@codec wrote:

Hello,

when logging into the CLI I get the following:

±-----------------------------------------------------------+
| There are 203 System updates available. |
| Run yum update to update them. |
| Your PBX is up to date. |
±-----------------------------------------------------------+

So I run yum update… this gives me:

–> Processing Dependency: kmod-forcedeth for package: sangoma-pbx-1805-3.sng7.noarch
–> Finished Dependency Resolution
Error: Package: sangoma-pbx-1805-3.sng7.noarch (sng-pkgs)
Requires: kmod-forcedeth
Removing: kmod-forcedeth-0.65-1.sng7.x86_64 (@anaconda/1707)
kmod-forcedeth = 0.65-1.sng7
Available: kmod-forcedeth-0.64-3.sng7.x86_64 (sng-pkgs)
kmod-forcedeth = 0.64-3.sng7
Available: kmod-forcedeth-0.64-4.sng7.x86_64 (sng-pkgs)
kmod-forcedeth = 0.64-4.sng7
Available: kmod-forcedeth-0.64-5.sng7.x86_64 (sng-pkgs)
kmod-forcedeth = 0.64-5.sng7
You could try using --skip-broken to work around the problem
You could try running: rpm -Va --nofiles --nodigest

I tried the --skip-broken and the rpm -Va --nofiles --nodigest
with no success…
What should I try next?

Version of freepbx is 14.0.3.6

Thanks
Codec

Posts: 2

Participants: 2

Read full topic

@fearx wrote:

Greetings.

I wanted to enable the recording function when needed (via * 1). Here are my settings:
(I only use internal calls) (that’s the same in all extensions)

Recording Options
Inbound External Calls Force Yes Don’t Care No Never
Outbound External Calls Force Yes Don’t Care No Never
Inbound Internal Calls Force Yes Don’t Care No Never
Outbound Internal Calls Force Yes Don’t Care No Never
On Demand Recording Disable Enable Override
Record Priority Policy 10

The call is recorded, but I still hear the “access denied” sound on the call. Any tips on that?

Posts: 1

Participants: 1

Read full topic

@dkimble wrote:

I was hoping someone could point us in the right direction. We are in the process of starting a virtual receptionist company. We were and hopefully will still use the software Open Answer. If you are not familiar with the software it basically also puts on there FreePBX, and Asterisk, which is how I ended up here. Support for Open Answer seems to be non-existent now.

So, I was curious if anyone here is using Open Answer and if so do you have a web developer you can recommend for us to use? We can’t seem to find anyone with experience with this system.

We have Open Answer pretty much setup but we can’t seem to figure out how to setup the calls within the system.

Does anyone have any recommendations for other software to use?

Thanks for any help.

Posts: 2

Participants: 2

Read full topic

@fearx wrote:

I did a talking test for about 2 minutes, but the CDR reported a 3-second audio. I use * 1 to write, and these are the settings on all users (I only make internal calls)

Recording Options
Inbound External Calls Force Yes Don’t Care No Never
Outbound External Calls Force Yes Don’t Care No Never
Inbound Internal Calls Force Yes Don’t Care No Never
Outbound Internal Calls Force Yes Don’t Care No Never
On Demand Recording Disable Enable Override
Record Priority Policy 10

Posts: 1

Participants: 1

Read full topic

@handleric wrote:

I’m having some issues with permissions in user manager i’m not sure if it’s a bug.

Say I have an admin that’s part of both my FreePBX Administrators group AND my UCP Users group. In my case it’s only respecting the access settings from the FreePBX Administrators group thus not able to access UCP. There should be a “Null” option for group privilege settings instead of just “Yes or No” so that I can create a group and only define relevant permissions for exact scenarios as this where groups may overlap.

Posts: 4

Participants: 2

Read full topic

@yannickg wrote:

Hi,

I have some local extensions and outbound calling working with a Twilio SIP Trunk, I also configured the inbound route but I always get this error when calling in.

[2018-07-14 11:45:59] ERROR[12401]: res_pjsip.c:3115 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss’

Anybody has a clue?

