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Aastra Paging


Detlef
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I verified it here and it worked fine:

 

[5] 2007/10/08 13:36:47: SIP Tx udp:192.168.0.155:5060:

INVITE sip:44@192.168.0.155 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.154:5060;branch=z9hG4bK-574d8fa8e8ff00be15c9d675fed0d3d5;rport

From: "41" <sip:41@test.pbxnsip.com>;tag=24656

To: "*9044" <sip:*9044@test.pbxnsip.com>

Call-ID: f259c65d@pbx

CSeq: 10040 INVITE

Max-Forwards: 70

Contact: <sip:44@192.168.0.154:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.0.2115

Call-Info: <sip:41@test.pbxnsip.com>;answer-after=0

Content-Type: application/sdp

Content-Length: 244

 

v=0

o=- 56981 56981 IN IP4 192.168.0.154

s=-

c=IN IP4 192.168.0.154

t=0 0

m=audio 60508 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[5] 2007/10/08 13:36:47: SIP Rx udp:192.168.0.155:5060:

SIP/2.0 200 OK

Call-ID: f259c65d@pbx

CSeq: 10040 INVITE

From: "41" <sip:41@test.pbxnsip.com>;tag=24656

To: "*9044" <sip:*9044@test.pbxnsip.com>;tag=596481733029d22

Via: SIP/2.0/UDP 192.168.0.154:5060;branch=z9hG4bK-574d8fa8e8ff00be15c9d675fed0d3d5;rport

Content-Length: 255

Call-Info: <sip:192.168.0.154>;appearance-index=1

Allow-Events: talk,hold,conference

Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO

Content-Type: application/sdp

Supported: replaces

Contact: "*9044" <sip:44@192.168.0.155>

User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5

 

v=0

o=MxSIP 0 1562308524 IN IP4 192.168.0.155

s=SIP Call

c=IN IP4 192.168.0.155

t=0 0

m=audio 3000 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

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When you do Intercom (*90xx) it should work. Paging as in Multicast paging will be a problem. What does the SIP INVITE packet say that goes to the Aastra phone?

 

The *90xxx works on my system too. I was just trying the Unicast SIP paging account and could not get any audio to play. The Aastra phones only show the caller ID on the display.

 

Also, is it possible to somehow enable the *90xx inbetween two PBXnSIP installations? I have one in the US and one in Mexico both have different extension number spaces and are connected via a SIP Gateway Trunk and a dial plan entry. Would that work if I add a dial plan entry that gets the *90 & EXT over to the other system?

 

Detlef

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The *90xxx works on my system too. I was just trying the Unicast SIP paging account and could not get any audio to play. The Aastra phones only show the caller ID on the display.

 

Then it is not a Aastra-specific problem... Must be something else, maybe a permission problem. Or just move to the latest and greatest 2.1.

 

Also, is it possible to somehow enable the *90xx inbetween two PBXnSIP installations? I have one in the US and one in Mexico both have different extension number spaces and are connected via a SIP Gateway Trunk and a dial plan entry. Would that work if I add a dial plan entry that gets the *90 & EXT over to the other system?

 

If you register a trunk then that should be possible (maybe you just create another trunk for this purpose). The incoming call from PBX1 will be treated like a regular extension call on PBX2 - then you can call whatever you like there.

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If you register a trunk then that should be possible (maybe you just create another trunk for this purpose). The incoming call from PBX1 will be treated like a regular extension call on PBX2 - then you can call whatever you like there.

 

Uh, that sounds easy, I could have had that idea. I'll try it if I get a chance.

 

As far as PBXnSIP version for the unicast paging I am using the 2.1.0.2114 currently. But I havent looked at the detailed log file why I dont get any audio with the unicast paging.

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