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Yealink "Hold" function not working

December 13, 2017, 5:32 am
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Freepbx 13.0.129 Asterisk 13.18

Yealink T41S phones. They are using Endpoint Manager to provision the phones. Using the Hold button on the phone an outside call gets put on Hold and they hear music on hold. When they press the resume button on the phone they have waited up to 70 seconds for the audio to return. If they use the hold butoon again the call is resumed immediately. We found thread that told of changing Rpid in the extension settings, it was set to Yes and using the Hold button dropped the call Changing the Rpid to No allowed the call to be put on Hold, and that is where we are now. I was always under the impression that the “Hold” function on a phone was internal and really didn’t interact with Asterisk for that function. Commenta as to what is happening so I can put this to bed?

Thanks

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Asterisk handles incoming in error

December 13, 2017, 5:34 am
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I agree with Daniel, you appear to have mixed up your chan_SIP/PJSIP ports. If you are using chan_sip trunks, make sure you are using to the chan_sip bind port.

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Yealink "Hold" function not working

December 13, 2017, 5:37 am
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@edlentz are you saying that disabling RPID for the extension fixes your audio issue?

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Queue and Agent Reporting

December 13, 2017, 5:40 am
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For my curiousity, why no to FOP and Queue Metrics?

Have you seen Asternic Call Center Stats? Or do you count that as FOP2?

I’m not sure it would meet all of your needs, but they have a demo site from thier main page, so you can see what you would get before buying. They also have a lite version.

https://www.asternic.net

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FreePBX disabling modules for pjsip

December 13, 2017, 5:42 am
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ok. I see it now.
it was set to "only sip"
but even with that setting , I was able to use pjsip endpoints if I load modules.
is this setting for module enable/disable only ?

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FreePBX disabling modules for pjsip

December 13, 2017, 5:47 am
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Correct. The proper place to enable/disable drivers is in advanced settings.

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Xlite config converted to FreePBX

December 13, 2017, 5:50 am
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Not sure I understand exactly what you’re asking for but, I’m assuming you’re asking how to configure FreePBX so that your posted Xlite config will work without changing any settings on the soft phone?

If so…
Create a new extension:
Extension Number: 043058370
Password: Same password as current Xlite config

Then…
PBX URL: Create a DNS A record for sip.freedom1.com that points to the public IP of your PBX (assuming you own the domain name freedom1.com)

Your Xlite config for your proxy address is probably going to have to change since 87.52.109.38 belongs to the current platform that you connect to I would imagine. You can just replace that with the FQDN of your PBX that you created as I mentioned above.

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Yealink "Hold" function not working

December 13, 2017, 5:52 am
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Yes, It would go on hold and when they tried to Resume the call it took up to 70 seconds to get audio back. This is strange

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FreePBX disabling modules for pjsip

December 13, 2017, 5:54 am
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Extension always busy

December 13, 2017, 6:09 am
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This topic was automatically closed 24 hours after the last reply. New replies are no longer allowed.

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Yealink "Hold" function not working

December 13, 2017, 6:19 am
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I’ve seen this recently. It was a Yealink T4x phone running an older firmware, whenever the PBX sent the INVITE with the RPID info, the phone audio would break, same thing happens when you park / retrieve a call from the parking lot. A non-Yealink device using the same extension does not have this issue.

As far as I am concerned, this is a firmware bug, which may or may not be solved by upgrading/downgrading the phone. Sending a reINVITE to a phone shouldn’t break audio, even if the INVITE is not formatted to the specific phone requirements.

There is interaction, otherwise Asterisk would not know when to play hold music to the waiting caller.

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Excluding Call Recordings Overide Location using Warm Spare Method

December 13, 2017, 6:21 am
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ahhh this was a dumb question… I just created a shell script with those command and called it in the post restore field in the backup module. Works great, thank you Lorne!

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Xlite config converted to FreePBX

December 13, 2017, 6:34 am
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Apologies for not being clear.

The Xlite config are the details from my VSP. I want to move off Xlite and move to a hard phone via FreePBX.

So looking to find out what my trunk configuration and registration string should be?

Is there enough information above to move to FreePBX or will I need to contact my VSP?

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TDM410P - Call waiting not working

December 13, 2017, 6:39 am
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James

Thanks for the reply. I will be at the remote location this weekend and I will be able to test then.

Sorry for the questions. What logs are you exactly looking for?

Renato

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Annoucement redirection

December 13, 2017, 6:41 am
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Well it’s not really up to me to change that.
The company is not ready for this changes. We didn’t plan for it, and it’s quiet busy here.

*no alternative then, on asterisk to configure that ?

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S705 Font Wont change

December 13, 2017, 6:42 am
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https not over wifi… will grab every other setting just stubborn on changing the font…

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Xlite config converted to FreePBX

December 13, 2017, 6:48 am
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Trunk registration is completely dependent on your provider that you choose.

Your Xlite settings are just connecting to another VPS server somewhere that is running PBX software that has a trunk configuration built out to a provider.

You need to start with a fresh install of FreePBX Distro and create your trunks from there, but as I said, trunk configuration is dependent on the provider you choose.

Some authenticate with username and passwords, some authenticate by IP, etc.

