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User Control Panel error after restoring FreePBX13 backup set to FreePBX15

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@ivana72 wrote:

When selecting UCP exception error is thrown showing:
Exception
please ask for a valid directory

/var
/www
/html
/admin
/modules
/userman
/Userman.class.php

     */
    public function getAllUsers($directory=null) {
        if(!empty($directory)) {
            $users = $this->directories[$directory]->getAllUsers();
        } else {
            $users = $this->globalDirectory->getAllUsers();
        }
        return $users;
    }
 
    /**
    * Get All Groups
    *
    * Get a List of all User Manager users and their data
    *
    * @return array
    */
    public function getAllGroups($directory=null) {
        if (!empty($directory) && empty($this->directories[$directory])) {
            throw new Exception("Please ask for a valid directory");
        }
 
        if(!empty($directory)) {
            $groups = $this->directories[$directory]->getAllGroups();
        } else {
            $groups = $this->globalDirectory->getAllGroups();
        }
        return $groups;
    }
 
    /** Get Default Groups
     *
     * Get a list of all default groups
     *

If I delete the Directory and recreate probably all stored user passwords and access rights will be erased. This is a 180+ users system. Any suggestion to fix the issue without deleting the directory and all user’s group? Already did UCP and User Manager re-install.

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Module Update Issue

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@thinksafe21 wrote:

I have recently purchased the “Call Recording Report” module but am unable to update my modules to make this active.

In Module Admin I still see this module in the listing with the buy option and when I run “Check Online” it gives an error cannot connect to mirror1 or mirror2 repositories.

Is this an issue on my end? Any help is appreciated.

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Fax Configuration and Dial System Fax

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@FastLukas12345 wrote:

So I’m curious about the existing “Fax Configuration” page and “Dial System Fax” feature code. Is there a way to use these to send and receive faxes or do I have to use one of the commercial modules?

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Cant add addresses to whitelist

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@bksales wrote:

Is there a limit to how many people can be in the white list or why would it turn red when I try to add a new line?

I see a recent thread about freepbx 15 having this issue, but this server is 14.0.16.4

Pasting the IP or tryping it out manually doesnt change anything, I can’t click the save button. Tried multiple browsers and I tried clearing cache.

image
image

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Want to add Numbers to my PBX with Caller Names

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@faisalkhan wrote:

I want to add a list of numbers from which when we receive calls they will show the CALLER Name as we saved them with just like Mobile Phone.

e.g Number = 12121212121, Name = US Test

when call comes from the above number it shows us the Caller Name as or Call From US Test instead of their CallerID.

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Trunk between Audiocodes MP-114 FXO and freepbx

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@wa2kjc wrote:

I am having a problem with the audiocodes mp-114 fxo port that is connected to a magicjack dongle. I made a trunk between Freepbx and the mp-114. I can dial out from an extension on the freepbx raspberry server using a voip yealink phone.
I can’t get into the voip extension when dialing the magicjack number. I get a connection and dial tone from the mp-114 but no matter what i dial i get disconnected within one minute. I have tried everything i can think of.
Dave

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Help with setting spam filter

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@moussa854 wrote:

PBX Version: 14.0.16.4
PBX Distro: 12.7.8-2012-1.sng7
Asterisk Version: 16.13.0

We have been using a simple IVR to stop spam calls. I would like to allow some numbers to skip the IVR to reach a human. I saw the post by lgaetz https://community.freepbx.org/t/of-robocalls-and-whitelists/55848 and we would like to use it. I installed the Dynamic Routes and added the lgaetz-cmcheck.php file to /var/lib/asterisk/agi-bin/ folder and chenged the file permission to asterisk:asterisk.
I appreciate help in the next steps:

Field name Value
Dynamic Route Name Allowed DID
Dynamic Route Description Allowed DID
Enable DTMF Input No
Announcement None
Timeout blank
Validation blank
Invalid Retries blank
Invalid Retry Recording None
Invalid Recording None
Invalid Destination Do not set this value
Saved input varibale name None
Saved result variable name None
Source Type AGI
AGI Lookup lgaetz-cmcheck.php
AGI Result Variable True,False
Default Destination IVR: Welcome
Dynamic Route Entries Match: False; Destination: IVR: Welcome
Match: True; Destination: Ring group
  • Do the above Dynamic Routes values looks right?

  • Do I need to add the code below to extensions_custom.conf

         [check-whitelist]
         exten => s,1,Noop(Entering user defined context [check-whitelist] in extensions_custom.conf)
         exten => s,n,AGI(lgaetz-cmcheck.php,${CALLERID(number)})
         exten => s,n,gotoif($["${whitelist}"="false"]?ivr-1,s,1)    ; edit for wherever non-whitelisted calleres should go
         exten => s,n,Return
    
  • Is there anything I need to do?

Thanks for your help and input.

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Blacklist regex does it work?

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@nobby6 wrote:

OK so we all know blacklist numbers works but what about regex?

0[0123578][01].

This should match 0011 and block any number with it, such as 0011642081957415
(or am I mixing something up here)

Yet it did not, it went through to privacy mode “say your name” and passed.

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How to hide phone extension on end user's phone

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@ameylele wrote:

Hi,
If I call anyone from my extension then only my name or display name that I have set should get displayed on end user’s phone (to whom I have called).
I have tried some options like Alias however it is not working. It is showing the extension number as well. Can anyone help me so that I can hide my extension number and only my name will get displayed?
Thanks in advance!

