hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints: Peer User/ANR Call ID Format Hold Last Message Expiry Peer peer.ip 1001 1...-5060 (ulaw) No Rx: ACK 1001 But I hear nothing at the peer's end. When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mobile context = home type = friend callerid = Five <1005> secret = XYZ host = dynamic mailbox = 1005 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) same = n,Playback(beep) same = n,Wait(1) same = n,Playback(im-sorry) same = n,Wait(1) same = n,Playback(number-not-answering) same = n,Wait(1) same = n,Hangup() exten => 1001,1,Dial(SIP/1001) ; grandstream phone exten => 1005,1,Dial(SIP/1005) ; mobile -- oooooooooo.io bibliotecha.info
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:> extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1)If this is copy and paste, then your dialplan is broken (= should be =>) But to debug, enable logging (core set verbose 5), when needed debugging (core set debug 5) and sip logging (sip set debug on / pjsip set logger on).
On Tuesday 06 June 2017 15:18:32 andre castro wrote:> I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server.> But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine.Tell us about your networking arrangement - are both phones and the Asterisk machine on the same network? Is there a router in between any of them? Is there any NAT involved?> Do I need to have alsa installed??No. Antony. -- Perfection in design is achieved not when there is nothing left to add, but rather when there is nothing left to take away. - Antoine de Saint-Exupery Please reply to the list; please *don't* CC me.
Le 06/06/2017 ? 16:25, Daniel Tryba a ?crit :> On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: >> extensions.conf: >> [home] >> exten = 102,1,Answer() >> same = n,Wait(1) > > If this is copy and paste, then your dialplan is broken (= should be =>)Well, no. = or => are the same. -- Daniel
Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote:> On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I just installed asterisk in a debian server. >> All seems to be running fine, but the audio sent by the server. > >> But I hear nothing at the peer's end. >> >> When one peer calls another, sound comes through just fine. > > Tell us about your networking arrangement - are both phones and the Asterisk > machine on the same network?Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another.> > Is there a router in between any of them?Yes. In the phones network.> > Is there any NAT involved?Yes in the phones' network. They both have different private IP address and one public IP.> >> Do I need to have alsa installed?? > > No.So I thought. Thanks guys!!> > > Antony. >-- oooooooooo.io bibliotecha.info
Try to use the echo app. If you can listen your echo, so it is something in the network. Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 14:18, andre castro <andre at andrecastro.info> wrote:> hello folks, > this might be a simple question... > > I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server. > If I have one of my registered peers call and extension (102) that plays > back audio (extension.conf and sip.conf coffee-pasted below), Asterisk > answers and prints no errors. > Its `sip show channels` prints: > > Peer User/ANR Call ID Format Hold Last Message Expiry > Peer > peer.ip 1001 1...-5060 (ulaw) No Rx: ACK > 1001 > > But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine. > So my hunch is that is something to do with the audio supplied by the > server. > Do I need to have alsa installed?? > Any hint? > > sip.conf: > > [general] > context = unauthenticated > bindport = 5060 > bindaddr = 0.0.0.0 > tcpbindaddr = 0.0.0.0 > tcpenable = yes > videosupport = no > textsupport=yes > alwaysauthreject=yes > allowguest=no > > [1001] ; grandstream 1 > context = home > type = friend > callerid = One <1001> > secret = XYZ > host = dynamic > mailbox = 1001 > disallow = all > allow = ulaw > transport = udp > dtmfmode=auto ; accept touch-tones from the devices, negotiated > automatically > nat=force_rport > > [1005] ; mobile > context = home > type = friend > callerid = Five <1005> > secret = XYZ > host = dynamic > mailbox = 1005 > disallow = all > allow = ulaw > transport = udp > dtmfmode=auto ; accept touch-tones from the devices, negotiated > automatically > nat=force_rport > > > extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1) > same = n,Playback(beep) > same = n,Wait(1) > same = n,Playback(im-sorry) > same = n,Wait(1) > same = n,Playback(number-not-answering) > same = n,Wait(1) > same = n,Hangup() > > exten => 1001,1,Dial(SIP/1001) ; grandstream phone > exten => 1005,1,Dial(SIP/1005) ; mobile > > > > > -- > oooooooooo.io > bibliotecha.info > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users