Should * be able to diferentiate between two voipuser number

Hi, I have an outgoing number,44844xxxxx29, and an incoming number, 44844xxxxx66. When I get an incoming call on the xxxxx29 number my * box routes this to the correct extention,ext2133 but when I get a call from my xxxxx66 number * sends it to the same extension which it is not configured to do. Looking at the * screen while calls are comming in from these two numbers shows that they are both coming from my xxxxx29 number so * cannot distinguish between them. Is this correct? Below is what happens when I get a call from my xxxxx66 number which should be routed to extention 2000.

-- Executing NoOp("SIP/p1cass0-47fc", "--- "0774xxxxx20" <0774xxxxx20> calling on VoIPUser (44844xxxxx29) ---") in new stack
-- Executing Dial("SIP/p1cass0-47fc", "SIP/2133|15") in new stack
-- Called 2133
-- SIP/2133-0a26 is ringing
== Spawn extension (voipuser-in, 44844xxxxx29, 2) exited non-zero on 'SIP/p1cass0-47fc'

Hi,

p1cass0 :

When I get an incoming call on the xxxxx29 number my * box routes this to the correct extention,ext2133 but when I get a call from my xxxxx66 number * sends it to the same extension which it is not configured to do.

Yeah, i've had that before with another provider - it's the way it matches the provider (I think it uses the host name) from sip.conf.

For the normal inbound numbers, if you are terminating at an Asterisk box, I advise you to use the IAX - this way, you can easily have different numbers going to different contexts and extensions.

See my info here:
http://www.voipuser.org/forum_topic_448.html

Or, Tj's here:
http://www.voipuser.org/forum_topic_95.html

--ian

Thanks for that ichilton. I'll give it a try.

ok, just report any problems here and i'm sure someone will try to help!

--ian

Could you post the appropriate sections of your sip.conf and extensions.conf?

I have multiple numbers with unlimitel.ca via SIP and I can send incoming calls to the right place, all from the extensions.conf.
The incoming call always matches the last entry in my sip.conf, which as far as I can tell only affects the channel name that gets chosen. AFAIK, the extra sections in sip.conf are required only for proper outgoing calls -- as long as you have approprate register lines, any incoming call will go into the default context you've defined in your sip.conf.

In my default incoming context, I use the DID to send the call to a different context:

exten => 514XXXZZZ,1,Goto(contextZZZ,s,1)
exten => 514XXXYYYY,1,Goto(contextYYY,s,1)


(Even if you decided to switch to IAX, I'd like to figure why it didn't work for you...)

bpgtek :

In my default incoming context, I use the DID to send the call to a different context: exten => 514XXXZZZ,1,Goto(contextZZZ,s,1) exten => 514XXXYYYY,1,Goto(contextYYY,s,1)

Yes, this is what I do with one provider. The problem is you can not seem to send numbers into different contexts (because they call come in and use the same sip.conf section even if you create more with the different username & passwords - it ignores them and just uses the 1st one it finds) - I have to have them all in 1 context and use Goto to jump contexts where required.

--ian

Thanks for the last tip.

I had a similar problem with multiple calls from single IAX provider getting confused about contexts. I have used the goto command and it is altogether much tidier - now i can resue the other contexts by having s,1 etc, which i couldnt before.

Not an easy problem to solve before i found this page though!

Thanks

Glad it works for you

--ian

bpgtek :

exten => 514XXXZZZ,1,Goto(contextZZZ,s,1) exten => 514XXXYYYY,1,Goto(contextYYY,s,1)

I've tried this with my setup and I can't get it to work. We have two SIP accounts with two phone numbers and we would like to have different dial plans for each number. We've tried the following with no luck, any help would be much appreciated.

exten => PHONE_NUMBER_ONE,1,Goto(mainmenu-one,s,1)
exten => PHONE_NUMBER_TWO,1,Goto(mainmenu-two,s,1)

Hi
I use Goto all over the place in my dialplans
for example

Quote:

exten => secure,1,Goto(demo,566,1)

What do you get on the CLI when a call is coming in. It is sure to show where its going wrong.

Also what do you get when calling, engaged, NU tone, ring tone no reply ?

Post the cli out put as well as what you have configured in the extensions.conf dont forget the Goto has to be in the context that is specifid in the sip.conf

If I use the following then it works.

exten => _.,1,Playback(vm-goodbye)
exten => _.,2,AbsoluteTimeout(15)
exten => _.,3,Congestion
exten => _.,4,Hangup

But if i use the following then the call just ends:

exten => PHONE_NUMBER_ONE,1,Goto(mainmenu-one,s,1)
exten => PHONE_NUMBER_TWO,1,Goto(mainmenu-two,s,1)

Log output at the time:

Code:

##### Testing 213.228.220.45 with 192.168.0.0 Aug 18 16:10:10 DEBUG[1538]: Target address 213.228.220.45 is not local, substituting externip Aug 18 16:10:10 DEBUG[1538]: Check for res for Aug 18 16:10:10 DEBUG[1538]: is not a local user Aug 18 16:10:10 DEBUG[1538]: is not a local user Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.38 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.38 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: Scheduled a registration timeout # 12277 Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.43 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.43 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: Scheduled a registration timeout # 12279 Aug 18 16:10:16 DEBUG[1538]: Stopping retransmission on '5a695b1e454be22f6329be0d20d86763@127.0.0.1' of Request 1120: Found Aug 18 16:10:16 DEBUG[1538]: Registration successful Aug 18 16:10:16 DEBUG[1538]: Cancelling timeout 12277 Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.38 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.38 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: That's odd... Got a response on a call we dont know about. Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.38 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.38 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: That's odd... Got a response on a call we dont know about. Aug 18 16:10:16 DEBUG[1538]: Stopping retransmission on '572127413b03ed5746f1d3a31c4d8ade@127.0.0.1' of Request 1125: Found Aug 18 16:10:16 DEBUG[1538]: Registration successful Aug 18 16:10:16 DEBUG[1538]: Cancelling timeout 12279 Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.43 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.43 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: That's odd... Got a response on a call we dont know about. Aug 18 16:10:16 DEBUG[1538]: ##### Testing 213.228.220.43 with 192.168.0.0 Aug 18 16:10:16 DEBUG[1538]: Target address 213.228.220.43 is not local, substituting externip Aug 18 16:10:16 DEBUG[1538]: That's odd... Got a response on a call we dont know about. Aug 18 16:10:16 WARNING[1538]: Maximum retries exceeded on call 73314b8958d584724c5ee6af75eadbfc [!at] 82.197.64.20 (replace the [!at] with a @) for seqno 102 (Critical Response) Aug 18 16:10:16 WARNING[1538]: Maximum retries exceeded on call 73314b8958d584724c5ee6af75eadbfc [!at] 82.197.64.20 (replace the [!at] with a @) for seqno 102 (Critical Response)

Ok.

First a few notes on contexts.

Your lines and phones must all be accessable from the contexts its passing through.

So for example you have a phone in the default context lines in incoming-sip and menus in a context called menu-1 menu-2
then default must have an include for menu-1 menu-2 and incoming-sip and vice vercer.

so the contexts should look something like this

[menu-1]
include => incoming-sip
include => default

[menu-2]
include => incoming-sip
include => default

[incoming-sip]
include => menu-1
include => menu-2
include => default

or somrthing like that anyway

What does the relevent bits of yours look like?
without seeing the relevent bits in full its V hard to say whats wrong.


As to the debug its doesnt help to much with call flow,

the best for faulting callflow is the normal console with 4 or 5 levels of verbouse. ie use "asterisk -vvvvvr"
and when you fire a call in you will see the call flow.



Ian

I have been unable to get IAX to work other than locally. Even though I've made the appropriate NAT router change to forward UDP 4569 to the Asterisk machine, I just don't see any incoming calls. I thought IAX was meant to be easier than SIP for NAT, but not turned out that way for me.

~~~~~~~~~~~~~~~~~~~~~~
The Asterisk SIP channel code seems to go out of its way to prevent you discovering the incoming call number, even though it is clearly known and displayed in the debug. The only hope seems to be the broken ${DIALLEDPEERNAME} and ${DIALEDPEERNUMBER}.

~~~~~~~~~~~~~~~~~~~~~~
However, I have managed to distinguish calls on different incoming numbers on SIP [without any need for change at the router]. Alas, this required not just two numbers but two Voipuser ids.

then in sip.conf
[general]
register => user1:xxxx@sip.voipuser.org
register => user2:xxxx@sip.voipuser.org

[VOIPUSER1]
type=friend
username=user1
realm=voipuser.org
secret=xxxx
host=sip.voipuser.org
context=VOIPUSER-IN1

[VOIPUSER2]
type=friend
username=user2
realm=voipuser.org
secret=xxxx
host=sip.voipuser.org
context=VOIPUSER-IN2

~~~
and in extensions.conf

[VOIPUSER-IN1]
exten => s,1,NoOp(incoming call on 1)
exten => s,2,etc etc appropriate for first callin number


[VOIPUSER-IN2]
exten => s,1,NoOp(incoming call on 2)
exten => s,2,etc etc appropriate for second callin number

~~~~~~~~~~~~~~~
Or you could route the two numbers into the same context (say VOIPUSER-IN) with
[general]
register => user1:xxxx@sip.voipuser.org/num1
register => user2:xxxx@sip.voipuser.org/num2

with appropriate change to the context= statements

and then extensions.conf
[VOIPUSER-IN]
exten => num1,1,NoOp(incoming call on 1)
exten => num1,2,etc etc appropriate for first callin number

exten => num2,1,NoOp(incoming call on 2)
exten => num2,2,etc etc appropriate for second callin number

~~~~~~~~~~~~~~~~~~~~~~~~
Hope that helps
Stephen

richardys :

bpgtek : exten => 514XXXZZZ,1,Goto(contextZZZ,s,1) exten => 514XXXYYYY,1,Goto(contextYYY,s,1) I've tried this with my setup and I can't get it to work. We have two SIP accounts with two phone numbers and we would like to have different dial plans for each number. We've tried the following with no luck, any help would be much appreciated. exten => PHONE_NUMBER_ONE,1,Goto(mainmenu-one,s,1) exten => PHONE_NUMBER_TWO,1,Goto(mainmenu-two,s,1)

I've managed to sort it out in the end

Rather then:

exten => PHONE_NUMBER_ONE,1,Goto(mainmenu-one,s,1)
exten => PHONE_NUMBER_TWO,1,Goto(mainmenu-two,s,1)

It needed to be:

exten => SIP_USERNAME_ONE,1,Goto(mainmenu-one,s,1)
exten => SIP_USERNAME_TWO,1,Goto(mainmenu-two,s,1)

I also added the sip username to the end of the register string:

SIP_USERNAME:SIP_PASSWORD@SIP_HOST/SIP_USERNAME

Hi Richard

You dont mention who your supplier is .

For voipuser and most suppliers the correct method in extensions .conf is phonenumber and the register line is account name.

Now if you are using sipgate then the internal phone number is your account name. This is a little confusing.
But here is an example

your phone number is 01234123456 you sipgate account name would be 5123456 for example, And 5123456 is a valid internal number. but 01234123456 is mapped to it.

Its a shame that there is no standard of how calls are presented , It all down to knowing the querks of each supplier.

Ian