有沒有人在用CMPhone(信通網絡電話)軟件版?

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64^#

發表於 2012-7-27 10:14 | 只看該作者

我用的是 asterisk, 打出打入是用 bria / xlite / linksys pap2t -> asterisk -> cmphone

電腦超人

管理員

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63^#

發表於 2012-7-26 20:23 | 只看該作者

有一個很特別的, 如果我打出的電話也是 cmphone 用戶, 是可以成功打出的, 但如果對方是其他用戶, 便會出 50 ...
lawleo 發表於 2012-7-26 14:09

對了...你有否用X-LITE之類login?

我好像也試過...之後隔了一晚後再reload後好像OK......

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62^#

發表於 2012-7-26 14:09 | 只看該作者

有一個很特別的, 如果我打出的電話也是 cmphone 用戶, 是可以成功打出的, 但如果對方是其他用戶, 便會出 503 error

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61^#

發表於 2012-7-26 14:08 | 只看該作者

請問大家的 asterisk 是甚麼版本呢? 我正在用 1.8

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60^#

發表於 2012-7-23 23:08 | 只看該作者

>_< 都係唔得，

另加 external ip & localnet 都唔 work....攪乜呀....

ckleea

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59^#

發表於 2012-7-21 16:23 | 只看該作者

I do not have any problem in using my sip.conf.

bubblestar

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58^#

發表於 2012-7-21 13:10 | 只看該作者

本帖最後由 bubblestar 於 2012-7-21 13:11 編輯

回復 57# lawleo

Try this:

sip.conf

[general]
…..
pedantic = yes
…..

register = 852350xxxxx:password@202.0.179.3/852350xxxxx

[cmphone]
type=peer
host=202.0.179.3
port=5060
fromdomain=huawei.com
fromuser=852350xxxxx
realm=huawei
secret=your_password
username=852350xxxxx
insecure=port,invite
context=from-cmphone
authname=852350xxxxx
dtmfmode=auto
canreinvite=no
qualify=no

extensions.conf

[viaCMPhone]
exten => _888X.,1,Dial(SIP/cmphone/${EXTEN:3})
exten => _888X.,n,Hangup()

[from-cmphone]
exten => 852350xxxxx(你的 CMphone號碼),1,Dial(SIP/6001,30,Ttr)
exten => 852350xxxxx(你的CMphone號碼),n,Hangup()

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57^#

發表於 2012-7-21 09:30 | 只看該作者

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'

Qnewbie

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56^#

發表於 2012-7-21 02:06 | 只看該作者

本帖最後由 Qnewbie 於 2012-7-21 02:25 編輯

Outbound:
you might try
exten => _Z.,1,Dial(SIP/852XXXXXXXX/{EXTEN},,60)
there 852XXXXXXXX is your username in cmphone.

Inbound:
[from_cmphone]
exten => s,1,Dial(SIP/10&SIP/20&SIP/30&SIP/40,30)

Change the type=peer to
type = friend
in sip.conf

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55^#

發表於 2012-7-21 01:52 | 只看該作者

我開個 ssh 出來，有人可以幫我看看嗎？

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54^#

發表於 2012-7-21 01:47 | 只看該作者

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.

Rank: 1

53^#

發表於 2012-7-21 01:46 | 只看該作者

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)

Rank: 1

52^#

發表於 2012-7-21 01:40 | 只看該作者

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

lamsoft

Rank: 1

51^#

發表於 2012-7-20 15:40 | 只看該作者

回復 ckleea

請問
fromdomain=huawei.com
realm=huawei

是否必要?

如我是用 HKBN 的 BB100, 也用 ...
lawleo 發表於 2012-7-19 11:39

CMPhone 的register domain不是華為吧

it is my cmphone config
[COMNET_PSTN]
context = DID_COMNET_PSTN
host = 202.0.179.3
username = 8523XXXXXXX
fromuser = 8523XXXXXXX
realm = 8523XXXXXXX
trunkname = COMNET_PSTN
secret = ??????????????
hasiax = no
registeriax = no
hassip = yes
registersip = yes
hasexten = no
insecure = port,invite
disallow = all
allow = ulaw
qualify = no
transport = udp
canreinvite = no
type = friend
dtmfmode = rfc2833
hasvoicemail = no

context 指返去自己既incoming call dial plan
使用sip show registry檢查返是否成功登入cmphone.