電訊茶室 - Powered by Discuz! Board

標題: VOIP in -> SPA3000/SPA3101 -> call forward to another VOIP account [打印本頁]

作者: ckleea 時間: 2011-6-14 15:49 標題: VOIP in -> SPA3000/SPA3101 -> call forward to another VOIP account

Does anyone know if this is possible? Normally it is call forward to PSTN line.

作者: bubblestar 時間: 2011-6-14 16:53

Yes, you can.

Enter into User 1 Tab of SPA3000/SPA3102. This is where you can find Call Forward Settings and Selective Call Forward Settings .

作者: ckleea 時間: 2011-6-14 18:21

Have you tried? I cannot make it work.

作者: bubblestar 時間: 2011-6-14 19:49

本帖最後由 bubblestar 於 2011-6-14 21:43 編輯

Here you are. To make VoIP call forwarding work, Caller ID feature of your telephone set and service must be enabled and available.

圖片附件: VoIP_Forward.png (2011-6-14 19:48, 77.28 KB) / 下載次數 724
http://telecom-cafe.com/forum/attachment.php?aid=807&k=08eaa5e1a0618f219fcc212682a3fef7&t=1732590226&sid=ppffFh

作者: Skypeus 時間: 2011-6-14 21:19

有无详细教程？

作者: bubblestar 時間: 2011-6-14 21:48

本帖最後由 bubblestar 於 2011-6-14 21:50 編輯

如果你需要整體SPA3000/SPA3102 的詳細教程，可以參考另文的一位師兄所編寫之Admin Guide，它結集了這裡多位高手的精華。

http://www.telecom-cafe.com/foru ... =3272&pid=11209

作者: ckleea 時間: 2011-6-15 06:30

回復 4# bubblestar

My CLI is like this

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [6299@DLPN_DP1:1] Dial("SIP/6100-00000035", "SIP/6299") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/6299
-- Got SIP response 302 "Moved Temporarily" back from 192.168.xxx.xxx:5060
-- Now forwarding SIP/6100-00000035 to 'SIP/6100::::UDP@192.168.xxx.yyy' (thanks to SIP/6299-00000036)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Jun 15 06:22:58] NOTICE[9725]: app_dial.c:851 do_forward: Not accepting call completion offers from call-forward recipient SIP/192.168.118.30-00000037
-- Got SIP response 482 "Loop Detected" back from 192.168.xxx.yyy:5060
-- SIP/192.168.118.30-00000037 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Timeout on SIP/6100-00000035

作者: bubblestar 時間: 2011-6-15 11:04

回復 7# ckleea

Mine in the CLI is as below:

== Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [2007@home:1] Macro("SIP/2002-00000006", "normal,SIP/2007,2007") in new stack
-- Executing [s@macro-normal:1] Dial("SIP/2002-00000006", "SIP/2007,40,m") in new stack
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/2007
-- Started music on hold, class 'default', on SIP/2002-00000006
-- Got SIP response 302 "Moved Temporarily" back from 192.168.XXX.YYY:5076
-- Now forwarding SIP/2002-00000006 to 'Local/2001@home' (thanks to SIP/2007-00000007)
[2011-06-15 10:57:14] NOTICE[10883]: app_dial.c:855 do_forward: Not accepting call completion offers from call-forward recipient Local/2001@home-1a09;1
-- Stopped music on hold on SIP/2002-00000006
-- Executing [2001@home:1] Macro("Local/2001@home-1a09;2", "normal,SIP/2001,2001") in new stack
-- Executing [s@macro-normal:1] Dial("Local/2001@home-1a09;2", "SIP/2001,40,m") in new stack
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/2001
-- Started music on hold, class 'default', on Local/2001@home-1a09;2
-- Local/2001@home-1a09;1 is making progress passing it to SIP/2002-00000006
-- SIP/2001-00000008 is ringing
-- SIP/2001-00000008 answered Local/2001@home-1a09;2
-- Stopped music on hold on Local/2001@home-1a09;2
-- Local/2001@home-1a09;1 answered SIP/2002-00000006
-- Executing [h@home:1] Verbose("Local/2001@home-1a09;2", " - Ending the call - ") in new stack
- Ending the call -
-- Executing [h@home:2] ResetCDR("Local/2001@home-1a09;2", "") in new stack
-- Locally bridging SIP/2002-00000006 and SIP/2001-00000008
-- Executing [h@home:3] Hangup("Local/2001@home-1a09;2", "") in new stack
  == Spawn extension (home, h, 3) exited non-zero on 'Local/2001@home-1a09;2'
  == Spawn extension (macro-normal, s, 1) exited non-zero on 'Local/2001@home-1a09;2' in macro 'normal'
  == Spawn extension (home, 2001, 1) exited non-zero on 'Local/2001@home-1a09;2'
-- Executing [h@home:1] Verbose("SIP/2002-00000006", " - Ending the call - ") in new stack
- Ending the call -
-- Executing [h@home:2] ResetCDR("SIP/2002-00000006", "") in new stack
-- Executing [h@home:3] Hangup("SIP/2002-00000006", "") in new stack
  == Spawn extension (home, h, 3) exited non-zero on 'SIP/2002-00000006'
  == Spawn extension (macro-normal, s, 1) exited non-zero on 'SIP/2002-00000006' in macro 'normal'
  == Spawn extension (home, 2007, 1) exited non-zero on 'SIP/2002-00000006'

作者: bubblestar 時間: 2011-6-15 11:13

試一試在SPA3000 User 1 TAB 裡面的Cfwd All Dest 直接放入最終收電話的extension@real_internal_ip_address:port_number 。不要放入Asterisk Server 的 IP 看看結果怎樣?

作者: ckleea 時間: 2011-6-15 11:24

回復 9# bubblestar

唔成功，其實我都看到你的 log，不過當接到最後 extension時就出 loop detected。如果去出面VOIP，asterisk 就唔識接。

作者: bubblestar 時間: 2011-6-15 11:34

Someone out there has similar problem.

QUOTE:

a 482 error code means that a request has been routed back to the proxy that previously forwarded it. Each server that forwards a request adds a Via header with its address in the top of the request.

I'm guessing that your polycom registers with openser. a call comes in from your provider to openser, openser routes to asterisk, which routes it right back to openser to deliver it to the polycom, and there is your loop.

UNQUOTE:

Try reading this to see if it has any clue for you.

http://www.voipuser.org/forum_topic_6556.html

作者: bubblestar 時間: 2011-6-15 11:36

該提問者已把問題解決，但卻沒有說詳情。

歡迎光臨電訊茶室 (http://telecom-cafe.com/forum/)