有沒有人在用CMPhone(信通網絡電話)軟件版? - Hard/Soft Phone - VoIP 研究室 - 電訊茶室

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發表於 2012-7-17 23:56 | 顯示全部帖子

回復 42# ckleea

原來 ckleea 也用 cmphone 的，請問現在你成功了嗎？
asterisk 的參數如何呢？試了幾天，很灰比了。

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2^#

發表於 2012-7-19 00:36 | 顯示全部帖子

回復 46# ckleea

thx....

以上的 setting 我也借過來用過，可惜結果一樣，看來是其他問題了。

剛好 raspberry pi 出了新的 raspbian 版本，正在安裝，希望再 compile 後會有點不同的結果。

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3^#

發表於 2012-7-19 11:39 | 顯示全部帖子

回復 46# ckleea

請問
fromdomain=huawei.com
realm=huawei

是否必要?

如我是用 HKBN 的 BB100, 也用以上的嗎?
我capture過 PAP2T 的 packet, 發覺他也是用 huawei.com & huawei

只奇怪為甚麼要用 huawei 呢?

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4^#

發表於 2012-7-19 11:44 | 顯示全部帖子

extension.conf

請問撥出電話, 你是用那一句呢? a or b?
[local]
;a) exten => _[1-9].,1,Dial(SIP/VoIPProvider/${EXTEN})
;b) exten => _[1-9].,1,Dial(SIP/${EXTEN}@VoIPProvider)

兩句我都試過, 也是 503 error

[from-cmphone]
exten => 85235021085,1,Dial(SIP/10&SIP/20,30)

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5^#

發表於 2012-7-20 15:30 | 顯示全部帖子

wu wu... 試左好耐都唔成功, 有無人可以救下我呀...

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6^#

發表於 2012-7-21 01:40 | 顯示全部帖子

試左都還是不能.
raspberrypi*CLI> sip show registry
Host                                  dnsmgr Username    Refresh State             Reg.Time
202.0.179.3:5060                      N    85235021085       105 Registered          Sat, 21 Jul 2012 01:21:16

這部份應該登入成功，但當我一打出時....
  == Using SIP RTP CoS mark 5
-- Executing [18503@users:1] Dial("SIP/30-00000011", "SIP/18503@cmphone") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/18503@cmphone
-- Got SIP response 503 "Service Unavailable" back from 202.0.179.3:5060
-- SIP/cmphone-00000012 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/30-00000011' status is 'CONGESTION'

立即 503 error, rejected

打入時都係差唔多
  == Using SIP RTP CoS mark 5
-- Called SIP/40
-- SIP/40-00000003 is ringing
-- SIP/40-00000003 answered SIP/cmphone-00000000
-- Locally bridging SIP/cmphone-00000000 and SIP/40-00000003
  == Spawn extension (from_cmphone, 85235021085, 1) exited non-zero on 'SIP/cmphone-00000000'
[Jul 21 01:31:24] WARNING[4395]: chan_sip.c:3639 retrans_pkt: Retransmission timeout reached on transmission 05c852db110d192113947b295dfca5c4@192.168.2.233:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

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7^#

發表於 2012-7-21 01:46 | 顯示全部帖子

我的登入句法是
register=8523502xxxx:xxxxxxxx@202.0.179.3/8523502xxxx <- 放在 sip.conf 裡的 general

dialplan for 打出打入 in extension.conf
[local]
exten => _[1-9].,1,Dial(SIP/cmphone/${EXTEN}) <- 唔work
;exten => _[1-9].,1,Dial(SIP/${EXTEN}@cmphone) <- 都唔 work

[users]
include => local

[from_cmphone]
exten => 8523502xxxx,1,Dial(SIP/10&SIP/20&SIP/30&SIP/40,30)

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8^#

發表於 2012-7-21 01:47 | 顯示全部帖子

以上的句法有沒有問題呢？
岩岩把 asterisk 都放在 DMZ 裡，情況一樣呢.

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9^#

發表於 2012-7-21 01:52 | 顯示全部帖子

我開個 ssh 出來，有人可以幫我看看嗎？

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10^#

發表於 2012-7-21 09:30 | 顯示全部帖子

還是不能....鞋...相同的 error message

撥出
= Using SIP RTP CoS mark 5
-- Executing [18503@users:1] Dial("SIP/40-00000001", "SIP/85235021085/18503,,60") in new stack
[Jul 21 09:14:27] WARNING[4838]: chan_sip.c:5440 create_addr: Purely numeric hostname (85235021085), and not a peer--rejecting!
[Jul 21 09:14:27] WARNING[4838]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/40-00000001' status is 'CHANUNAVAIL'
  == Using SIP RTP CoS mark 5
-- Executing [85235021085@from_cmphone:1] Dial("SIP/cmphone-00000002", "SIP/10&SIP/20&SIP/30&SIP/40,30") in new stack
-- SIP/40-00000004 answered SIP/cmphone-00000002
-- Locally bridging SIP/cmphone-00000002 and SIP/40-00000004
  == Spawn extension (from_cmphone, 85235021085, 1) exited non-zero on 'SIP/cmphone-00000002'