SPY no sound
Using the latests freepbx distro asterisk 13 and PJSIP, installed version 0.15 and the popup did not show, i changed to the devel version wich showed the popup but after setting the SPY to PJSIP/1101, i recieve a call to the extension with the name Anonymous but there is no sound on the call or the bridge.
Regards
Can you share a sip configs to simulate your environment here.
I not sure is this a issue a Qpanel but I want to research a little more.
same issue when testing spy option ring extension with the name Anonymou after answer no sound witch file do you want sip.conf ?
Can you explain more the scenario?
So, we also need the some trace of the call, tcpump, asterisk cli verbose and sip.conf
== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/5005000 -- Connected line update to Local/5005000@from-queue-0000005b;2 prevented. -- Local/5005000@from-queue-0000005b;1 is ringing -- SIP/5005000-0000008c is ringing -- Local/5005000@from-queue-0000005b;1 is ringing > 0x1d37160 -- Strict RTP learning complete - Locking on source address X.X.X.X:4000 > 0x7fad5c028a80 -- Strict RTP learning after remote address set to: X.X.X.X:59454 -- Connected line update to Local/5005000@from-queue-0000005b;2 prevented. -- SIP/5005000-0000008c answered Local/5005000@from-queue-0000005b;2 -- Executing [s@macro-auto-blkvm:1] Set("SIP/5005000-0000008c", "__MACRO_RESULT=") in new stack -- Executing [s@macro-auto-blkvm:2] Set("SIP/5005000-0000008c", "CFIGNORE=") in new stack -- Executing [s@macro-auto-blkvm:3] Set("SIP/5005000-0000008c", "MASTER_CHANNEL(CFIGNORE)=") in new stack -- Executing [s@macro-auto-blkvm:4] Set("SIP/5005000-0000008c", "FORWARD_CONTEXT=from-internal") in new stack -- Executing [s@macro-auto-blkvm:5] Set("SIP/5005000-0000008c", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack -- Executing [s@macro-auto-blkvm:6] Macro("SIP/5005000-0000008c", "blkvm-clr,") in new stack -- Executing [s@macro-blkvm-clr:1] Set("SIP/5005000-0000008c", "SHARED(BLKVM,SIP/4004000-0000008b)=") in new stack -- Executing [s@macro-blkvm-clr:2] Set("SIP/5005000-0000008c", "GOSUB_RETVAL=") in new stack -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/5005000-0000008c", "") in new stack -- Executing [s@macro-auto-blkvm:7] ExecIf("SIP/5005000-0000008c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=5005000)") in new stack -- Executing [s@macro-auto-blkvm:8] ExecIf("SIP/5005000-0000008c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Ag voipsense)") in new stack -- Local/5005000@from-queue-0000005b;1 answered SIP/4004000-0000008b -- Stopped music on hold on SIP/4004000-0000008b > 0x7fad5c028a80 -- Strict RTP switching to RTP remote address X.X.X.X:59454 as source > 0x7fad5c028a80 -- Strict RTP learning complete - Locking on source address 196.127.109.103:59454 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 > 0x7fad7c13bc20 -- Strict RTP learning after remote address set to: X.X.X.X:10000 > Channel SIP/20002000-0000008d was answered. > Launching ChanSpy(hint:5005000@ext-local,q) on SIP/20002000-0000008d > 0x7fad5c028a80 -- Strict RTP learning after remote address set to: 0.0.0.0:59454 -- Started music on hold, class 'voipsense', on Local/5005000@from-queue-0000005b;2

Call between SIP/4004000 and queue member SIP/5005000

Spy EXT:SIP/20002000
@roramirez Exactamente el mismo problema aquí con FreePBX y Qpanel. La solución es cambiar el formato del canal en la petición POST de "hint:106@ext-local" a "PJSIP/106" (por ejemplo). Entonces funciona perfecto.
channel=PJSIP%2F106 to_exten=PJSIP%2F108
¿Como se puede arreglar el problema? ¿Regexp en el código para cambiar el formato?
What version of FreePBX you have?
I'll think what is the more clean solution for that. The trouble here is we not know the channel type of this hint
I had the same problem and this solved my problem.
I just modified the code in spy method of qpanel/asterisk.py to change channel's format from hint: 1111@ext-local to SIP/1111 and now I can hear the voices too
Thanks to @joanrierapol Asterisk version: 11.25.3 QPanel code that I tested with: HEAD of master at 7c7a53d
I had the same problem and this solved my problem. I just modified the code in
spymethod ofqpanel/asterisk.pyto changechannel's format fromhint: 1111@ext-localtoSIP/1111and now I can hear the voices tooThanks to @joanrierapol Asterisk version: 11.25.3 QPanel code that I tested with: HEAD of master at 7c7a53d
Which file do I make this change?
just overwrite this method in "/qpanel/qpanel/app.py". issue will solved.
def spy(): channel = request.form['channel'] to_exten = request.form['to_exten'] channelid = 'SIP/'+''.join(filter(lambda i: i.isdigit(), channel)) r = backend.spy(channelid, to_exten) return jsonify(result=r)
Its Works perfectly thank you