I can post logs if anyone has time to look into this.

Thanks

Posts: 2

Participants: 2

Read full topic

@wars_t wrote:

Hi,

I’m using the latest versions of Asterisk 13 & FreePBX 14 with all the updates as this system was just built on Thursday.

When I try to generate a certificate, the system sits at ‘Generating… Please wait’ and does nothing. I can’t open another session of the GUI in a new tab it just sits there, however I can access through SSH. Rebooting kicks it all back into life.

Ports 80 & 443 are open on the firewall, external DNS resolves to the WAN IP so, not really sure where to go from here… Any ideas? Couldn’t find any letsencrypt specific logs and I also read somewhere about making sure the server hostname was the same as the letsencrypt expected one, tried that & no change.

Only other thing that could be a pointer, the dashboard has a warning - Forced MODULEADMINWGET to true.

Thanks for reading!

Posts: 1

Participants: 1

Read full topic

@oberite wrote:

I have an issue where inbound calls from my trunk work only sometimes. There is no pattern I can discern as to what may be causing this failure. It will work for one call, only to fail the next, just to work on the third one; other times, it will work without error for several hours.

When a call fails, I see “No matching peer” in my logs, followed by returning 401 Unauthorized to my trunk provider.

My provider is Voyant, and I am using IP authentication.

Here is a log snippet of when it works:

<--- SIP read from UDP:137.192.80.33:5060 --->
INVITE sip:+1231597XXXX@206.189.X.X:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKqo2r1n206o9mpatbljp0.1
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c2c5cff
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628223420_113246044@199.199.12.56
CSeq: 443724 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
P-Asserted-Identity: "CNAM " <sip:+1231818XXXX@137.192.80.33:5060>
Supported: timer,100rel,precondition
Session-Expires: 1800
Min-SE: 90
Content-Length: 364
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 77861 169058 IN IP4 137.192.80.33
s=SIP Media Capabilities
c=IN IP4 137.192.80.33
t=0 0
m=audio 25498 RTP/AVP 96 0 18 101 100
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-15
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
a=record:on
<------------->
[2018-07-15 15:25:18] VERBOSE[17369] chan_sip.c: --- (17 headers 16 lines) ---
[2018-07-15 15:25:18] VERBOSE[17369] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Using INVITE request as basis request - 1628223420_113246044@199.199.12.56
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found peer 'voyant' for '+1231818XXXX' from 137.192.80.33:5060
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] netsock2.c: Using SIP RTP TOS bits 184
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] netsock2.c: Using SIP RTP CoS mark 5
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 96
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 0
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 18
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 101
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 100
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format opus for ID 96
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format PCMU for ID 0
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format G729 for ID 18
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found unknown media description format telephone-event for ID 101
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format telephone-event for ID 100
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726|g722|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|g729|opus)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Peer audio RTP is at port 137.192.80.33:25498
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Looking for +1231597XXXX in from-pstn-e164-us (domain 206.189.X.X)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] sip/route.c: sip_route_dump: route/path hop: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: 
<--- Transmitting (no NAT) to 137.192.80.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKqo2r1n206o9mpatbljp0.1;received=137.192.80.33
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c2c5cff
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628223420_113246044@199.199.12.56
CSeq: 443724 INVITE
Server: FPBX-13.0.195.4(13.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+1231597XXXX@206.189.X.X:5060>
Content-Length: 0

And when it doesn’t:

<--- SIP read from UDP:137.192.80.33:5060 --->
INVITE sip:+1231597XXXX@206.189.X.X:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKostdol30boa06ev7lud0.1
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c3466a1
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628224886_117117091@199.199.12.56
CSeq: 529262 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
P-Asserted-Identity: "CNAM " <sip:+1231818XXXX@137.192.80.33:5060>
Supported: timer,100rel,precondition
Session-Expires: 1800
Min-SE: 90
Content-Length: 365
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 600124 360022 IN IP4 137.192.80.33
s=SIP Media Capabilities
c=IN IP4 137.192.80.33
t=0 0
m=audio 25440 RTP/AVP 96 0 18 101 100
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-15
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
a=record:on
<------------->
[2018-07-15 15:31:43] VERBOSE[17369] chan_sip.c: --- (17 headers 16 lines) ---
[2018-07-15 15:31:43] VERBOSE[17369] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: Using INVITE request as basis request - 1628224886_117117091@199.199.12.56
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: No matching peer for '+1231818XXXX' from '137.192.80.33:5060'
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 137.192.80.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKostdol30boa06ev7lud0.1;received=137.192.80.33
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c3466a1
To: <sip:+1231597XXXX@137.192.80.33:5060>;tag=as57460408
Call-ID: 1628224886_117117091@199.199.12.56
CSeq: 529262 INVITE
Server: FPBX-13.0.195.4(13.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44b2217b"
Content-Length: 0

PEER Details:

type=peer
trustrpid=no
sendrpid=no
insecure=port,invite
host=X.st.sip.global
dtmfmode=rfc2833
;disallow=all
context=from-pstn-e164-us
canreinvite=no
;allow=ulaw
allow=all

I changed the allow/disallow settings thinking that the trunk may be choosing random codecs (grasping at straws), but both configurations work the same.

I have read through other posts reporting similar issues, but those seemed to happen when the provider is using more than one IP address for invites, but that isn’t the case with me.

It may be important to know that I also have a Vitelity trunk configured, which is working fine.

I would appreciate any input on this issue.

Posts: 6

Participants: 3

Read full topic

@PitzKey wrote:

Hello everyone.

Issue is basically with internal calls.

We have a ring group 200 which rings to 201, 202 & 203. Employees were instructed to call 200, but they somehow are still calling the individual extensions.
Se we added the following in extensions_custom.conf for each extension

[from-internal]
exten => 201,1,Answer()
exten => 201,n,Goto(ext-group,200,1)

It worked perfectly fine.

We are trying to set this up with queues.

[from-internal]
exten => 201,1,Answer()
exten => 201,n,Goto(ext-queues,200,1)

But the above is just causing a loop.

Can anyone please guide me how to accomplish this with queues?

Thanks

Posts: 1

Participants: 1

Read full topic

@dobrosavljevic wrote:

So a new thing started happening recently on a couple of new PBX installations.

When somebody calls out from a SIP phone that’s going through the PBX (Polycom with Vitelity as the trunk) if they reach the VM greeting of the person they are calling they are intermittently not able to leave a VM. The greeting plays for them but then they never hear the beep and then the call hangs up on them after a few seconds of silence.

This does not happen every time, end users estimate about half the time but they might be overestimating.

Thus far I am unable to get anything from the asterisk CLI as they are unable to replicate the issue for every call.

Has anyone run into this issue before?

Posts: 3

Participants: 2

Read full topic

@mvogel4949 wrote:

I connected to an AD and in the process created almost 10k of users. Now I need to delete them, but I see no delete button when looking through the users. If I delete the AD Directory will that then delete the users it made?

Posts: 2

Participants: 2

Read full topic

@nicklan000 wrote:

Is it possible to record video from calls?
I set recording options to force, but I can’t see the video in my recording folder folder (only wav recordings)

Posts: 1

Participants: 1

Read full topic

@tobywimberley wrote:

Good morning everyone. I am new to freepbx and have run into an issue with routing calls between two freepbx servers at different locations. Both pbx’s make outgoing calls on the trunk to my sip provider, however, the sip trunk i have between the two isn’t routing. “Sip show peers” in the cli looks like they are connected fine but I get a “500 Internal server error” as a reply from the remote pbx. Below is a selection from my outbound system. I hope it contains the correct information.

