Hi,

I have signed up for pfingo which offers a VOIP trunk which enables me to receive and place calls in one region. My first setup was fine I can make make and receive calls perfectly. But after some time, I can no longer place any call. I can place call and it will ring on the other end and I can hear the other end's voice from a IP phone. However, he (from PSTN) cannot hear anything I say.

I thought this was a NAT/RTP issue so I did some tests. However, when the party called my DID(incoming) I am able to speak to him just fine without any issue. Is it possible that the Provider itself is blocking the conversation making it appear like a one way voice problem? Here are the test scenarios:

1.) IP Phone to PSTN --> Voice can be heard on the IP Phone but the PSTN guy cannot hear anything
2.) PSTN --> IP Phone --> The same.Voice can be heard on the IP Phone but PSTN cannot hear anything
3.) Incoming directly from DID number --> Voice can be heard both ways

Am I missing something here? Here is my sip.conf:

;!
;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Tue Nov 17 10:18:42 2009
;!

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
useragent = IP0x
allowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = yes
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain =
dtmfmode =
dumphistory = no
externhost = xxx.dyndns.org
externip = xxx.dyndns.org
externrefresh = 10
fromdomain =
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize =
jbresyncthreshold =
language =
localnet = x.x.x.x/x.x.x.x
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = no
rtpholdtimeout =
rtptimeout =
sendrpid = no
sipdebug = no
subscribecontext =
t1min = 100
t38pt_udptl = no
tos_audio = none
tos_sip = none
tos_video = none
trustrpid = no
usereqphone = no
disallow = all
allow = undefined,ulaw,alaw,gsm,speex,g726,g729

This is because there is a bug in asterisk to fix it do the following
in /etc/asterisk/extensions.conf
under external context put

exten => s,1,Playback(silence/1)

that should resolve it, if it doesnt let me know.

Hi,

I was able to resolve the problem by restarting my NAT router. Thanks a lot for the advice. Though my problem is now is I cannot get my incoming DID to route to IVR. But when in Ring all seems to work fine.