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Clients can find the same agent that speak on a previous call

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@paperakis wrote:

Hello all,

I’m using a freepbx v15 asterisk 16 and i was wondering if a call answered by an agent… and after 1 hour or so the callee do another call is it possible to find the same agent that answered the previous call if he’s available?

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Asterisk - CISCO phone registration via PJSIP

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@Grandpoohbah2 wrote:

Good day to all,

Is it possible to get the following CISCO phone models to register using PJSIP & if so what settings are required?

7821s, 7941’s & 7942s?

They were able to register previously with CHANSIP if I modified a few advanced extension settings, but the same settings don’t appear to be available for the PJSIP driver.

During the registration attempt the request is generated from a port other than IP and port other than 5060, however the device UAC anticipates receiving responses on port 5060 ( as specified within its contact header)

Does anyone have a work around for this? & thank you very much for your time and assistance in advance.

GPB

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Cisco Phone 8831

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@AndreasB wrote:

Hello,

i try to get an cisco phone 8831 working on a “test freepbx”.
Here are some informations:
Asterisk Version: 16.15.1
static ip 10.0.30.17
8831 current firmware: sip8831.9-3-3-5
static ip 10.0.30.35

I created a chan_pjsip extension
To make things easier while i am still testing the extension number and the secret is the same.

In the (commercial) endpoint manager i created a template and linked it to the extension. Because there is no 8831 i have chosen the 9971 as available phone.
I altered the automatically built xml file to this:

SEPAC44F21156C2.cnf.xml
<device>
	<fullConfig>true</fullConfig>
	<deviceProtocol>SIP</deviceProtocol>
	<ipAddressMode>0</ipAddressMode>
<devicePool>
	<dateTimeSetting>
		<dateTemplate>D.M.Y</dateTemplate>
		<timeZone>Central Europe Standard/Daylight Time</timeZone>
		<ntps>
				<name>ntp.pool.org</name>
				<ntpMode>directedbroadcast</ntpMode>
			</ntp>
			<name>ntp.pool.org</name> 
		</ntps>
	</dateTimeSetting>
	<callManagerGroup>
		<members>
			<member priority="0">
				<callManager>
					<ports>
						<sipPort>5060</sipPort>
					</ports>
					<processNodeName>10.0.30.17</processNodeName>
				</callManager>
			</member>
		</members>
	</callManagerGroup>
</devicePool>
<sipProfile>
	<sipProxies>
		<registerWithProxy>true</registerWithProxy>
	</sipProxies>
	<sipCallFeatures>
		<cnfJoinEnabled>true</cnfJoinEnabled>
		<localCfwdEnable>true</localCfwdEnable>
		<callForwardURI>service-uri-cfwdall</callForwardURI>
		<callPickupURI>service-uri-pickup</callPickupURI>
		<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
		<meetMeServiceURI>service-uri-meetme</meetMeServiceURI>
		<callHoldRingback>2</callHoldRingback>
		<semiAttendedTransfer>true</semiAttendedTransfer>
		<anonymousCallBlock>2</anonymousCallBlock>
		<callerIdBlocking>2</callerIdBlocking>
		<dndControl>2</dndControl>
		<dndCallAlert>1</dndCallAlert>
		<remoteCcEnable>true</remoteCcEnable>
	</sipCallFeatures>
	<sipStack>
		<timerRegisterExpires>3600</timerRegisterExpires>
		<timerRegisterDelta>5</timerRegisterDelta>
		<timerInviteExpires>180</timerInviteExpires>
		<timerKeepAliveExpires>3600</timerKeepAliveExpires>
		<timerSubscribeExpires>3600</timerSubscribeExpires>
		<timerSubscribeDelta>5</timerSubscribeDelta>
		<timerT1>500</timerT1>
		<timerT2>4000</timerT2>
		<remotePartyID>true</remotePartyID>
	</sipStack>
	<sipLines>
		<line button="1" lineIndex="1">
			<featureID>9</featureID>
			<featureLabel>CP8831_Test</featureLabel>
			<speedDialNumber></speedDialNumber>
			<proxy>USECALLMANAGER</proxy>
			<port>5060</port>
			<voipControlPort>5060</voipControlPort>
			<name>301</name>
			<displayName>CP8831_Test</displayName>
			<autoAnswer>
				<autoAnswerEnabled>0</autoAnswerEnabled>
			</autoAnswer>
			<callWaiting>1</callWaiting>
			<authName>301</authName>
			<authPassword>301</authPassword>
			<sharedLine>false</sharedLine>
			<messagesNumber></messagesNumber>
			<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
			<ringSettingActive>5</ringSettingActive>
			<forwardCallInfoDisplay>
				<callerName>true</callerName>
				<callerNumber>false</callerNumber>
				<redirectedNumber>false</redirectedNumber>
				<dialedNumber>true</dialedNumber>
			</forwardCallInfoDisplay>
			<maxNumCalls>6</maxNumCalls>
			<busyTrigger>2</busyTrigger>
		</line>
	</sipLines>
	<enableVad>true</enableVad>
	<preferredCodec>g711alaw</preferredCodec>
	<softKeyFile>softkeyDefault_kpml.xml</softKeyFile>
	<dialTemplate></dialTemplate>
	<kpml>0</kpml>
	<phoneLabel>CP8831_Test</phoneLabel>
	<stutterMsgWaiting>0</stutterMsgWaiting>
	<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
	<dscpForAudio>184</dscpForAudio>
	<dscpVideo>136</dscpVideo>
	<startMediaPort>16384</startMediaPort>
	<stopMediaPort>32766</stopMediaPort>
</sipProfile>
<commonProfile>
	<phonePassword></phonePassword>
	<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>sip8831.10-3-1SR3-5-EU</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
	<videoCapability>0</videoCapability>
	<ciscoCamera>0</ciscoCamera>
</commonConfig>
<sshAccess>0</sshAccess>
<sshUserId>301</sshUserId>
<sshPassword>301</sshPassword>
<versionStamp>1332742429550542</versionStamp>
<userLocale>
	<name>SIP_German_Germany</name>
	<langCode>de_DE</langCode>
</userLocale>
<networkLocale>SIP_Germany</networkLocale>
<networkLocaleInfo>
	<name>Germany</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL></servicesURL>
<transportLayerProtocol>1</transportLayerProtocol>
</device>