Screen Shot 2018-07-17 at 10.24.14 AM.png3554x3166 1.42 MB

Posts: 4

Participants: 2

Read full topic

@mvogel4949 wrote:

I have some SIP extensions that just will not register. When I go into the logfiles I see the following

[2018-07-17 11:53:55] NOTICE[10374] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘sip:120@10.100.200.245’ failed for ‘10.100.200.93:5160’ (callid: 8ace0fd5-5351-4767-a745-38f17d7e8d18) - No matching endpoint found
[2018-07-17 11:53:55] NOTICE[10374] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘sip:120@10.100.200.245’ failed for ‘10.100.200.93:5160’ (callid: 8ace0fd5-5351-4767-a745-38f17d7e8d18) - No matching endpoint found
[2018-07-17 11:53:55] NOTICE[10374] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘sip:120@10.100.200.245’ failed for ‘10.100.200.93:5160’ (callid: 8ace0fd5-5351-4767-a745-38f17d7e8d18) - Failed to authenticate

It looks like it is trying to register as a pjsip? Currently on this system I have pjsip set for 5060 and sip set for 5160. any ideas?

Posts: 2

Participants: 2

Read full topic

@NucleusData wrote:

Hello everyone. I am setting up a SIP trunk between a Cisco router running Call Manager express and my FreePBX system. The pourpose of this is to use the FreePBX system for Voicemail, presence, etc. I am following an article I found on google on setting this up. I can see the trunk is up but when I place a call to the voicemail ext I just get fast busy. A debug of my SIP messages on my Cisco router shows the call being sent to my freepbx system but I am getting 401 unauthorized. A SIP debug on the FreePBX system doesn’t return any messages so I assume the call isnt even getting to the FreePBX server. Please help! I have been stuck on this for days. If needed I will post my router config and SIP messages I am getting to pastebin. Thank you in advance.

Posts: 3

Participants: 2

Read full topic

@johnms wrote:

Hello,

After upgrading Freepbx 13 to 14 , i had a burning flame about UCP not running !
As i am not need it, i decided to uninstall it, and i did !
But as you can see on the attached picture , the ucp login icon is not removed.
Is it possible to remove it?

Thank u!!

Capture|690x308

Posts: 1

Participants: 1

Read full topic

@vespino wrote:

My client deals with both domestic as international customers. I have set up an announcement before the ring group with a kind greeting. I was now wondering if it’s possible to greet clients from abroad in English instead of the native language using multiple announcements.

I was thinking of using a module like “dynamic routes” that checks the CID, assuming the country code is always in the CID. Could that work or are there better solutions for this?

Posts: 1

Participants: 1

Read full topic

@Hawkeye wrote:

After distro-upgrade is complete, it tries to boot default kernel: 3.10.8-862.6.3.e17.x86_64 and this causes a panic.

Reboot it again and select 2nd kernel line: 3.10.0.693.5.2.e17.x86_64 and it boots.

Wondering if anyone knows what the cause is with default kernel line when booting.

Below was the grub line (commented) causing the panic:

## title Sangoma Linux (3.10.0-862.6.3.el7.x86_64) 7 (Core)
## root (hd0,0)
## kernel /vmlinuz-3.10.0-862.6.3.el7.x86_64 root=UUID=969d9a35-508d-4485-a172-0217e279091e ro rd.luks=0 rd.lvm=0 rd.locale.LANG=en_US.UTF-8 rd.md=0 vconsole.font=latarcyrheb-sun16 crashkernel=auto KEYBOARDTYPE=pc vconsole.keymap=us rd.dm=0 rhgb quiet biosdevname=0 net.ifnames=0 LANG=en_US.UTF-8

The grub line that works is:

title Sangoma Linux (3.10.0-693.5.2.el7.x86_64) 7 (Core)
root (hd0,0)
kernel /vmlinuz-3.10.0-693.5.2.el7.x86_64 root=UUID=969d9a35-508d-4485-a172-0217e279091e ro rd.luks=0 rd.lvm=0 rd.locale.LANG=en_US.UTF-8 rd.md=0 vconsole.font=latarcyrheb-sun16 crashkernel=auto KEYBOARDTYPE=pc vconsole.keymap=us rd.dm=0 rhgb quiet biosdevname=0 net.ifnames=0 LANG=en_US.UTF-8
initrd /initramfs-3.10.0-693.5.2.el7.x86_64.img

Posts: 1

Participants: 1

Read full topic

@ewheelerinc wrote:

Hello all,

Long time FreePBX user, first time posting!

