Voip Call Problem

Asterisk Asterisk General
1

Dear Asterisk Users,

I have registered a sip phone and a voip account to asterisk … and am calling on Sip phone which uses Voip account internally in the dialplan to call the number (a number maybe any number local or international )

But when i call the number … i can hear the destination user’s voice but the destination user cant hear my voice …
what could be the possible problem…
am only using 4 or 5 configuration files viz extensions.conf, sip.conf, indications.conf, rtp.conf, asterisk.conf, modules.conf

is these configuration files enough for a voip call?

Please guide where am going wrong

Following is my Configurations

Sip.conf

[general]
context=unauthenticated
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
register=>xxxxx:xxxxxx@sip.voipdiscount.com

sipphone
type=friend
context=LocalSets
host=dynamic
nat=no
;secret=poiuy
dtmfmode=auto
;disallow=all
;allow=ulaw
;allow=alaw
;canreinvite=no
allow=all
qualify=yes

[sip.voipdiscount.com]
type=peer
context=LocalSets
host=sip.voipdiscount.com
username=xxxxxxx
secret=xxxxxxx
dtmfmode=rfc2833
insecure=invite
;canreinvite=no
nat=yes
allow=all

Xlite001
secret=qwert

Extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[LocalSets]
exten=>_X.,1,Dial(SIP/${EXTEN}@sip.voipdiscount.com)

exten=>1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.voipdiscount.com)

Rtp.conf

[general]
rtpstart=10000
rtpend=20000

Asterisk.conf

directories
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
nocolor = yes ; Disable console colors.
documentation_language = en_US ; Set the language you want documentation

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

Indications.conf

[general]
country=in ; default location

[in]
description = India
ringcadence = 400,200,400,2000
dial = 40025
busy = 400/750,0/750
ring = 40025/400,0/200,400*25/400,0/2000
congestion = 400/250,0/250
callwaiting = 400/200,0/100,400/200,0/7500
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0/1000
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

2

If you’re successfully registering your device and are able to make / take a call on it, I would recommend looking into anything that might be intercepting the media stream (RTP) outbound from your network or inbound to the recipient’s network.

3

Thank you bmj for your immediate reply …

If you can elaborate on what you mean by " looking into anything that might be intercepting the media stream (RTP) outbound from your network or inbound to the recipient’s network "

the called(recipient) party is a gsm phone in US

thanks in advance

4

bmj is implying that this is likely to be a “Routing” issue.
The signaling to/from the provider seems to be fine since the call connects.
The codec negotiation also seems ok since you at least gets one way audio and that the call connects.

So you are trying to make a call through the provider to an external PSTN Phone.
This leaves us with a plausibility that someting is blocking the UDP Stream outbound from your home network.
Start with inspecting the firewall/router rules from “inside nets” to "ISP nets"regarding the RTP UDP ports.
Im not shure how possible NAT issues would translate in UDP terms but that could be worth looking in to.
Please let us know how you progress.

5

thanks for help guys …

i fixed it … the problem was with codecs in the xlite phone. i removed all codecs and kept only alaw an ulaw. and in sip.conf also i used disallow=all and allow=alaw and allow=ulaw