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Change password (CLI)

December 13, 2017, 6:50 am
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[root@localhost ~]# fwconsole a r

[InvalidArgumentException]
Command “a” is not defined.
Did you mean one of these?
start
certificates
firewall
userman
restart
reload
ma
moduleadmin
notification
externalip

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FreePBX 14.0.1.20 SIP Trunk

December 13, 2017, 6:56 am
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I have a brand new FreePBX Server that I am trying to configure. My previous FreePBX server is having issues so I am manually configuring the new server with the same settings on the old server.

I have noticed that on this version of Asterisk 13.17.2 chan_sip has been replaced with pjsip and is the biggest struggle for me at the present time in regards to my SIP Trunks.

Currently I have created a chan_sip trunk with the settings from the old server. I am able to receive inbound calls on all of the numbers of this trunk. On dialing out to any number I am getting All Circuits are busy

Here is a copy of the output when dialing out

  • <PJSIP/108-00000002>AGI Script sangomacrm.agi completed, returning 0
    – Executing [s@macro-dialout-trunk:24] Set(“PJSIP/108-00000002”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
    – Executing [s@macro-dialout-trunk:25] NoOp(“PJSIP/108-00000002”, “CRM Finished”) in new stack
    – Executing [s@macro-dialout-trunk:26] GotoIf(“PJSIP/108-00000002”, “0?bypass,1”) in new stack
    – Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/108-00000002”, “1?Set(CONNECTEDLINE(num,i)=574xxxxxxx”) in new stack
    – Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/108-00000002”, “1?Set(CONNECTEDLINE(name,i)=CID:5742540111)”) in new stack
    – Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/108-00000002”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)5742540111)”) in new stack
    – Executing [s@macro-dialout-trunk:30] GotoIf(“PJSIP/108-00000002”, “0?customtrunk”) in new stack
    – Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/108-00000002”, “SIP/SoTelRegistrationOutgoing/574261xxxx,300,T”) in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    [2017-12-13 14:47:18] ERROR[19176][C-00000001]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“SoTelRegistrationOutgoing”, “(null)”, …): Name or service not known
    [2017-12-13 14:47:18] WARNING[19176][C-00000001]: chan_sip.c:6320 create_addr: No such host: SoTelRegistrationOutgoing
    [2017-12-13 14:47:18] WARNING[19176][C-00000001]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
    == Everyone is busy/congested at this time (1:0/0/1)
    – Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/108-00000002”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20”) in new stack
    – Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/108-00000002”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
    – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    – Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/108-00000002”, “RC=20”) in new stack
    – Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“PJSIP/108-00000002”, “20,1”) in new stack
    – Goto (macro-dialout-trunk,20,1)
    – Executing [20@macro-dialout-trunk:1] Goto(“PJSIP/108-00000002”, “continue,1”) in new stack
    – Goto (macro-dialout-trunk,continue,1)
    – Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/108-00000002”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks”) in new stack
    – Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/108-00000002”, “1?Set(CALLERID(number)=108)”) in new stack
    – Executing [5742614839@from-internal:8] Macro(“PJSIP/108-00000002”, “outisbusy,”) in new stack
    – Executing [s@macro-outisbusy:1] Progress(“PJSIP/108-00000002”, “”) in new stack
    – Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/108-00000002”, “0?emergency,1”) in new stack
    – Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/108-00000002”, “0?intracompany,1”) in new stack
    – Executing [s@macro-outisbusy:4] Playback(“PJSIP/108-00000002”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
    – <PJSIP/108-00000002> Playing ‘all-circuits-busy-now.g722’ (language ‘en’)
    > 0x7f9b58019680 – Strict RTP learning after remote address set to: 172.16.253.4:16500
    > 0x7f9b58019680 – Strict RTP switching to RTP target address 172.16.253.4:16500 as source

Things I do not understand. Within the chan_sip settings of the trunk it asks for Trunk Name on Outgoing. I use the same name from the old server and it is trying to do a DNS Lookup of this name which is not valid. Since the provider is using SRV records, I have tried to replace the trunk name on outgoing with the host entry of the sip settings and get the same result with a different message

– Executing [s@macro-dialout-trunk:24] Set(“PJSIP/108-00000005”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:25] NoOp(“PJSIP/108-00000005”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:26] GotoIf(“PJSIP/108-00000005”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/108-00000005”, “1?Set(CONNECTEDLINE(num,i)=5742614839)”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/108-00000005”, “1?Set(CONNECTEDLINE(name,i)=CID:5742540111)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/108-00000005”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)5742540111)”) in new stack
– Executing [s@macro-dialout-trunk:30] GotoIf(“PJSIP/108-00000005”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/108-00000005”, “SIP/voip.sotelsystems.com/574261xxxx,300,T”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/voip.sotelsystems.com/5742614839
[2017-12-13 14:51:59] NOTICE[18946][C-00000003]: chan_sip.c:23996 handle_response_invite: Failed to authenticate on INVITE to ‘sip:5742540111@165.138.207.10:5160;tag=as4d93509b’
– SIP/voip.sotelsystems.com-00000002 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

The External IP Address of the server is the 165.138.207.10

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FreePBX disabling modules for pjsip

December 13, 2017, 7:09 am
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And FYI, many conf files can’t be edited directly, including modules.conf. Note the warning at the top:

[root@lgaetzdev2 ~]# head /etc/asterisk/modules.conf
;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
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