Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS
(channel 1). My SIP extension phone located inside the LAN is a SNOM
300 IP phone.
This one way audio problem only happens when the SIP extension phone
(let's call it the CALLER) places an outbound call to a mobile phone
or analog telephone (let's call it the CALLEE) via FXO/POTS. The
CALLER can hear the CALLEE's voice but the CALLEE cannot hear the
CALLER's voice. I used this command "ztmonitor 4 -vv -f
/tmp/test.raw"
to monitor the RX/TX but the TX is totally zero. Below is a sample
output of the ztmonitor command:
- - - < s n i p > - - -
# ztmonitor 4 -vv -f /tmp/test.raw
Output to /tmp/test.raw
Run e.g., 'sox -r 8000 -s -w -c 1 /tmp/test.raw /tmp/test.raw.wav' to
convert.
Visual Audio Levels.
--------------------
Use zapata.conf file to adjust the gains if needed.
( # = Audio Level * = Max Audio Hit )
<----------------(RX <----------------(TX
###############*
Rx: 718 ( 718) Tx: 0 ( 0)
- - - < s n i p > - - -
Anyone can help me here?
Thank you in advance.
Regards,
GNUbie
On Sun, Oct 12, 2008 at 11:53:18PM +0800, GNUbie wrote:> Hello all, > > I've been lobbying for some time at the #asterisk IRC channel. Until > now, I still can't find a solution to my one way audio problem. I > rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my > Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS > (channel 1). My SIP extension phone located inside the LAN is a SNOM > 300 IP phone. > > This one way audio problem only happens when the SIP extension phone > (let's call it the CALLER) places an outbound call to a mobile phone > or analog telephone (let's call it the CALLEE) via FXO/POTS. The > CALLER can hear the CALLEE's voice but the CALLEE cannot hear the > CALLER's voice. I used this command "ztmonitor 4 -vv -f /tmp/test.raw" > to monitor the RX/TX but the TX is totally zero. Below is a sample > output of the ztmonitor command: > > - - - < s n i p > - - - > > # ztmonitor 4 -vv -f /tmp/test.raw > Output to /tmp/test.raw > Run e.g., 'sox -r 8000 -s -w -c 1 /tmp/test.raw /tmp/test.raw.wav' to convert. > > Visual Audio Levels. > -------------------- > Use zapata.conf file to adjust the gains if needed. > > ( # = Audio Level * = Max Audio Hit ) > <----------------(RX <----------------(TX > ###############* > Rx: 718 ( 718) Tx: 0 ( 0)This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? Can you call from the FXO to Asterisk? (e.g.: to echo test) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
On Sun, 12 Oct 2008, GNUbie wrote:> Hello all, > > I've been lobbying for some time at the #asterisk IRC channel. Until > now, I still can't find a solution to my one way audio problem. I > rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my > Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS > (channel 1). My SIP extension phone located inside the LAN is a SNOM > 300 IP phone.You mention the SIP phone being inside the LAN. Where is the Asterisk box? IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. Gordon
On Oct 13, 2008, at 9:29 AM, asterisk-users-request at lists.digium.com wrote:> IME: One-way audio problems are almost always casued by NAT gateways > and/or incorrect NAT settings in sip.conf and/or incorrect IP > address or > extenal proxy settings in the SIP phone.And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com
Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie <gnubie at gmail.com> wrote:> Hello Karsten, > > On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <kwem at gmx.de> wrote: >> >> Please post Your sip.conf. >> Which IP-Address do You configure in the snom for Your asterisk? (eth0 >> or eth1)? > > The SNOM 300 is using the NET interface beside the DC 5V port to > connect to the LAN. > > The Asterisk box is using the eth1 to connect to the LAN. > > As per your instruction, below is my /etc/asterisk/sip.conf : > > - - - < s n i p > - - - > > [general] > realm=pbx.domain.com > bindport=5060 > bindaddr=0.0.0.0 > rtptimeout=60 > disallow=all > allow=ulaw > allow=alaw > allow=gsm > externip=pbx.domain.com > localnet=192.168.101.0/255.255.255.0 > jbforce=yes > allowtransfers=yes > maxexpiry=3600 > minexpiry=1800 > videosupport=no > > [internal-phones](!) > type=friend > host=dynamic > context=family > dtmfmode=rfc2833 > insecure=port,invite > canreinvite=no > nat=no > qualify=yes > port=5060 > > [102](internal-phones) > username=102 > secret=102 > callerid="GNUbie"<102> > mailbox=102 at family > > - - - < s n i p > - - - > > Thank you in advance. > > Regards, > > GNUbie > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie:> Hello Karsten, > > On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <kwem at gmx.de> wrote: > > > > Please post Your sip.conf. > > Which IP-Address do You configure in the snom for Your asterisk? (eth0 > > or eth1)? > > The SNOM 300 is using the NET interface beside the DC 5V port to > connect to the LAN. > > The Asterisk box is using the eth1 to connect to the LAN. > > As per your instruction, below is my /etc/asterisk/sip.conf : > > - - - < s n i p > - - - > > [general] > realm=pbx.domain.com > bindport=5060 > bindaddr=0.0.0.0 > rtptimeout=60 > disallow=all > allow=ulaw > allow=alaw > allow=gsm > externip=pbx.domain.com > localnet=192.168.101.0/255.255.255.0 > jbforce=yes > allowtransfers=yes > maxexpiry=3600 > minexpiry=1800 > videosupport=no > > [internal-phones](!) > type=friend > host=dynamic > context=family > dtmfmode=rfc2833 > insecure=port,invite > canreinvite=no > nat=no > qualify=yes > port=5060 > > [102](internal-phones) > username=102 > secret=102 > callerid="GNUbie"<102> > mailbox=102 at family > > - - - < s n i p > - - -Thanks for the information. In an earlier post You told us, that the local phones talk to asterisk on eth1 using 192.168.101.0 network. Could You please double check, that the phone did not try to register on another IP? The asterisk is IIRC on a dual homed machine. Is Your phone using a DNS lookup to find the asterisk? To which address is that lookup resolved? Another hint: Is Your SNOM using some sort of STUN to lookup an public address? (Just to eliminate some things). HTH, Karsten