comnet phone outgoing not ok

ttmuskie

Just registered and have a trial run. Incoming no problem but can't make any Outgoing call.

ttmuskie

Android csipsimple and Linux ekiga

ttmuskie

don't have any ata

ttmuskie

I want to try it under asterisk. Any brother can share the sip.conf and extension.conf? Thanks.

ttmuskie

Thanks. Just sent.

ttmuskie

I setup asterisk to play a DEMO sound for cmphone. When I called 35012565, it seems try to play to sound file but it immediate hang up. Here is the detail:

extensions.conf

1. [from_cmphone]
   
2. exten => 85235012565,n,Playback(demo-echotest)
   
3. exten => 85235012565,n,Hangup()

複製代碼

sip.conf

1. [cmphone]
   
2. type=peer
   
3. host=202.0.179.3
   
4. port=5060
   
5. fromdomain=huawei.com
   
6. fromuser=85235012565
   
7. realm=huawei
   
8. secret=XXXX
   
9. username=85235012565
   
10. insecure=port,invite
    
11. context=from_cmphone
    
12. authname=85235012565
    
13. dtmfmode=auto
    
14. canreinvite=no
    
15. qualify=no

複製代碼

cli

1. == Using SIP RTP CoS mark 5
   
2.     -- Executing [85235012565@from_cmphone:1] Playback("SIP/cmphone-00000000", "demo-echotest") in new stack
   
3.   == Spawn extension (from_cmphone, 85235012565, 1) exited non-zero on 'SIP/cmphone-00000000'

複製代碼

ttmuskie

yeah.. i copied the old config . it should be:

exten => 85235012565,1,Playback(demo-echotest)

And it is still the same problem. You could try to call the 35012565 and know what I mean.

ttmuskie

Have that under general already:

[general]
pedantic=yes
register=85235012565:XXXX@202.0.179.3/85235012565

ttmuskie

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

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.

ttmuskie

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

ttmuskie

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

ttmuskie

asterisk 1.8.7.1
ekiga 3.26 on linux
csipsimple on android

ttmuskie

Note: I'm using port 9060 rather than the default 5060

[general]
bindport=9060
pedantic=yes
register=85235012565:my_pwd@202.0.179.3/85235012565

sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
202.0.179.3:5060                        N      85235012565        105 Registered           Sun, 11 Dec 2011 14:05:29
1 SIP registrations.

call to 28805522
cli>
== Using SIP RTP CoS mark 5
    -- Executing [28805522@default:1] Dial("SIP/nexus-00000006", "SIP/cmphone/28805522") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/cmphone/28805522
    -- Got SIP response 503 "Service Unavailable" back from 202.0.179.3:5060
    -- SIP/cmphone-00000007 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [28805522@default:2] Hangup("SIP/nexus-00000006", "") in new stack
  == Spawn extension (default, 28805522, 2) exited non-zero on 'SIP/nexus-00000006'

ttmuskie

1. enabled the DMZ to my machine.
2. changed the bindport=5060
3. restarted asterisk
4. call from my mobile to asterisk and here the result with debug enabled:

1. <--- Transmitting (NAT) to 202.0.179.3:5060 --->
   
2. SIP/2.0 200 OK
   
3. Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK9b25a61f7;received=202.0.179.3;rport=5060
   
4. From: <sip:my_mobile@202.0.179.3;user=phone>;tag=615e35ff
   
5. To: <sip:85235012565@119.236.110.45;user=phone>;tag=as0035d85b
   
6. Call-ID: 3f4457e5135ab023d1b27dba2066ae65@sx3000
   
7. CSeq: 2 BYE
   
8. Server: Asterisk PBX 1.8.7.1
   
9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
   
10. Supported: replaces, timer
    
11. Content-Length: 0
    
12. 
    
13. 
    
14. <------------>
    
15.   == Spawn extension (from_cmphone, 85235012565, 1) exited non-zero on 'SIP/cmphone-00000001'

複製代碼