The phone “kind of” connects to the freepbx. I can call my catch all number and speak in both ways.

But of course i have some problems. In Asterisk info the extension is shown as “unknown” status.
More important i cannot call it. This is a log when i try:

call attempt

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2[2021-02-17 13:47:04] VERBOSE[2349] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘10.0.120.17’
3[2021-02-17 13:47:04] VERBOSE[2349] netsock2.c: Using SIP RTP Audio TOS bits 184
4[2021-02-17 13:47:04] VERBOSE[2349] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
5[2021-02-17 13:47:04] VERBOSE[2349] netsock2.c: Using SIP RTP Audio CoS mark 5
6[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [301@from-internal:1] GotoIf(“PJSIP/333-00000003”, “1?ext-local,301,1:followme-check,301,1”) in new stack
7[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (ext-local,301,1)
8[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [301@ext-local:1] Set(“PJSIP/333-00000003”, “__RINGTIMER=15”) in new stack
9[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [301@ext-local:2] ExecIf(“PJSIP/333-00000003”, “0?Set(__CWIGNORE=)”) in new stack
10[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [301@ext-local:3] Macro(“PJSIP/333-00000003”, “exten-vm,novm,301,0,0,0”) in new stack
11[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:1] Macro(“PJSIP/333-00000003”, “user-callerid,”) in new stack
12[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/333-00000003”, “TOUCH_MONITOR=1613566024.4”) in new stack
13[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/333-00000003”, “AMPUSER=333”) in new stack
14[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:3] Set(“PJSIP/333-00000003”, “HOTDESCKCHAN=333-00000003”) in new stack
15[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:4] Set(“PJSIP/333-00000003”, “HOTDESKEXTEN=333”) in new stack
16[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:5] Set(“PJSIP/333-00000003”, “HOTDESKCALL=0”) in new stack
17[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:6] ExecIf(“PJSIP/333-00000003”, “0?Set(HOTDESKCALL=1)”) in new stack
18[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:7] ExecIf(“PJSIP/333-00000003”, “0?Set(CALLERID(name)=)”) in new stack
19[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:8] GotoIf(“PJSIP/333-00000003”, “0?report”) in new stack
20[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:9] ExecIf(“PJSIP/333-00000003”, “1?Set(REALCALLERIDNUM=333)”) in new stack
21[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:10] Set(“PJSIP/333-00000003”, “AMPUSER=333”) in new stack
22[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:11] GotoIf(“PJSIP/333-00000003”, “0?limit”) in new stack
23[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:12] Set(“PJSIP/333-00000003”, “AMPUSERCIDNAME=77333 abi Softphone”) in new stack
24[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/333-00000003”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
25[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/333-00000003”, “0?report”) in new stack
26[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:15] Set(“PJSIP/333-00000003”, “AMPUSERCID=333”) in new stack
27[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:16] Set(“PJSIP/333-00000003”, “__DIAL_OPTIONS=HhTtr”) in new stack
28[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:17] Set(“PJSIP/333-00000003”, “CALLERID(all)=“77333 abi Softphone” <333>”) in new stack
29[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:18] ExecIf(“PJSIP/333-00000003”, “0?Set(CUSDIAL=301)”) in new stack
30[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:19] ExecIf(“PJSIP/333-00000003”, “0?Set(CALLERID(all)=“77333 abi Softphone” <333>)”) in new stack
31[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:20] GotoIf(“PJSIP/333-00000003”, “0?limit”) in new stack
32[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:21] ExecIf(“PJSIP/333-00000003”, “0?Set(GROUP(concurrency_limit)=333)”) in new stack
33[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:22] ExecIf(“PJSIP/333-00000003”, “0?Set(CHANNEL(language)=)”) in new stack
34[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:23] NoOp(“PJSIP/333-00000003”, “Macro Depth is 2”) in new stack
35[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:24] GotoIf(“PJSIP/333-00000003”, “1?report2:macroerror”) in new stack
36[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-user-callerid,s,25)
37[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:25] GotoIf(“PJSIP/333-00000003”, “0?continue”) in new stack
38[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:26] ExecIf(“PJSIP/333-00000003”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
39[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:27] Set(“PJSIP/333-00000003”, “__TTL=64”) in new stack
40[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:28] GotoIf(“PJSIP/333-00000003”, “1?continue”) in new stack
41[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-user-callerid,s,44)
42[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:44] Set(“PJSIP/333-00000003”, “CALLERID(number)=333”) in new stack
43[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:45] Set(“PJSIP/333-00000003”, “CALLERID(name)=77333 abi Softphone”) in new stack
44[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:46] GotoIf(“PJSIP/333-00000003”, “0?cnum”) in new stack
45[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:47] Set(“PJSIP/333-00000003”, “CDR(cnam)=77333 abi Softphone”) in new stack
46[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:48] Set(“PJSIP/333-00000003”, “CDR(cnum)=333”) in new stack
47[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-user-callerid:49] Set(“PJSIP/333-00000003”, “CHANNEL(language)=de_DE”) in new stack
48[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:2] Set(“PJSIP/333-00000003”, “RingGroupMethod=none”) in new stack
49[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:3] Set(“PJSIP/333-00000003”, “__EXTTOCALL=301”) in new stack
50[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:4] Set(“PJSIP/333-00000003”, “__PICKUPMARK=301”) in new stack
51[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:5] Set(“PJSIP/333-00000003”, “RT=”) in new stack
52[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
53[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:6] ExecIf(“PJSIP/333-00000003”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
54[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
55[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
56[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:7] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
57[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
