[asterisk-users] Problems with one-way audio in ACR Phone

Tue Nov 23 14:37:45 CST 2021

Here is my problems (sent a copy to the developer of ACR Phone aswell):

I have a mobile phone with ACR Phone client, and a fixed Cisco 8961
connected to a PBX with DID.

Incoming calls answered in the 8961 - audio works both ways fine

Outgoing calls called from the 8961 - audio works both ways fine

Incoming calls answered in the ACR Phone client - I can only hear the
calling person, the calling person cannot hear me.

Outgoing calls called from the ACR Phone client - audio works both ways fine

Im using a VPN tunnel to the PBX server, so there no problems firewall-wise.

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.

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.

<------------->

--- (9 headers 0 lines) ---

sip_route_dump: no route/path

    -- SIP/sip09-00000015 is ringing

<--- SIP read from TCP:192.168.2.2:58984 --->

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.10:5060;branch=z9hG4bK343a024e;rport

From: "()" <sip:asterisk at 192.168.1.10>;tag=as10b547cd

To: <sip:sip09 at 192.168.2.2:58984;transport=tcp>;tag=z4lyL7B

Call-ID: 3f53be0b41bf957d6359b9d40e92f843 at 192.168.1.10:5060

CSeq: 101 INVITE

User-Agent: ACR Phone/0.102-playStore-WithAccessibility-arm8/12
(SM-G998B)/5.2.0-alpha.5+ff5a92f (master)

Supported: replaces, outbound, gruu

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO, PRACK, UPDATE

Contact:
<sip:192.168.2.2:58984;transport=tcp>;+sip.instance="<urn:uuid:b5cdf596-3ab2
-00f7-b3dc-c7bc3b050365>"

Content-Type: application/sdp

Content-Length: 147

v=0

o=linphone 808 1413 IN IP4 192.168.2.2

s=Talk

c=IN IP4 192.168.2.2

t=0 0

m=audio 7078 RTP/AVP 8 0 101

a=rtpmap:101 telephone-event/8000

<------------->

--- (12 headers 7 lines) ---

Got SDP version 1413 and unique parts [linphone 808 IN IP4 192.168.2.2]

Found RTP audio format 8

Found RTP audio format 0

Found RTP audio format 101

Found audio description format telephone-event for ID 101

Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)

       > 0x7f12e4055e00 -- Strict RTP learning after remote address set to:
192.168.2.2:7078

Peer audio RTP is at port 192.168.2.2:7078

sip_route_dump: route/path hop: <sip:192.168.2.2:58984;transport=tcp>

Transmitting (NAT) to 192.168.2.2:58984:

ACK sip:192.168.2.2:58984;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:5060;branch=z9hG4bK5c5f7293;rport

Max-Forwards: 70

From: "()" <sip:asterisk at 192.168.1.10>;tag=as10b547cd

To: <sip:sip09 at 192.168.2.2:58984;transport=tcp>;tag=z4lyL7B

Contact: <sip:asterisk at 192.168.1.10:5060;transport=tcp>

Call-ID: 3f53be0b41bf957d6359b9d40e92f843 at 192.168.1.10:5060

CSeq: 101 ACK

User-Agent: Asterisk PBX 16.9.0

Content-Length: 0

---

    -- SIP/sip09-00000015 answered Local/inq at agents-00000001;2

    -- Stopped music on hold on Local/inq at agents-00000001;2

    -- Channel SIP/sip09-00000015 joined 'simple_bridge' basic-bridge
<9d34252d-6f7a-4827-b048-1483298c5c17>

    -- Channel Local/inq at agents-00000001;2 joined 'simple_bridge'
basic-bridge <9d34252d-6f7a-4827-b048-1483298c5c17>

       > Move-swap optimizing Local/inq at agents-00000001;1 <--
SIP/sip09-00000015.

    -- Channel SIP/sip09-00000015 left 'simple_bridge' basic-bridge
<9d34252d-6f7a-4827-b048-1483298c5c17>

    -- Channel Local/inq at agents-00000001;1 left 'simple_bridge' basic-bridge
<0b23fdff-4594-42b1-8fd8-5c85677f4204>

    -- Channel SIP/sip09-00000015 swapped with Local/inq at agents-00000001;1
into 'simple_bridge' basic-bridge <0b23fdff-4594-42b1-8fd8-5c85677f4204>

    -- Channel Local/inq at agents-00000001;2 left 'simple_bridge' basic-bridge
<9d34252d-6f7a-4827-b048-1483298c5c17>

  == Spawn extension (agents, inq, 2) exited non-zero on
'Local/inq at agents-00000001;2'

    -- Executing [h at agents:1] Set("Local/inq at agents-00000001;2",
"SHARED(DIALSTATUS,SIP/cellip-0000000d)=ANSWER") in new stack

       > 0x7f12e4055e00 -- Strict RTP switching to RTP target address
192.168.2.2:7078 as source

       > 0x7f12b400e140 -- Strict RTP learning complete - Locking on source
address 193.105.226.102:57816

<--- SIP read from TCP:192.168.2.2:58984 --->

BYE sip:asterisk at 192.168.1.10:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.2.2:58984;branch=z9hG4bK.q73Huv9fk;rport

From: <sip:sip09 at 192.168.2.2>;tag=z4lyL7B

To: "()" <sip:asterisk at 192.168.1.10>;tag=as10b547cd

CSeq: 111 BYE

Call-ID: 3f53be0b41bf957d6359b9d40e92f843 at 192.168.1.10:5060

Max-Forwards: 70

User-Agent: ACR Phone/0.102-playStore-WithAccessibility-arm8/12
(SM-G998B)/5.2.0-alpha.5+ff5a92f (master)

Content-Length: 0

<------------->

--- (9 headers 0 lines) ---

Sending to 192.168.2.2:58984 (NAT)

Scheduling destruction of SIP dialog
'3f53be0b41bf957d6359b9d40e92f843 at 192.168.1.10:5060' in 32000 ms (Method:
BYE)

<--- Transmitting (NAT) to 192.168.2.2:58984 --->

SIP/2.0 200 OK

Via: SIP/2.0/TCP
192.168.2.2:58984;branch=z9hG4bK.q73Huv9fk;received=192.168.2.2;rport=58984

From: <sip:sip09 at 192.168.2.2>;tag=z4lyL7B

To: "()" <sip:asterisk at 192.168.1.10>;tag=as10b547cd

Call-ID: 3f53be0b41bf957d6359b9d40e92f843 at 192.168.1.10:5060

CSeq: 111 BYE

Server: Asterisk PBX 16.9.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

Supported: replaces,timer

Content-Length: 0

<------------>

    -- Channel SIP/sip09-00000015 left 'simple_bridge' basic-bridge
<0b23fdff-4594-42b1-8fd8-5c85677f4204>

    -- Channel SIP/cellip-0000000d left 'simple_bridge' basic-bridge
<0b23fdff-4594-42b1-8fd8-5c85677f4204>

  == Spawn extension (authok, s, 11) exited non-zero on
'SIP/cellip-0000000d'

    -- Executing [h at authok:1] ExecIf("SIP/cellip-0000000d",
"0?Set(FILE(/var/secure_files/missedcalls.txt,,,al,u)=20211123212225,,,)")
in new stack

Huh?  Child handler, but nobody there?

  == MixMonitor close filestream (mixed)

  == End MixMonitor Recording SIP/cellip-0000000d

Any ideas what could be wrong?

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).

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20211123/b23d7e71/attachment.html>