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</o:shapelayout></xml><![endif]--></head><body lang=SV link="#0563C1" vlink="#954F72" style='word-wrap:break-word'><div class=WordSection1><p class=MsoNormal><span lang=EN-US>Also a addition:<br><br>If I call from the 8961 TO the ACR phone client – audio works fine both ways.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Its only when I receive a call from the DID where audio becomes one-way on the ACR Phone client but nowhere else.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>And notice that no asterisk sip.conf settings were changed before switching from native Android to ACR Phone.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span style='mso-fareast-language:SV'>Från:</span></b><span style='mso-fareast-language:SV'> Sebastian Nielsen <sebastian@sebbe.eu> <br><b>Skickat:</b> den 23 november 2021 21:38<br><b>Till:</b> asterisk-users@lists.digium.com; 'NLL APPS' <m@nllapps.com><br><b>Ämne:</b> Problems with one-way audio in ACR Phone<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US>Here is my problems (sent a copy to the developer of ACR Phone aswell):<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I have a mobile phone with ACR Phone client, and a fixed Cisco 8961 connected to a PBX with DID.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Incoming calls answered in the 8961 – audio works both ways fine<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Outgoing calls called from the 8961 – audio works both ways fine<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Incoming calls answered in the ACR Phone client – I can only hear the calling person, the calling person cannot hear me.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Outgoing calls called from the ACR Phone client – audio works both ways fine<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Im using a VPN tunnel to the PBX server, so there no problems firewall-wise.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>The SIP on the phone worked perfectly when I ran Android’s native SIP, but since native SIP support was dropped in android 12, I switched to ACR phone as SIP client, and since then the audio problems on incoming calls started.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Here is a incoming call where audio doesn’t work (calling party doesn’t hear me). Calling party CAN hear the IVR however, so I didn’t include the IVR, but only the parts from when sip09 (ACR phone) rings.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><-------------><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>--- (9 headers 0 lines) ---<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>sip_route_dump: no route/path<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- SIP/sip09-00000015 is ringing<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><--- SIP read from TCP:192.168.2.2:58984 ---><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP/2.0 200 Ok<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Via: SIP/2.0/TCP 192.168.1.10:5060;branch=z9hG4bK343a024e;rport<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>From: "()" <<a href="sip:asterisk@192.168.1.10">sip:asterisk@192.168.1.10</a>>;tag=as10b547cd<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>To: <<a href="sip:sip09@192.168.2.2:58984;transport=tcp">sip:sip09@192.168.2.2:58984;transport=tcp</a>>;tag=z4lyL7B<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Call-ID: <a href="mailto:3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060">3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>CSeq: 101 INVITE<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>User-Agent: ACR Phone/0.102-playStore-WithAccessibility-arm8/12 (SM-G998B)/5.2.0-alpha.5+ff5a92f (master)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Supported: replaces, outbound, gruu<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Contact: <<a href="sip:192.168.2.2:58984;transport=tcp">sip:192.168.2.2:58984;transport=tcp</a>>;+sip.instance="<urn:uuid:b5cdf596-3ab2-00f7-b3dc-c7bc3b050365>"<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Content-Type: application/sdp<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Content-Length: 147<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>v=0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>o=linphone 808 1413 IN IP4 192.168.2.2<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>s=Talk<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>c=IN IP4 192.168.2.2<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>t=0 0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>m=audio 7078 RTP/AVP 8 0 101<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>a=rtpmap:101 telephone-event/8000<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><-------------><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>--- (12 headers 7 lines) ---<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Got SDP version 1413 and unique parts [linphone 808 IN IP4 192.168.2.2]<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Found RTP audio format 8<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Found RTP audio format 0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Found RTP audio format 101<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Found audio description format telephone-event for ID 101<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> > 0x7f12e4055e00 -- Strict RTP learning after remote address set to: 192.168.2.2:7078<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Peer audio RTP is at port 192.168.2.2:7078<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>sip_route_dump: route/path hop: <<a href="sip:192.168.2.2:58984;transport=tcp">sip:192.168.2.2:58984;transport=tcp</a>><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Transmitting (NAT) to 192.168.2.2:58984:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>ACK <a href="sip:192.168.2.2:58984;transport=tcp">sip:192.168.2.2:58984;transport=tcp</a> SIP/2.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Via: SIP/2.0/TCP 192.168.1.10:5060;branch=z9hG4bK5c5f7293;rport<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Max-Forwards: 70<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>From: "()" <<a href="sip:asterisk@192.168.1.10">sip:asterisk@192.168.1.10</a>>;tag=as10b547cd<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>To: <<a href="sip:sip09@192.168.2.2:58984;transport=tcp">sip:sip09@192.168.2.2:58984;transport=tcp</a>>;tag=z4lyL7B<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Contact: <<a href="sip:asterisk@192.168.1.10:5060;transport=tcp">sip:asterisk@192.168.1.10:5060;transport=tcp</a>><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Call-ID: <a href="mailto:3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060">3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>CSeq: 101 ACK<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>User-Agent: Asterisk PBX 16.9.