I'm trying to get a working dialplan for my spa 112.
- I live in holland, land code 0031
- The region code is 0165 (the city of Roosendaal)
- I have line 1 configured als 'regular' voipprovider (with a phone number so people can call me) and line 2 is betamax (freevoipdeal).
- I want call all numbers except numbers that start with 0900, 0800, 14 and 112 (so 4 exceptions) via line 2 (via betamax). The 4 numbers i wrote i want to call via line 1, because that are numbers it's not possible or reliable to call via betamax.
I tried a lot dialplans but i'm not going to get it perfect.
Please help me,
Well, as phone permanently stays connected in line 2 connector, we need to speak about dial plan of second line only.
I know the only way to direct particular call to particular provider. Read the Administrator guide, see example on bottom of page 95. Yes, it's mad to search Dial Plan configuration example here, not in paragraph related to Dial Plan Configuration, and yes, again, it's wrong that something important like description of substitution syntax allowing redirect of call to another operator is mentioned just in such semi-hidden example only, but that's life.
As the NL numbers have no fixed length, it seems you can't do something like 000XXXXXXX<:@other_operator_settings> and I'm not sure if something like 000!<:@other_operator_settings> will work. The Dial Plan matching alghoritm is not described in the documentation and syntax description is very far from complete and perfect. So many Dial Plan I wrote in the past didn't worked as expected. So we need to guess and you need to try it.
The SPA112 has two independent lines that cannot communicate with each other.
Just to be clear, PHONE 1 makes calls based on dial plan for Line 1 and PHONE 2 makes calls based on dial plan for Line 2.
PHONE 1 cannot make a call using Proxy of Line 2 and PHONE 2 cannot make a call using Proxy of Line 1.
Stated differently, the SPA112 does not have any gateway functionality. If you need this functionality, go for the SPA122 or SPA232D.
I wrote some advanced dial plan usage material for the SPA232D which is similar but slightly more advanced than the SPA112 in that it has a PSTN gateway and DECT. Aside from that, the dial plan section in the
https://supportforums.cisco.com/docs/DOC-30269 document may be of some use to you. Take a look at the "Outbound VoIP Calls" section starting page 20/30.
I also wrote a document for the SPA3102, also more advanced than the SPA112 but shares similar dial plan syntax so perhaps you'll find it useful: https://supportforums.cisco.com/docs/DOC-9902
Now that you're aware of what your SPA112 can and cannot do, just create a dial plan first for Line 1 and then when that's working, create a different dial plan for Line 2.
Specifics about your post:
You need to configure as follows:
Quick Setup tab > Line 1:"regular voipprovider" information
Quick Setup tab > Line 2: "betamax" information
Attach analog phone to PHONE 1 and attach another analog phone to PHONE 2. Either phone should play dial tone when you go off hook.
If you want to make a call using your "regular voipprovider" dial out using phone attached to PHONE 1.
If you want to make a call using "betamax", dial out using phone attached to PHONE 2.
Based on what you say, this may help get you started:
Dial plan for Line 1 will allow only numbers starting with 0900, 0800, 14, and 112
Dial plan for Line 2 will deny numbers starting with 0900, 0800, 14, and 112 while allowing everything else
Note: dial plans are read left to right, as soon as a pattern match is found, the rest of the dial plan is ignored.
Note: if a sequence is not allowed you will immediately hear fast-busy play from the ATA, example with dial plan 1, if you dial 3, you will immediately hear fast busy because nothing matches a 3. If you dial 0, you will hear nothing, dial 7, you will hear fast busy because 07 is not part of dial plan 1.
I suspect you missed the question or I missed your answer. It seems you are trying to say it's not possible to use one phone connected to one line to call two independent SIP servers. I can't confirm it.
As long as <:@provider;;> substitution work in Dial Plan, the administrator can route different numbers to different exchanges. And it's what Ronald seems to be asked for.
Well, I tried it. I configuration as follows (server names are obfuscated a little):
SPA112 firmware 1.3.2, Line 1 configured to account DEV-6018 on server test.sip._____.cz
Dial Plan: (2xxx|6xxx<:@kgw.___.cz:5060;uid=6xxx;pwd=xyz)
I tried to dial 2002 - sip INVITE message has been sent to server test.sip._____.cz as usual.
Then I tried to call 6002 and catched following INVITE packed transsmited to kgw.____.cz:
INVITE sip:6002@kgw._____.cz:5060 SIP/2.0
Via: SIP/2.0/UDP 10.____.11:5060;branch=z9hG4bK-e931085c;rport
Based on the test described above I wish it's possible to connect one to one line of SPA112 and use "Dial Plan" to route calls from such one line to two independent servers.
hi, I tried on spa112
|Firmware Version:||1.4.1 (002) Oct 26 2015|
But did not work for me I tried as @localphone.com;5060;uid=abc;pwd=abc
can u please explain more
eg; do i have to replace pwd with password
You didn't disclosed your Dial Plan (which may be wrong) you didn't catched SIP session packets, so we can't guess what vague description "it not work for me" mean.
It's so hard to advice something to you ...
Hi Dan Lukes, thanks for reply.
Default dial plan on my spa112 is ( *xx | 11 | 0 | [2-9]xxxxxx | 1xxx[2-9]xxxxxxS0 | xxxxxxxxxxxx. )
works fine with current voip provider. I am in canada. I want long distance calls route through other voip provider. for that I made changes in my dial plan as follow.
If I dial starting with one (1xxxxxxxxxx) it should route through other voip (eg; localphone.com)
and calls does not route through lcalphone.com
and I also want calls starting with 00 also should route through other voip
this thing work fine with spa3102 with dial plan
by entering gw1 info in gateways account section
I still can't decide the rule 1[2-8]xxx.<:@localphone.com:5060;uid=abc;pwd=xyz> apply to your particular call at all. Most numbers that fit this rule comply with 1xxx[2-9]xxxxxxS0 as well. You should stay away from overlapping plans (e.g. particular number fit more than one rule).
At least for the purpose of debugging start with simplest dial plan possible - just two non overlapping rule, one without routing, one with routing. Such test allow us narrow the issue cause.
It just should be noted the ATA has not been designed for such kind of tasks (e.g. a smart routing of calls). It's just POTS<->SIP converter. All complex routing decisions should be dedicated to PBX. So you are pushing the device to the edges.