58[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
59[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:8] ExecIf(“PJSIP/333-00000003”, “0?Gosub(ext-intercom,*80301,1())”) in new stack
60[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
61[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
62[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:9] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
63[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
64[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
65[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:10] ExecIf(“PJSIP/333-00000003”, “0?ChanSpy(PJSIP/301,q)”) in new stack
66[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
67[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
68[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:11] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
69[2021-02-17 13:47:04] WARNING[11147][C-00000009] chan_sip.c: This function can only be used on SIP channels.
70[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:12] ExecIf(“PJSIP/333-00000003”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
71[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:13] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
72[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:14] ExecIf(“PJSIP/333-00000003”, “0?Gosub(ext-intercom,*80301,1())”) in new stack
73[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:15] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
74[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:16] ExecIf(“PJSIP/333-00000003”, “0?ChanSpy(PJSIP/301,q)”) in new stack
75[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:17] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
76[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:18] Gosub(“PJSIP/333-00000003”, “sub-record-check,s,1(exten,301,dontcare)”) in new stack
77[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:1] GotoIf(“PJSIP/333-00000003”, “0?initialized”) in new stack
78[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:2] Set(“PJSIP/333-00000003”, “__REC_STATUS=INITIALIZED”) in new stack
79[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:3] Set(“PJSIP/333-00000003”, “NOW=1613566024”) in new stack
80[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:4] Set(“PJSIP/333-00000003”, “__DAY=17”) in new stack
81[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:5] Set(“PJSIP/333-00000003”, “__MONTH=02”) in new stack
82[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:6] Set(“PJSIP/333-00000003”, “__YEAR=2021”) in new stack
83[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:7] Set(“PJSIP/333-00000003”, “__TIMESTR=20210217-134704”) in new stack
84[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:8] Set(“PJSIP/333-00000003”, “__FROMEXTEN=333”) in new stack
85[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:9] Set(“PJSIP/333-00000003”, “__MON_FMT=wav”) in new stack
86[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:10] NoOp(“PJSIP/333-00000003”, “Recordings initialized”) in new stack
87[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:11] ExecIf(“PJSIP/333-00000003”, “0?Set(ARG3=dontcare)”) in new stack
88[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:12] Set(“PJSIP/333-00000003”, “REC_POLICY_MODE_SAVE=”) in new stack
89[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:13] ExecIf(“PJSIP/333-00000003”, “0?Set(REC_STATUS=NO)”) in new stack
90[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:14] GotoIf(“PJSIP/333-00000003”, “5?checkaction”) in new stack
91[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (sub-record-check,s,17)
92[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@sub-record-check:17] GotoIf(“PJSIP/333-00000003”, “1?sub-record-check,exten,1”) in new stack
93[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (sub-record-check,exten,1)
94[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:1] NoOp(“PJSIP/333-00000003”, “Exten Recording Check between 333 and 301”) in new stack
95[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:2] Set(“PJSIP/333-00000003”, “CALLTYPE=internal”) in new stack
96[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:3] ExecIf(“PJSIP/333-00000003”, “0?Set(CALLTYPE=)”) in new stack
97[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:4] Set(“PJSIP/333-00000003”, “CALLEE=dontcare”) in new stack
98[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:5] ExecIf(“PJSIP/333-00000003”, “0?Set(CALLEE=dontcare)”) in new stack
99[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:6] GotoIf(“PJSIP/333-00000003”, “0?callee”) in new stack
100[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:7] GotoIf(“PJSIP/333-00000003”, “1?caller”) in new stack
101[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (sub-record-check,exten,13)
102[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:13] Set(“PJSIP/333-00000003”, “RECMODE=dontcare”) in new stack
103[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:14] ExecIf(“PJSIP/333-00000003”, “0?Set(RECMODE=dontcare)”) in new stack
104[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:15] ExecIf(“PJSIP/333-00000003”, “1?Set(RECMODE=dontcare)”) in new stack
105[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:16] Gosub(“PJSIP/333-00000003”, “recordcheck,1(dontcare,internal,301)”) in new stack
106[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/333-00000003”, “Starting recording check against dontcare”) in new stack
107[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/333-00000003”, “dontcare”) in new stack
108[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
109[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:3] Return(“PJSIP/333-00000003”, “”) in new stack
110[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [exten@sub-record-check:17] Return(“PJSIP/333-00000003”, “”) in new stack
111[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:19] GotoIf(“PJSIP/333-00000003”, “1?macrodial”) in new stack
112[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-exten-vm,s,25)
113[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:25] GosubIf(“PJSIP/333-00000003”, “0?clrheader,1()”) in new stack
114[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:26] Macro(“PJSIP/333-00000003”, “dial-one,HhTtr,301”) in new stack
115[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:1] Set(“PJSIP/333-00000003”, “DEXTEN=301”) in new stack
116[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:2] Set(“PJSIP/333-00000003”, “__CRM_SOURCE=333”) in new stack
117[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:3] ExecIf(“PJSIP/333-00000003”, “0?Set(__EXTTOCALL=301)”) in new stack
118[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:4] Set(“PJSIP/333-00000003”, “DIALSTATUS_CW=”) in new stack
