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標題: 有沒有人在用CMPhone(信通網絡電話)軟件版? [打印本頁]

作者: 電腦超人 時間: 2011-2-16 13:08 標題: 有沒有人在用CMPhone(信通網絡電話)軟件版?

CMPhone是有軟件版和ATA版的...
軟件版是有些參數提供...不知可否放在asterisk裡使用呢?

軟件版現在$50/月...
如果可以的話...也可取代2b...(或多一組call-in號碼使用)

雖然比nwt(新世界)的貴一點...
但nwt要拿到參數應該很困難...(加一點運氣吧

)

作者: 雯雯 時間: 2011-2-16 13:30

回復 1# 電腦超人

CMPhone軟件版我有在用, 參數應該可以放落asterisk!

作者: 電腦超人 時間: 2011-2-16 13:40

回復電腦超人

CMPhone軟件版我有在用, 參數應該可以放落asterisk!
雯雯發表於 2011-2-16 13:30

今天忽然想起CMPhone...所以便出帖問了...

質素可以嗎?

作者: 雯雯 時間: 2011-2-16 13:42

回復 3# 電腦超人

質素可以.

作者: ckleea 時間: 2011-2-16 14:19

可否用於asterisk 當 sip trunk 用。

作者: 電腦超人 時間: 2011-2-16 16:39

我就是想問這個...

看了一點安裝...設定看來可以的...

email問了一下...要申請的話一天便可以了...

作者: 電腦超人 時間: 2011-2-16 16:40

安裝DIY及使用指南
http://www.cmphone.com/big5/c_softphone_installation_guide.html

作者: ckleea 時間: 2011-2-16 16:58

如果可以的話，先裝係softphone, zoiper my favorite 試下先

作者: 角色 時間: 2011-2-16 18:17

如果2b到期不减价，我都考虑用CM的service。

角色

作者: ckleea 時間: 2011-2-16 18:31

回復 6# 電腦超人
要信用卡和地址證明

作者: ckleea 時間: 2011-2-17 08:59

我將會申請一個試，主要做overseas backup，因為最近的2b 同 network 成日有問題！

作者: 角色 時間: 2011-2-17 09:07

我將會申請一個試，主要做overseas backup，因為最近的2b 同 network 成日有問題！ ...
ckleea 發表於 2011-2-17 08:59

希望有你的CM Phone的Asterisk测试报告，那么我下次跟2b说就可以“大”她了。

角色

作者: bubblestar 時間: 2011-2-17 10:07

回復 11# ckleea

咁樣你就可以同時多幾條不同Providers 的TRUNK，測試Super Failover Trunk 的可靠度及準繩度了。

另外，是否要也要簽一年約的?

作者: ckleea 時間: 2011-2-17 10:43

要一年，不過HK$50都唔錯，特別是在海外用

作者: ckleea 時間: 2011-2-17 13:49

可以在其他client 裝和用，暫時上唔asterisk

Zoiper 冇問題

作者: 角色 時間: 2011-2-17 13:53

如果用自己的ATA，不知道CM是否能提供SIP 参数呢？

角色

作者: 雯雯 時間: 2011-2-17 14:05

回復 16# 角色

CMPhone本身就是提供SIP參數, 可以放在ATA上用! 我把它放了在我部Draytek Vigor 2920vn.

作者: 角色 時間: 2011-2-17 14:08

谢谢雯雯小师妹的信息，你用ＣＭ　ｐｈｏｎｅ多久呢？不知道新的ｐｌａｎ是否也能提供ＳＩＰ参数呢？

如果有的时候，那么就会吸引更加多的ｍｅｍｂｅｒ参与。

角色

作者: ckleea 時間: 2011-2-17 14:13

software version provides the user account, password and server address (SIP)

Zoiper can register and dial. I try on Siemens phone already. But not working in asterisk. Authenticate failed. Time out

作者: 雯雯 時間: 2011-2-17 14:15

回復 18# 角色

我用了CMPhone約3個月, 它的plan到現時為止暫時未有改變.

