Detlef Posted October 5, 2007 Report Share Posted October 5, 2007 Has anyone got paging working with the Aastra phones? I am using Aastra 9133i with Firmware Version: 1.4.1.1077, Firmware Release Code: SIP, Boot Version: 1.1.0.10 All those phones do is display that it is paged but don?t play any audio. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 7, 2007 Report Share Posted October 7, 2007 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? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Report Share Posted October 8, 2007 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 Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 8, 2007 Author Report Share Posted October 8, 2007 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Report Share Posted October 8, 2007 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. Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 8, 2007 Author Report Share Posted October 8, 2007 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. Quote Link to comment Share on other sites More sharing options...
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