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Content-Length: 0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>---<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- SIP/sip09-00000015 answered Local/inq@agents-00000001;2<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Stopped music on hold on Local/inq@agents-00000001;2<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel SIP/sip09-00000015 joined 'simple_bridge' basic-bridge <9d34252d-6f7a-4827-b048-1483298c5c17><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel Local/inq@agents-00000001;2 joined 'simple_bridge' basic-bridge <9d34252d-6f7a-4827-b048-1483298c5c17><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> > Move-swap optimizing Local/inq@agents-00000001;1 <-- SIP/sip09-00000015.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel SIP/sip09-00000015 left 'simple_bridge' basic-bridge <9d34252d-6f7a-4827-b048-1483298c5c17><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel Local/inq@agents-00000001;1 left 'simple_bridge' basic-bridge <0b23fdff-4594-42b1-8fd8-5c85677f4204><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel SIP/sip09-00000015 swapped with Local/inq@agents-00000001;1 into 'simple_bridge' basic-bridge <0b23fdff-4594-42b1-8fd8-5c85677f4204><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel Local/inq@agents-00000001;2 left 'simple_bridge' basic-bridge <9d34252d-6f7a-4827-b048-1483298c5c17><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> == Spawn extension (agents, inq, 2) exited non-zero on 'Local/inq@agents-00000001;2'<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Executing [h@agents:1] Set("Local/inq@agents-00000001;2", "SHARED(DIALSTATUS,SIP/cellip-0000000d)=ANSWER") in new stack<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> > 0x7f12e4055e00 -- Strict RTP switching to RTP target address 192.168.2.2:7078 as source<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> > 0x7f12b400e140 -- Strict RTP learning complete - Locking on source address 193.105.226.102:57816<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><--- SIP read from TCP:192.168.2.2:58984 ---><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>BYE <a href="sip:asterisk@192.168.1.10:5060;transport=tcp">sip:asterisk@192.168.1.10:5060;transport=tcp</a> SIP/2.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Via: SIP/2.0/TCP 192.168.2.2:58984;branch=z9hG4bK.q73Huv9fk;rport<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>From: <<a href="sip:sip09@192.168.2.2">sip:sip09@192.168.2.2</a>>;tag=z4lyL7B<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>To: "()" <<a href="sip:asterisk@192.168.1.10">sip:asterisk@192.168.1.10</a>>;tag=as10b547cd<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>CSeq: 111 BYE<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Call-ID: <a href="mailto:3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060">3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Max-Forwards: 70<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>User-Agent: ACR Phone/0.102-playStore-WithAccessibility-arm8/12 (SM-G998B)/5.2.0-alpha.5+ff5a92f (master)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Content-Length: 0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><-------------><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>--- (9 headers 0 lines) ---<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Sending to 192.168.2.2:58984 (NAT)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Scheduling destruction of SIP dialog '3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060' in 32000 ms (Method: BYE)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><--- Transmitting (NAT) to 192.168.2.2:58984 ---><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP/2.0 200 OK<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Via: SIP/2.0/TCP 192.168.2.2:58984;branch=z9hG4bK.q73Huv9fk;received=192.168.2.2;rport=58984<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>From: <<a href="sip:sip09@192.168.2.2">sip:sip09@192.168.2.2</a>>;tag=z4lyL7B<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>To: "()" <<a href="sip:asterisk@192.168.1.10">sip:asterisk@192.168.1.10</a>>;tag=as10b547cd<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Call-ID: <a href="mailto:3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060">3f53be0b41bf957d6359b9d40e92f843@192.168.1.10:5060</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>CSeq: 111 BYE<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Server: Asterisk PBX 16.9.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Supported: replaces,timer<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Content-Length: 0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><------------><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel SIP/sip09-00000015 left 'simple_bridge' basic-bridge <0b23fdff-4594-42b1-8fd8-5c85677f4204><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Channel SIP/cellip-0000000d left 'simple_bridge' basic-bridge <0b23fdff-4594-42b1-8fd8-5c85677f4204><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> == Spawn extension (authok, s, 11) exited non-zero on 'SIP/cellip-0000000d'<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> -- Executing [h@authok:1] ExecIf("SIP/cellip-0000000d", "0?Set(FILE(/var/secure_files/missedcalls.txt,,,al,u)=20211123212225,,,)") in new stack<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Huh? Child handler, but nobody there?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> == MixMonitor close filestream (mixed)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> == End MixMonitor Recording SIP/cellip-0000000d<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Any ideas what could be wrong?<br><br><br>Note that I have NAT on for the client in question, because the client in question changes IP. For example, when in wifi reachability, it will not connect through VPN but directly via wifi (on the same network as PBX).<o:p></o:p></span></p></div></body></html>