作者: ckleea 時間: 2011-2-17 14:18

I use this in my sip.conf

[cmphone]
type = friend
secret = secret
username = 8523xxxxxxx
host = 202.0.179.3
fromuser = 8523xxxxxxx
authuser = 8523xxxxxxx
disallow = all
allow = ulaw
allow = alaw
qualify = yes
nat = yes
canreinvite = yes
context = from-cmphone
outboundproxy = 202.0.179.3
insecure = port, invite

作者: 雯雯 時間: 2011-2-17 14:19

回復 19# ckleea

ckleea兄, 會不會是你的Asterisk setting出了問題? 不過就算不能把它放在Asterisk, 也可以把它放在ATA當作PSTN連接Asterisk!

作者: ckleea 時間: 2011-2-17 14:27

其他trunks冇問題

作者: 角色 時間: 2011-2-17 14:43

回復 21# ckleea

你有CM Trunk？

角色

作者: ckleea 時間: 2011-2-17 14:47

回復 24# 角色

剛剛試加上去，唔work。

因為已試過用software and Siemens phone, 全部work。

理論上，加入 asterisk應該 work

作者: bubblestar 時間: 2011-2-17 15:20

試一試 canreinvite = no

作者: ckleea 時間: 2011-2-17 15:25

Will try the various settings tonight. Hope to get it running.

FYI, the quality of voice is quite good. I put in UK Siemens Phone and work flawlessly. No latency or echo.

作者: ckleea 時間: 2011-2-18 14:48

Some one use this setting

register => 85235015001:secretpwd@202.0.179.3

[35015001-cm]
type=peer
host=202.0.179.3
fromdomain=huawei.com
fromuser=85235015001
secret=secretpwd
username=85235015001
insecure=very
context=from-pstn
authname=85235015001
dtmfmode=auto
canreinvite=no

作者: ckleea 時間: 2011-2-18 18:23

Can registered now but can't dial in and out.

作者: bubblestar 時間: 2011-2-18 18:33

Sometimes, registration and dial may need this, case by case.

register => 85235015001:secretpwd@202.0.179.3/85235015001

type=friend
insecure=port,invite
qualify=yes

作者: ckleea 時間: 2011-2-19 07:23

So far the progress is dialing in works.

作者: ckleea 時間: 2011-2-19 11:00

As far as I can tell, CMPhone allows one line in

作者: bubblestar 時間: 2011-2-19 11:08

At last, which sip configuration settinig you use to make it work. Could you share with us?

作者: ckleea 時間: 2011-2-19 11:17

No dial out. In sip.conf, you pedantic = yes.
Standarr register string

作者: 電腦超人 時間: 2011-2-19 23:35

ckleea兄~我想問你的asterisk是建在linux上嗎?
有試過連其他trunk嗎?

作者: ckleea 時間: 2011-2-20 06:01

My asterisk is built on linux. I have 1 hk2b, a analog line through SPA3000 FXO out, several VOIP accounts e.g. IPtel. All working except 2b that fail intermittently.

For CMphone, it can be registered as a client when coded in sip.conf. It rings and can be dialed in, For out, its status is never reachable.

作者: ckleea 時間: 2011-2-20 06:08

The parameter available from cmphone includes

1, HK tel number
2. regustrar/backup server ip
3. user id - 852+ the HK number
4. password.

I can use them to login in their server through 2 pieces of thing.
Zoiper
Siemens IP phone

In and out, no problem with good quality

For asterisk, I need to put pedantic = yes
in [general] of sip.conf
and the usual register string
userid:password@hostip

I don't know the timeout period for cmphone, but at least few minutes for reconnection

作者: ckleea 時間: 2011-2-20 06:12

It allows only one login at one time. As far as I can tell, only one call any time. This is difficult from 2b and some other voip providers.

作者: 角色 時間: 2011-2-20 09:40

CK

Are you able to change the Agent name to other name instead of Asterisk when you pass the information to CM server? I guess CM rejects outbound call if the user agent is Asterisk.