119[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:5] GosubIf(“PJSIP/333-00000003”, “0?screen,1()”) in new stack
120[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:6] GosubIf(“PJSIP/333-00000003”, “0?cf,1()”) in new stack
121[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:7] GotoIf(“PJSIP/333-00000003”, “1?skip1”) in new stack
122[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-dial-one,s,10)
123[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:10] GotoIf(“PJSIP/333-00000003”, “0?nodial”) in new stack
124[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:11] GotoIf(“PJSIP/333-00000003”, “0?continue”) in new stack
125[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:12] Set(“PJSIP/333-00000003”, “EXTHASCW=ENABLED”) in new stack
126[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:13] GotoIf(“PJSIP/333-00000003”, “0?next1:cwinusebusy”) in new stack
127[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-dial-one,s,25)
128[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:25] GotoIf(“PJSIP/333-00000003”, “0?next3:continue”) in new stack
129[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-dial-one,s,27)
130[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:27] GotoIf(“PJSIP/333-00000003”, “0?nodial”) in new stack
131[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:28] GosubIf(“PJSIP/333-00000003”, “1?dstring,1():dlocal,1()”) in new stack
132[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:1] Set(“PJSIP/333-00000003”, “DSTRING=”) in new stack
133[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:2] Set(“PJSIP/333-00000003”, “DEVICES=301”) in new stack
134[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/333-00000003”, “0?Return()”) in new stack
135[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/333-00000003”, “0?Set(DEVICES=01)”) in new stack
136[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:5] Set(“PJSIP/333-00000003”, “LOOPCNT=1”) in new stack
137[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:6] Set(“PJSIP/333-00000003”, “ITER=1”) in new stack
138[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:7] Set(“PJSIP/333-00000003”, “THISDIAL=PJSIP/301”) in new stack
139[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:8] GotoIf(“PJSIP/333-00000003”, “0?docheck”) in new stack
140[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:9] NoOp(“PJSIP/333-00000003”, “Debug: Found PJSIP Destination PJSIP/301”) in new stack
141[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:10] GotoIf(“PJSIP/333-00000003”, “0?doset”) in new stack
142[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:11] NoOp(“PJSIP/333-00000003”, “Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS”) in new stack
143[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:12] Set(“PJSIP/333-00000003”, “THISDIAL=”) in new stack
144[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:13] ExecIf(“PJSIP/333-00000003”, “1?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack
145[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:14] GotoIf(“PJSIP/333-00000003”, “1?skipset”) in new stack
146[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-dial-one,dstring,16)
147[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:16] Set(“PJSIP/333-00000003”, “ITER=2”) in new stack
148[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:17] GotoIf(“PJSIP/333-00000003”, “0?begin”) in new stack
149[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [dstring@macro-dial-one:18] ExecIf(“PJSIP/333-00000003”, “1?Return()”) in new stack
150[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:29] GotoIf(“PJSIP/333-00000003”, “1?nodial”) in new stack
151[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-dial-one,s,61)
152[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:61] NoOp(“PJSIP/333-00000003”, “”) in new stack
153[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:62] ExecIf(“PJSIP/333-00000003”, “0?Set(DIALSTATUS=NOANSWER)”) in new stack
154[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:63] NoOp(“PJSIP/333-00000003”, “Returned from dial-one with nothing to call and DIALSTATUS: CHANUNAVAIL”) in new stack
155[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-dial-one:64] MacroExit(“PJSIP/333-00000003”, “”) in new stack
156[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:27] Set(“PJSIP/333-00000003”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
157[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:28] GosubIf(“PJSIP/333-00000003”, “0?docfu,1()”) in new stack
158[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:29] GosubIf(“PJSIP/333-00000003”, “0?docfb,1()”) in new stack
159[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:30] Set(“PJSIP/333-00000003”, “DIALSTATUS=CHANUNAVAIL”) in new stack
160[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:31] ExecIf(“PJSIP/333-00000003”, “0?MacroExit()”) in new stack
161[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-exten-vm:32] GotoIf(“PJSIP/333-00000003”, “1?s-CHANUNAVAIL,1”) in new stack
162[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-exten-vm,s-CHANUNAVAIL,1)
163[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf(“PJSIP/333-00000003”, “0?exit,1”) in new stack
164[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones(“PJSIP/333-00000003”, “congestion”) in new stack
165[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion(“PJSIP/333-00000003”, “10”) in new stack
166[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] app_macro.c: Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on ‘PJSIP/333-00000003’ in macro ‘exten-vm’
167[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Spawn extension (ext-local, 301, 3) exited non-zero on ‘PJSIP/333-00000003’
168[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [h@ext-local:1] Macro(“PJSIP/333-00000003”, “hangupcall,”) in new stack
169[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/333-00000003”, “1?theend”) in new stack
170[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-hangupcall,s,3)
171[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/333-00000003”, “0?Set(CDR(recordingfile)=)”) in new stack
172[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/333-00000003”, " montior file= ") in new stack
173[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/333-00000003”, “1?skipagi”) in new stack
174[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx_builtins.c: Goto (macro-hangupcall,s,7)
175[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/333-00000003”, “”) in new stack
176[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/333-00000003’ in macro ‘hangupcall’
177[2021-02-17 13:47:04] VERBOSE[11147][C-00000009] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/333-00000003’

