.157:5060>;q=;expires=403
>>
>>
>>
>>192.168.1.60 is the asterisk server. This is a static route added by
>>serctl.
>>
>>192.168.1.157 is my IAD which registers with the switch every 10 minutes.
>>
>>So when calls are made to (864) 640-4810 then SER will send an INVITE
>>to both location.
>>
>>
>>
>>I explained all of this just to explain now what my problem is, and
>>ask if anyone may know a possible solution.
>>
>>
>>
>>Now, we also use asterisk to perform call fwd?ing functions.
>>Asterisk will answer the call and then originate another call out
>>back to SER to a new location. Now the problem! (finally!) This
>>call fwd?ing method works very well in most cases, except that if the
>>call fwd?ing is being sent to another location registered with SER
>>then it will be redirected back to asterisk again, albeit to a
>>different URI, and asterisk will kill the call because it thinks that
>>it has looped (which I guess it has? sort of).
>>
>>
>>
>>So, I?m wondering if there is possibly a way to retrieve only the
>>usrloc locations that don?t contain the IP address 192.168.1.60 in
>>the contact URI? This way I can just check if the src_ip is
>>192.168.1.60 and if so then retrieve all the usrloc locations ?
>>without asterisk ? and the call will not be redirected back to asterisk.
>>
>>
>>
>>Is this possible? Or if anyone has any other ideas that may help
>>then I am definitely open to suggestions.
>>
>>
>>
>>Thanks for your help!!
>>
>>
>>
>>Darren Nay
>>
>>Ionosphere, Inc
>>
>>VoIP Network Development
>>
>>dnay(a)ionosphere.net <mailto:dnay@ionosphere.net>
>>
>>
>>
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18 Nov
18 Nov
7:04 a.m.
I have already answered...
http://lists.iptel.org/pipermail/serusers/2005-November/025670.html
----- Original Message -----
From: Darren Nay
To: serusers(a)lists.iptel.org
Sent: Thursday, November 17, 2005 4:39 PM
Subject: [Serusers] Usrloc question
One last try... :)
Does anyone out there have any advice for me on this issue? I'm stuck and could
really use some help. I've been trying to find a solution myself for the past week or
so and have been less than successful.
Thanks in advance!
Darren Nay
Hey all,
I have a question regarding usrloc. I have run into a problem..
We have static routes sent to an asterisk server for all of our SIP usernames. In
addition our IAD’s will register with the same username, so that calls coming into our
switch for that username will be routed to both the asterisk box and the SIP IAD. This
way, whichever endpoint (IAD or asterisk) answers the call first will take the call.
For example.
root:/ # serctl ul show +18646404810
<sip:+18646404810@192.168.1.60>;q=1;expires=-1012151
<sip:+18646404810@192.168.1.157:5060>;q=;expires=403
192.168.1.60 is the asterisk server. This is a static route added by serctl.
192.168.1.157 is my IAD which registers with the switch every 10 minutes.
So when calls are made to (864) 640-4810 then SER will send an INVITE to both
location.
I explained all of this just to explain now what my problem is, and ask if anyone
may know a possible solution.
Now, we also use asterisk to perform call fwd’ing functions. Asterisk will answer
the call and then originate another call out back to SER to a new location. Now the
problem! (finally!) This call fwd’ing method works very well in most cases, except that
if the call fwd’ing is being sent to another location registered with SER then it will be
redirected back to asterisk again, albeit to a different URI, and asterisk will kill the
call because it thinks that it has looped (which I guess it has… sort of).
So, I’m wondering if there is possibly a way to retrieve only the usrloc locations
that don’t contain the IP address 192.168.1.60 in the contact URI? This way I can just
check if the src_ip is 192.168.1.60 and if so then retrieve all the usrloc locations –
without asterisk – and the call will not be redirected back to asterisk.
Is this possible? Or if anyone has any other ideas that may help then I am
definitely open to suggestions.
Thanks for your help!!
Darren Nay
Ionosphere, Inc
VoIP Network Development
dnay(a)ionosphere.net
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3 comments
3 participants
participants (3)
-
Darren Nay -
Greger V. Teigre -
Jan Janak