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So far the progress is dialing in works.

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

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

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

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

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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.

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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

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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.

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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

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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.

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回復 40# bubblestar


    Will try again later to see.

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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.

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我都想申請 CM Phone, 可以係深圳用嗎?

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以前有member就是这样做,用普通的Linksys ATA就可以搞定。

角色

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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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