The other main problem is that when i call from 8831 only the first typed in digit is transfered.
I remember that i had to use “no digit collect kpml” on the cisco server to circumvent that but i cannot find a similar option in freepbx.

A third (minor for me) problem is that it seems the phone never tries to get the firmware load i configured.

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Freepbx DID issue in outgoing

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@abhishek2021 wrote:

I am not able to my DID as CLI on outgoing calls, I only get my pilot number as CID for all.

below is the dial plan logic which has been configured, help please to resolve the issue. Thank You.

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Could not create dialog to invalid URI?

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@Peter2020 wrote:

Hello, I want to make an outbound call, but it doesn’t work. I always get the info “all lines are busy. Please call again later”. I don’t know what to do to solve the problem. I use a trunk from sipgate (de) (basic). I can make inbound call, but outbound doesn’t work,

res_pjsip.c: Endpoint ‘3330858e0’: Could not create dialog to invalid URI ‘3330858e0’. Is endpoint registered and reachable?
204 [2021-02-17 16:11:32] ERROR[31907] chan_pjsip.c: Failed to create outgoing session to endpoint ‘3330858e0’
205 [2021-02-17 16:11:32] WARNING[17437][C-00000059] app_dial.c: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)

Do someone know how what to do and how to solve this problem?
I am grateful for any advice
Peter.

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Cisco 7975 - Unprovisioned

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@chet wrote:

Evening, wonder if you could help with my Cisco 7975 connecting to version FreePBX 15.0.17.24, PBX server is on 192.168.5.259 and the phones are on 192.168.30.0/24

