角色

因为我没有CMPhone，可能要能bubblestar师兄给你打打脉。

角色

bubblestar

回復 11# ttmuskie


   
I use your same dialplan and give it a try.  There is no problem as you mentioned.  The message from CLI is as below:

== Using SIP RTP CoS mark 5
    -- Executing [852350XXXXX@from-cmphone:1] Playback("SIP/cmphone-00000008", "demo-echotest") in new stack
    -- <SIP/cmphone-00000008> Playing 'demo-echotest.ulaw' (language 'en')    -- Executing [852350XXXXX@from-cmphone:2] Hangup("SIP/cmphone-00000008", "") in new stack
  == Spawn extension (from-cmphone, 852350XXXXX, 2) exited non-zero on 'SIP/cmphone-00000008'


I can hear the playback before it hangs up.

The only message difference that I got is marked in RED above.  Hence, I think you may put the codec allow = ulaw in the first place of general section in sip.conf and try again.

ttmuskie

Thanks bublestar, tried the suggestion. But it's still the same output as I posted in the previous.

bubblestar

Despite of the echo test problem, can you get the normal dial in call properly?

ttmuskie

Under asterisk, both incoming and outgoing not working.
Under sip client (not connect to asterisk), incoming ok and outgoing not working.

I called cmphone's cs and they are now looking into my issue.

ckleea

Please show your sip.conf

sleepsheep

回復 20# ttmuskie


    what if you using software on PC? say their openeye (xp) or Xlite (win 7)? to make sure the account itself works for your network? sometimes the firewall lead to such problem, if that, you won't be able to use it normally on PC or any other facility.

ttmuskie

The CS from cmphone just asked me to use other program. And I tried the xlite and still no joy.

ckleea

I can use the old cmphone account in my siemens ip phone. Soft phone is also working.

角色

Perhaps， the problem comes from the router. Try to change another router to see if any improvement can be made.

YH

ckleea

Has it been fixed?

No problem here

ttmuskie

No, still not working. May try again after when i get the mood

ckleea

回復 27# ttmuskie

Tell us once again which sip software/ client or ATA you used?

I have no problem in using it with iphone apps, pc zoiper, ATA, siemens ip phone and even asterisk.
Just set the three parameters provided.

ttmuskie

asterisk 1.8.7.1
ekiga 3.26 on linux
csipsimple on android

ckleea

回復 29# ttmuskie


    I have no experience with both clients you use. But if you are using the setup as you asked from bubblestar. It should work both ways. can you show your [general] of sip.conf?