Telbo.com: one way audio with PAP2-T and asterisk

Asterisk Asterisk Support
1

Hi, i’ve a problem with pap2t, if i call to a telephone number from a device connected to pap2t the receiver can’t hear me, while if i call from sipdroid or Ekiga, all works fine.

In [general] settings i’ve this:
disallow=all
allow=alaw

I tried with ulaw and gsm, and it seems to have always the same problem.

If i make a call between the internal phones (eg: pap2t to sipdroid), it works, but it seems a problem between asterisk and pap2t.
I tried to call from telbo connecting to sip.telbo.com with pap2t and it works, while if i use asterisk it doesn’t work.

Here there is the controll panel of pap2t:

I tried G711u too, but doesn’t work.

Thanks

2

Codec problems tend to cause the call to be dropped immediately, not one way audio. Why do you believe you have a codec problem?

The only codec problem that might produce one way audio is if you negotiate a non-free codec and don’t have enough licences.

3

and so is not a codec issue… but i can’t undestand why pap2t can call an internal phone and works fine while if i call an phone number it doesn’t work… there is some misconfiguration in my pap2t or in asterisk?
The other devices work fine with telbo.com

Telbo: telbo.com/sip
my sip.conf:

[code][general]
port=5060
bindaddr=0.0.0.0
context=sipin
nat=yes
disallow=all
allow=alaw
realm=sip.messagenet.it
rtptimeout=60
rtpholdtimeout=300

nat=yes
externip=151.63.21.4
localnet=192.168.1.105/255.255.255.0

; ---------------- PROVIDER VOIP -----------------
register => :@sip.messagenet.it:5061/
register =>:@sip.telbo.com/*******
register => *****:*@sip.voipstunt.com/

[telbo]
type=peer
context=chiamate-uscita
username=*********
fromuser=*********
secret=*********
host=sip.telbo.com
fromdomain=sip.telbo.com
qualify=yes
insecure=invite,port
nat=yes

[voipstunt]
type=peer
context=chiamate-uscita
username=*********
fromuser=*********
secret=*********
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
qualify=yes
insecure=invite,port
nat=yes
;--------------------------------------------------

; ---------------- ACCOUNT LOCALI -----------------
; Telefono fisso - PAP2T
[2000]
type=friend
username=2000
secret=****
context=telefoni-locali
host=dynamic

;Galaxy S - Sipdroid
[2010]
type=friend
username=2010
secret=****
context=telefoni-locali
host=dynamic

;Admin-PC
[2100]
type=friend
username=2100
secret=****
context=telefoni-locali
host=dynamic

;XFX-PC
[2101]
type=friend
username=2101
secret=****
context=telefoni-locali
host=dynamic[/code]

my extensions.conf:

[code][telefoni-locali]

; Telefono fisso - PAP2T
exten => 2000,1,Dial(SIP/2000,20,Tt)

;Galaxy S - Sipdroid
exten => 2010,1,Dial(SIP/2010,20,Tt)

;Admin-PC
exten => 2100,1,Dial(SIP/2100,20,Tt)

;XFX-PC
exten => 2101,1,Dial(SIP/2101,20,Tt)

;Chiamate in uscita
exten => _0X.,1,Dial(SIP/0039${EXTEN}@telbo)
exten => _3X.,1,Dial(SIP/0039${EXTEN}@telbo)
exten => _00X.,1,Dial(SIP/${EXTEN}@voipstunt)

[sipin]
exten => _********,1,Dial(SIP/2000)[/code]

pap2t control panel: docs.google.com/open?id=0B7AuYU … GprdDBVRGM
pap2t sip config: docs.google.com/open?id=0B7AuYU … UN5cmZLejg

4

I much prefer to have listings, etc. embedded in the posting (probably using code, at least for longer ones). Looking at the titles of those presented, I see nothing that looks like it might be a full log, so, whilst I might spot an error in the configuration, it would be a lot more difficult because I don’t know what is actually going wrong.

In the mean time note that the number one cause of one way audio is misconfigured routers.

5

Ok, i edited the post…

there is a misconfiguration in my router? like closed doors? But the other devices connected to the same router can call :neutral_face:

Thanks

6

Solved! :smiley:
append in the [general] section:

directmedia=no

7

It’s normally better to put directmedia = no for just those devices that don’t support directmedia, typically those outside the NAT boundary.

8

Ok, i put it on section [2000]

Thank you!