I’m using SiP ver 8-5-4 on my 7975 with the following XML

     <device>
        <fullConfig>true</fullConfig>
        <deviceProtocol>SIP</deviceProtocol>
        <sshUserId>admin</sshUserId>
        <sshPassword>cisco</sshPassword>
        <devicePool>
            <dateTimeSetting> 
                <dateTemplate>D.M.Y</dateTemplate> 
                <timeZone>GMT Standard/Daylight Time</timeZone> 
                <ntps> 
                    <ntp>
                        <name>0.uk.pool.ntp.org</name> 
                        <ntpMode>Unicast</ntpMode> 
                    </ntp>
                </ntps>
            </dateTimeSetting>
            <callManagerGroup>
                <tftpDefault>true</tftpDefault>
                <members>
                    <member priority="0">
                        <callManager>
                            <ports>
                                <ethernetPhonePort>2000</ethernetPhonePort>
                                <sipPort>5060</sipPort>
                                <securedSipPort>5061</securedSipPort>
                            </ports>
                            <processNodeName>192.168.5.249</processNodeName>
                        </callManager>
                    </member>
                </members>
            </callManagerGroup>
        </devicePool>
        <commonProfile>
            <phonePassword></phonePassword>
            <backgroundImageAccess>true</backgroundImageAccess>
            <callLogBlfEnabled>0</callLogBlfEnabled>
        </commonProfile>
        <loadInformation>SIP75.8-5-4S</loadInformation>
        <vendorConfig>
            <disableSpeaker>false</disableSpeaker>
            <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
            <pcPort>0</pcPort>
            <settingsAccess>1</settingsAccess>
            <garp>0</garp>
            <voiceVlanAccess>0</voiceVlanAccess>
            <videoCapability>0</videoCapability>
            <autoSelectLineEnable>0</autoSelectLineEnable>
            <daysDisplayNotActive>1,7</daysDisplayNotActive>
            <displayOnTime>10:30</displayOnTime>
            <displayOnDuration>06:05</displayOnDuration>
            <displayIdleTimeout>00:05</displayIdleTimeout> 
            <webAccess>0</webAccess>
            <spanToPCPort>1</spanToPCPort>
            <loggingDisplay>1</loggingDisplay>
            <loadServer></loadServer>
        </vendorConfig>
        <userLocale>
            <name>United_Kingdom</name>
            <uid>1</uid>
            <langCode>en_US</langCode>
            <version>1.0.0.0-1</version>
            <winCharSet>iso-8859-1</winCharSet>
        </userLocale>
        <networkLocale>United_Kingdom</networkLocale> 
        <networkLocaleInfo> 
            <name>United_Kingdom</name> 
            <uid>64</uid> 
            <version>1.0.0.0-1</version> 
        </networkLocaleInfo> 
        <deviceSecurityMode>1</deviceSecurityMode>
        <authenticationURL>http://192.168.5.249/xmlservices/authentication.php</authenticationURL>
        <directoryURL>http://192.168.249/xmlservices/PhoneDirectory.php</directoryURL>
        <idleTimeout>0</idleTimeout>
        <idleURL></idleURL>
        <informationURL>http://192.168.5.249/xmlservices/index.php</informationURL>
        <messagesURL></messagesURL>
        <proxyServerURL></proxyServerURL>
        <servicesURL>http://192.168.5.249/xmlservices/index.php</servicesURL>
        <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
        <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
        <dscpForCm2Dvce>96</dscpForCm2Dvce>
        <transportLayerProtocol>4</transportLayerProtocol>
        <capfAuthMode>0</capfAuthMode>
        <capfList>
            <capf>
                <phonePort>3804</phonePort>
            </capf>
        </capfList>
        <certHash></certHash>
        <encrConfig>false</encrConfig>
        <sipProfile>
            <sipProxies>
                <backupProxy>192.168.5.249</backupProxy>
                <backupProxyPort>5060</backupProxyPort>
                <emergencyProxy></emergencyProxy>
                <emergencyProxyPort></emergencyProxyPort>
                <outboundProxy></outboundProxy>
                <outboundProxyPort></outboundProxyPort>
                <registerWithProxy>true</registerWithProxy>
            </sipProxies>
            <sipCallFeatures>
                <cnfJoinEnabled>true</cnfJoinEnabled>
                <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
                <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
                <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
                <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
                <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
                <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
                <rfc2543Hold>true</rfc2543Hold>
                <callHoldRingback>2</callHoldRingback>
                <localCfwdEnable>true</localCfwdEnable>
                <semiAttendedTransfer>true</semiAttendedTransfer>
                <anonymousCallBlock>2</anonymousCallBlock>
                <callerIdBlocking>0</callerIdBlocking>
                <dndControl>0</dndControl>
                <remoteCcEnable>true</remoteCcEnable>
            </sipCallFeatures>
            <sipStack>
                <sipInviteRetx>6</sipInviteRetx>
                <sipRetx>10</sipRetx>
                <timerInviteExpires>180</timerInviteExpires>
                <timerRegisterExpires>3600</timerRegisterExpires>
                <timerRegisterDelta>5</timerRegisterDelta>
                <timerKeepAliveExpires>120</timerKeepAliveExpires>
                <timerSubscribeExpires>120</timerSubscribeExpires>
                <timerSubscribeDelta>5</timerSubscribeDelta>
                <timerT1>500</timerT1>
                <timerT2>4000</timerT2>
                <maxRedirects>70</maxRedirects>
                <remotePartyID>false</remotePartyID>
                <userInfo>None</userInfo>
            </sipStack>
            <autoAnswerTimer>1</autoAnswerTimer>
            <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
            <autoAnswerOverride>true</autoAnswerOverride>
            <transferOnhookEnabled>true</transferOnhookEnabled>
            <enableVad>false</enableVad>
            <preferredCodec>g711u</preferredCodec>
            <dtmfAvtPayload>101</dtmfAvtPayload>
            <dtmfDbLevel>3</dtmfDbLevel>
            <dtmfOutofBand>avt</dtmfOutofBand>
            <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
            <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
            <kpml>3</kpml>
            <stutterMsgWaiting>1</stutterMsgWaiting>
            <callStats>false</callStats>
            <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
            <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
            <startMediaPort>16384</startMediaPort>
            <stopMediaPort>32766</stopMediaPort>
            <voipControlPort>5060</voipControlPort>
            <dscpForAudio>184</dscpForAudio>
            <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
            <dialTemplate>dialplan.xml</dialTemplate>
            <phoneLabel>Chetnet LTD </phoneLabel>
            <natEnabled>true</natEnabled>
            <sipLines>
                <line button="1">
                    <featureID>9</featureID>
                    <featureLabel>Martyn</featureLabel>
                    <name>001FCA368894</name>
                    <displayName>Martyn</displayName>
                    <contact>CONTACT</contact>
                    <proxy>192.168.5.249</proxy>
                    <port>5060</port>
                    <autoAnswer>
                        <autoAnswerEnabled>2</autoAnswerEnabled>
                    </autoAnswer>
                    <callWaiting>3</callWaiting>
                    <authName>3000</authName>
                    <authPassword>12341234</authPassword>
                    <sharedLine>false</sharedLine>
                    <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                    <messagesNumber>*97</messagesNumber>
                    <ringSettingIdle>4</ringSettingIdle>
                    <ringSettingActive>5</ringSettingActive>
                    <forwardCallInfoDisplay>
                        <callerName>true</callerName>
                        <callerNumber>false</callerNumber>
                        <redirectedNumber>false</redirectedNumber>
                        <dialedNumber>true</dialedNumber>
                    </forwardCallInfoDisplay>
                </line>
            </sipLines>
        </sipProfile>
    </device>

I’m getting an IP address and the phone is grabbing the XML from my TFTP server but it just sits there saying Unprovisioned.

pbx

Thanks

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Time conditions based on calendar mode using iCal and google calendar integration

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@Scion wrote:

Hello,

I was able to get the calendar created easily and integrated with a Google calendar. I understand that the calendar must be manipulated from the google account.

