Hello,
I have just set up an asterisk box (Debian unstable) and I would like
to test it with a H.323 application (gnomemeeting). When I call the
demo voice menu, I can't hear any sound. asterisk says that the
soundfile is played:
-- Executing BackGround("H323/ip$212.9.189.172:30005/29597",
"demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
Using strace I found out that the file is actually read, but there is
no subsequent traffic on the network. Neither a firewall nor NAT is
involved. gnomemeeting says in its history that the codec gets
immediately closed after the connection is established:
12:05:56 Opened codec G.711-ALaw-64k{sw} for transmission
12:05:56 Connected with root using The NuFone Network's H.323 Channel
Driver for Asterisk 1.0.0
12:05:56 Closed codec G.711-ALaw-64k{sw} which was opened for transmission
I edited h323.conf to use different codecs but all produced the same
result. Strangely, I can send dtmf tones and they are interpreted by
asterisk.
Do you have an idea on what's going on here and how to fix it?
Thanks a lot,
Hendrik
Hi, I hope this isn't a double-post...but here goes. I have setup an * box using WBEL, and I have * up and running. The problem I have is that when I dial an extension I cannot hear anything. It's not my sound card either. I can see the call going through on the CLI and I see where it goes to voicemail. I thought at first it was a firewall issue so I disalbed the firewall but it still is not working. Thanks for any help on this... Mike
Hi Mike -> I hope this isn't a double-post...but here goes. I have setup an * box > using WBEL, and I have * up and running. The problem I have is that > when I dial an extension I cannot hear anything. It's not my sound > card either. I can see the call going through on the CLI and I see > where it goes to voicemail. I thought at first it was a firewall issue > so I disalbed the firewall but it still is not working. Thanks for any > help on this...We'll need a few more details - 1) what phone devices are you using? Analog, SIP, IAX? 2) If SIP or IAX, the corresponding configuration files - sip.conf, iax.conf, if Analog phones - zapata.conf and zaptel.conf for (or the configuration files for your hardware, if non-zaptel) 3) It wouldn't hurt to also provide extensions.conf, though it sounds like your calls are getting directed to the right place. Thanks! Noah
I am using Firefly for the softphone (IAX option). IAX.CONF bindport=4569 bindaddr=0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=yes mailboxdetail=yes #include iax_additional.conf Hope this helps. Mike ----- Original Message ----- From: Noah Miller <noah@rosecompanies.com> Date: Friday, January 21, 2005 12:23 pm Subject: [Asterisk-Users] Re: No sound> Hi Mike - > > > I hope this isn't a double-post...but here goes. I have setup an > * box > > using WBEL, and I have * up and running. The problem I have is > that > > when I dial an extension I cannot hear anything. It's not my > sound > > card either. I can see the call going through on the CLI and I > see > > where it goes to voicemail. I thought at first it was a firewall > issue > > so I disalbed the firewall but it still is not working. Thanks > for any > > help on this... > > We'll need a few more details - > > 1) what phone devices are you using? Analog, SIP, IAX? > 2) If SIP or IAX, the corresponding configuration files - > sip.conf, > iax.conf, if Analog phones - zapata.conf and zaptel.conf for (or > the > configuration files for your hardware, if non-zaptel) > 3) It wouldn't hurt to also provide extensions.conf, though it > sounds > like your calls are getting directed to the right place. > > Thanks! > Noah > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hi Mike -> I am using Firefly for the softphone (IAX option). > > IAX.CONF > > bindport=4569 > bindaddr=0.0.0.0 > delayreject=yes > disallow=all > allow=ulaw > allow=alaw > allow=gsm > jitterbuffer=yes > mailboxdetail=yes > > #include iax_additional.confWell, I don't see any definitions for devices. I'm assuming that they're in iax_additional.conf? Can we see that? I've never used firefly personally, but here's the config suggestion from the WIKI: [firefly1] type=friend accountcode=iaxy host=dynamic secret=xxxx context=home disallow=all allow=ilbc allow=gsm auth=md5 trunk=no qualify=no I probably wouldn't use ilbc, but the person posting on the WIKI had success with it, I guess. Thanks, Noah
Hi again, Should the include # iax_additional.conf have the # in front of it? Here's the include... [201] username=201 type=friend secret=password qualify=no notransfer=yes mailbox=201 host=dynamic context=from-internal callerid="Mike"<201> That's it... Mike ----- Original Message ----- From: Noah Miller <noah@rosecompanies.com> Date: Friday, January 21, 2005 3:42 pm Subject: [Asterisk-Users] Re: No sound> Hi Mike - > > > I am using Firefly for the softphone (IAX option). > > > > IAX.CONF > > > > bindport=4569 > > bindaddr=0.0.0.0 > > delayreject=yes > > disallow=all > > allow=ulaw > > allow=alaw > > allow=gsm > > jitterbuffer=yes > > mailboxdetail=yes > > > > #include iax_additional.conf > > Well, I don't see any definitions for devices. I'm assuming that > they're in iax_additional.conf? Can we see that? I've never used > firefly personally, but here's the config suggestion from the WIKI: > > [firefly1] > type=friend > accountcode=iaxy > host=dynamic > secret=xxxx > context=home > disallow=all > allow=ilbc > allow=gsm > auth=md5 > trunk=no > qualify=no > > I probably wouldn't use ilbc, but the person posting on the WIKI > had > success with it, I guess. > > Thanks, > Noah > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I have an asterisk box installed, but all connections to outside of the private network do not have a sound. Can you give me a hint what it could it be? bye Ronald
Hi, That would probably be a problem with nat. Just read this on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+NAT+solutions Best regards, Stevanus Ronald Wiplinger wrote:> I have an asterisk box installed, but all connections to outside of > the private network do not have a sound. > > Can you give me a hint what it could it be? > > > bye > > Ronald > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
So you mean your SIP server is on the outside network and your peers on the private one? If so, sound behind NAT is always an issue, because it goes through a different port than 5060 udp (which is the reg. port). I had the same problem a few days ago. Posted the question to the list twice, and found help in the #asterisk channel at irc.freenode.net. Probably they can help you as well. But give www.voip-info.org a shot. Search fot NAT related issues. On 7/7/05, Ronald Wiplinger <ronald@elmit.com> wrote:> I have an asterisk box installed, but all connections to outside of the > private network do not have a sound. > > Can you give me a hint what it could it be? > > > bye > > Ronald > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Julio C. Ody http://rootshell.be/~julioody
Julio Cesar Ody wrote: Thanks all of you. I forgot to set the stun server, .... bye Ronald>So you mean your SIP server is on the outside network and your peers >on the private one? > >If so, sound behind NAT is always an issue, because it goes through a >different port than 5060 udp (which is the reg. port). I had the same >problem a few days ago. Posted the question to the list twice, and >found help in the #asterisk channel at irc.freenode.net. > >Probably they can help you as well. But give www.voip-info.org a shot. >Search fot NAT related issues. > > >On 7/7/05, Ronald Wiplinger <ronald@elmit.com> wrote: > > >>I have an asterisk box installed, but all connections to outside of the >>private network do not have a sound. >> >>Can you give me a hint what it could it be? >> >> >>bye >> >>Ronald >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > > >