返回列表 發帖
No dial out. In sip.conf, you pedantic = yes.
Standarr register string

TOP

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.

TOP

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

TOP

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.

TOP

回復 40# bubblestar


    Will try again later to see.

TOP

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.

TOP

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

TOP

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

TOP

返回列表