When using calendar mode in a time condition, how does they system know to “holiday” mode the actual holidays on the calendar. How does one establish normal business hours on the integrated calendar so the system knows open hours?

Has anyone done this because I am having trouble finding any substantial documentation.

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Core 15.0.12.48 changes something with outbound CID logic

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@sorvani wrote:

I upgraded to core 15.0.12.48 (via fwconsole ma upgrade core --edge) yesterday. Right now, I have no idea why. Brain not working.

Either way outbound calls to T-Mobile started failing. Looking at sngrep, the from was my extension instead of the CID that should have been set.

image

rolled back to 15.0.12.46 and the problem went away.
image

Upgraded to edge again to verify and got version 15.0.12.50.
There was no problem with this version.

Manually installed version 15.0.12.48 and the problem came back.

sudo fwconsole ma downloadinstall core --tag=15.0.12.48

Upgraded back to 15.0.12.50 and all good again.

sudo fwconsole ma upgrade core --edge

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After IVR change, one of the extensions cannot be dialed to

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@MacsOffice wrote:

v14.0.16.4

I just changed the IVR config from a scheduled IVR paths with a greeting that prompted press 1 for paul 2 for Ringo etc, to a single greeting with no schedule and they are given the 3 digit extension to dial directly. I don’t like this option but it’s what the new boss wanted.
After setting that up, I have a single extension, 234, The bosses of course… that when dialed from teh IVR prompt will return to the IVR prompt after timeout and never actually dial the extension. All the others work fine and that extension can be dialed internally just fine.
I cant’ see any difference between this extension and any other.
Ideas please?

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从虚拟机导入ova后,初始化配置,不成功

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@simperfect wrote:

麻烦问下,我要对接freeswitch,从虚拟机导入ova后,一些相关网络,如何配置?
一直初始化,无法继续配置TIM%E5%9B%BE%E7%89%8720210218094350

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AGI script doesn't work

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@snaggy wrote:

extensions_custom.conf

[sms-sender]
exten => s,1,Noop(send sms to ${CALLERID(num)})
exten => s,n,AGI(sms.php, ${CALLERID(number)})
exten => s,n,Goto(app-announcement-9,s,1)

sms.php

#!/usr/bin/php -q
<?php
require('phpagi.php'); 
$agi = new AGI();
$cid = $agi->request['agi_callerid'];
$user = "smsuser";
$pas = "smsuser";
$text = "message ";
$sms = str_replace(" ", "+", $text);
$prov = "1";
$meth = "2";
$url = "https://my-domain/goip/en/dosend.php?USERNAME=$user&PASSWORD=$pas&smsprovider=$prov&smsnum=%2B$cid&method=$meth&Memo=$sms";
	
if(preg_match('/^79[0-9]{9}/',$cid))	{
	$ch = curl_init();
	curl_setopt($ch, CURLOPT_URL, $url);
	curl_setopt($ch, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);
	curl_setopt($ch, CURLOPT_RETURNTRANSFER, true);
	$out = curl_exec($ch);
	curl_close($ch);
	echo $out;
}
else	{
	echo "not mobile";
}
?>

in the console I see an error message

-- Executing [2@ivr-6:1] Set("SIP/XXXXXXXXXXX-00000232", "__ivrreturn=0") in new stack
-- Executing [2@ivr-6:2] Goto("SIP/XXXXXXXXXXX-00000232", "customdests,dest-4,1") in new stack
-- Goto (customdests,dest-4,1)
-- Executing [dest-4@customdests:1] NoOp("SIP/XXXXXXXXXXX-00000232", "Entering Custom Destination sms-sender") in new stack
-- Executing [dest-4@customdests:2] Gosub("SIP/XXXXXXXXXXX-00000232", "sms-sender,s,1()") in new stack
-- Executing [s@sms-sender:1] NoOp("SIP/XXXXXXXXXXX-00000232", "send sms to XXXXXXXXXXX") in new stack
-- Executing [s@sms-sender:2] AGI("SIP/XXXXXXXXXXX-00000232", "sms.php, XXXXXXXXXXX") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sms.php
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_request: sms.php
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_channel: SIP/XXXXXXXXXXX-00000232
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_language: ru
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_type: SIP
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_uniqueid: 1613638224.1108
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_version: 16.15.1
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_callerid: XXXXXXXXXXX
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_calleridname: XXXXXXXXXXX
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_callingpres: 0
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_callingani2: 0
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_callington: 0
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_callingtns: 0
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_dnid: XXXXXXXXXXX
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_rdnis: unknown
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_context: sms-sender
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_extension: s
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_priority: 2
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_enhanced: 0.0
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_accountcode:
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_threadid: 140256500864768
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> agi_arg_1:  XXXXXXXXXXX
<SIP/XXXXXXXXXXX-00000232>AGI Tx >>
<SIP/XXXXXXXXXXX-00000232>AGI Rx << Could not open input file:  XXXXXXXXXXX
<SIP/XXXXXXXXXXX-00000232>AGI Tx >> 510 Invalid or unknown command
-- <SIP/XXXXXXXXXXX-00000232>AGI Script sms.php completed, returning 0
-- Executing [s@sms-sender:3] Goto("SIP/XXXXXXXXXXX-00000232", "app-announcement-9,s,1") in new stack
-- Goto (app-announcement-9,s,1)

where did i go wrong?

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