PRI issues connecting asterisk as Gateway to Swyx

Asterisk Asterisk Support
1

Hi all,

I’m stumped with a PRI issue here…

We are trying to use an Asterisk box as a pri gateway between our public lines and some systems behind it.
Our setup:
Asterisk 1.6 server with 4 port PRI card

  • one port connected to the public network
  • one port to our Swyx PBX system.
    Calls coming in on the span connected to the swyx are connected to the public line span using a simple Dial(${span_1}/${EXTEN},r) command.

Swyx Server with multiple IP phones.

  • Some Swyx/Siemens phones (apparently these don’t use SIP but some other protocol to connect to the Swyx PBX)
  • Some SNOM SIP phones

So a call would look like this
[SNOM phone]-[color=#FF0000]----sip-----[/color][PBX][color=#FF0000]-----PRI-----[/color][ASTERISK][color=#FF0000]-----PRI-----[/color][PUBLIC NETWORK]
[Siemens phone]-[color=#FF0000]----???-----[/color][PBX][color=#FF0000]-----PRI-----[/color][ASTERISK][color=#FF0000]-----PRI-----[/color][PUBLIC NETWORK]

Now, when I call out using a SNOM phone, all is well! I can dial out and call etc. When using a ‘fancy’ Siemens phone though… It seems the calling number is never received by asterisk, so it does not know where to connect me to.

When testing by directly connecting my Swyx PBX to the public network, everything works fine!

I spoke to Digium support, and we found some interesting information… I’m not sure what to do with it though…

When viewing the d-channel logs on the PBX, I find that when a ‘normal’ SIP phone calls through my pbx to the asterisk:

The everything works fine and the logs shows this:

[quote]16:42:55.361 L3 1 01110000 INFORMATION ELEMENT : Called party number
16:42:55.361 L3 2 00001011 IE Length : 11 octet(s)
16:42:55.361 L3 3 1------- Extension bit : Not continued
16:42:55.361 L3 -000---- Type of number : Unknown
16:42:55.361 L3 ----0001 Numbering plan : ISDN numbering plan E.164
16:42:55.361 L3 4 00110000 Number digits : 0123456789
16:42:55.361 L2 U<-N 0 R 0 F=0 S:RR 128 65
16:42:55.361 L2 U<-N 0 C 0 P=0 I:INFO 128 65 68
16:42:55.361 L3 Q.931 CALL PROCEEDING U<=N Crn = 1 Dest
[/quote]

When I use the ‘fancy’ siemens phone though it does not have the ‘Called party number’ information element, but it shows a ‘Keypad’ information element

[quote]16:46:45.220 L3 1 00101100 INFORMATION ELEMENT : Keypad
16:46:45.236 L3 2 00001010 IE Length : 10 octet(s)
16:46:45.236 L3 30 31 32 33 34 35 36 37 38 39 : 0123456789
16:46:45.236 L2 U<-N 0 C 0 P=0 I:INFO 128 68 72
16:46:45.236 L3 Q.931 ALERTING U<=N Crn = 1 Dest
[/quote]

So… It seems asterisk does not ‘understand’ this form of dialling…

Is there anybody who could shed some light on this…

2

I wouldn’t expect that information to matter. Coming out of the PBX is a PRI. If you connect it to the phone company or into an Asterisk box should not make a difference (provided it is configured correctly).

What you are attempting to do, I have done with a Nortel Meridian 1 system. Took an Asterisk box, installed a quad PRI card, dual PRI card and a single PRI card. 3 PRI went to the PSTN, 2 to the Nortel, 1 to the Fax server (dialogic card) and the last one was for experimenting.

Are you able to plug the PRI into the Asterisk box and using a SIP phone on the Asterisk box, make outgoing calls? Or recieve calls to that phone?

You do have your PRI port to the PBX set up as signalling = pri_net and the PRI port to the PSTN as signalling = pri_cpe?

3

Hi Mazzic,

I’m happy to hear that something like this should work for Nortel, our second system is a meridian, so that will probably not pose any issues.

Outgoing calls work just fine over sip directly to the asterisk…

I’ve done some more research… The Q.931 (en.wikipedia.org/wiki/Q.931) specification allows for two ways of passing the called party number:

[quote]NOTE 6 – Either the Called party number or the Keypad facility information element is included by
the user to transfer called party number information to the network during overlap sending. The
Keypad facility information element may also be included if the user wants to convey other call
establishment information to the network, or to convey supplementary service information.[/quote]
For more info see the full specs @ itu.int/rec/T-REC-Q.931-199805-I/en

I will email Digium and ask if Asterisk is supposed to stick to this specification, and if the Keypad facility is supported…

4

The commercially supported version of Asterisk is based on a much older version of the code than the open source one.

5

Yes… I figured mailing Digium might not be the best option after all… Now looking for the best place to take this…