YH

作者: bubblestar 時間: 2011-2-20 10:54

Yes, I recall that Asterisk The Future of Telephony also mentions that some servers might not accept Asterisk as user Agent name for unknown reasons. You may change whatever name you like other than Asterisk and try again. No harm.

作者: ckleea 時間: 2011-2-20 14:13

回復 40# bubblestar

Will try again later to see.

作者: ckleea 時間: 2011-2-21 22:41

Change the user agent does not work

May need to think about this as well

Phone registers, but I can't receive calls

This problem is most likely to happen if the phone is behind a NAT router, thus loosing its connectivity to the mydivert.com server.

While default phone settings work correctly in environments without a NAT, for phones behind a NAT you must change the phone time-out period - this is the amount of time after which the phone tries to register again to the server.

Most phones have a Registration expires/Re-register timeout/Registration timeout setting. The name varies, but the function is always the same. Default values are 1 hour or 3,600 seconds.

While this is alright for typical connections that are normally closed after 7,200 seconds, for connections behind NAT the value must be set to 60 seconds or 1 minute, or, in any case, lower then 120 seconds. This is mandatory because most routers close the connection after 120 seconds and when a call comes from a public IP after this period of time, the router just drops it since it does not know what to do with the packets.

Check the phone advanced settings. Set a low registration period and check to see if it offers NAT keep-alive options or other helpers.

The problem may also be caused by router settings. So, it's probably best to try different settings. If nothing else works, consider using a STUN server (there are public STUN servers available on the net, example stunserver.org, or use stun.mydivert.com).

Check your firewall/router

Many registration problems are caused by firewall applications. To make sure your problem is not caused by the firewall, open all the VoIP ports on the firewall/router in front of the phone. If you want to create strict rules, then make sure that at least the UDP ports 5060-5070, 10000-20000 and 53 are not blocked.

Log into your router or modem/router administration. There may be options available to enable NAT support.

作者: dingli 時間: 2011-3-6 01:43

我都想申請 CM Phone, 可以係深圳用嗎?

作者: 角色 時間: 2011-3-6 07:42

以前有member就是这样做，用普通的Linksys ATA就可以搞定。

角色

作者: lawleo 時間: 2012-7-17 23:56

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。

作者: ckleea 時間: 2012-7-18 21:52

回復 45# lawleo

The setting in sip.conf is

[cmphone]
type=peer
host=202.0.179.3
port=5060
fromdomain=huawei.com
fromuser=8523xxxxxxx
realm=huawei
secret=you password
username=8523xxxxxxx
insecure=port,invite
context=from-cmphone
authname=8523xxxxxxxx
dtmfmode=auto
canreinvite=no
qualify=no

Please look up further in this forum for direction

作者: lawleo 時間: 2012-7-19 00:36

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。

作者: lawleo 時間: 2012-7-19 11:39

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?

作者: lawleo 時間: 2012-7-19 11:44

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)

作者: lawleo 時間: 2012-7-20 15:30

wu wu... 試左好耐都唔成功, 有無人可以救下我呀...

作者: lamsoft 時間: 2012-7-20 15:40

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.

作者: lawleo 時間: 2012-7-21 01:40

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

作者: lawleo 時間: 2012-7-21 01:46

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)

作者: lawleo 時間: 2012-7-21 01:47

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.

作者: lawleo 時間: 2012-7-21 01:52

我開個 ssh 出來，有人可以幫我看看嗎？

作者: Qnewbie 時間: 2012-7-21 02:06

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

作者: lawleo 時間: 2012-7-21 09:30

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'

作者: bubblestar 時間: 2012-7-21 13:10

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()

作者: ckleea 時間: 2012-7-21 16:23

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

作者: lawleo 時間: 2012-7-23 23:08

>_< 都係唔得，

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

作者: lawleo 時間: 2012-7-26 14:08

請問大家的 asterisk 是甚麼版本呢? 我正在用 1.8

作者: lawleo 時間: 2012-7-26 14:09

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

作者: 電腦超人 時間: 2012-7-26 20:23

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......

作者: lawleo 時間: 2012-7-27 10:14

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

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