We have a strange issue where Asterisk doesn’t see the ‘ext-vqueues’ context, but clearly it is available via show dialplan.

Yes, vqueues is commercial, but this doesn’t look like a module problem—it looks like asterisk is confused. Can someone tell what is going on here?

[...]
[2018-07-18 11:07:39] VERBOSE[9518][C-0000039f] file.c: <SIP/trunk-00000135> Playing 'cricket/robert_disc2.ulaw' (language 'en')
[2018-07-18 11:07:47] VERBOSE[9518][C-0000039f] pbx.c: Executing [1@ivr-6:1] Goto("SIP/trunk-00000135", "ext-vqueues,1,1") in new stack
[2018-07-18 11:07:47] VERBOSE[9518][C-0000039f] pbx_builtins.c: Goto (ext-vqueues,1,1)
**[2018-07-18 11:07:47] WARNING[9518][C-0000039f] pbx.c: Channel 'SIP/trunk-00000135' sent to invalid extension but no invalid handler: context,exten,priority=ext-vqueues,1,1**



cricket-acd-new*CLI> dialplan show ivr-6
[ Context 'ivr-6' created by 'pbx_config' ]
  '1' =>            1. Goto(ext-vqueues,1,1)                      [pbx_config]
  'h' =>            1. Hangup()                                   [pbx_config]
  'hang' =>         1. Playback(vm-goodbye)                       [pbx_config]
                    2. Hangup()                                   [pbx_config]
  'i' =>            1. Set(INVALID_LOOPCOUNT=$[${INVALID_LOOPCOUNT}+1]) [pbx_config]
                    2. GotoIf($[${INVALID_LOOPCOUNT} > 3]?final)  [pbx_config]
                    3. Set(IVR_MSG=no-valid-responce-pls-try-again) [pbx_config]
                    4. Goto(s,start)                              [pbx_config]
     [final]        5. Playback(no-valid-responce-transfering)    [pbx_config]
                    6. Goto(app-blackhole,hangup,1)               [pbx_config]
  'return' =>       1. Set(_IVR_CONTEXT=${CONTEXT})               [pbx_config]
                    2. Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT_${CONTEXT}}) [pbx_config]
                    3. Set(IVR_MSG=cricket/robert_disc2)     [pbx_config]
                    4. Goto(s,start)                              [pbx_config]
  's' =>            1. Set(TIMEOUT_LOOPCOUNT=0)                   [pbx_config]
                    2. Set(INVALID_LOOPCOUNT=0)                   [pbx_config]
                    3. Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT}) [pbx_config]
                    4. Set(_IVR_CONTEXT=${CONTEXT})               [pbx_config]
                    5. Set(__IVR_RETVM=)                          [pbx_config]
                    6. GotoIf($["${CDR(disposition)}" = "ANSWERED"]?skip) [pbx_config]
                    7. Answer()                                   [pbx_config]
                    8. Wait(1)                                    [pbx_config]
     [skip]         9. Set(IVR_MSG=cricket/robert_disc2)     [pbx_config]
     [start]        10. Set(TIMEOUT(digit)=3)                     [pbx_config]
                    11. ExecIf($["${IVR_MSG}" != ""]?Background(${IVR_MSG})) [pbx_config]
                    12. WaitExten(10,)                            [pbx_config]
  't' =>            1. Set(TIMEOUT_LOOPCOUNT=$[${TIMEOUT_LOOPCOUNT}+1]) [pbx_config]
                    2. GotoIf($[${TIMEOUT_LOOPCOUNT} > 3]?final)  [pbx_config]
                    3. Set(IVR_MSG=no-valid-responce-pls-try-again) [pbx_config]
                    4. Goto(s,start)                              [pbx_config]
     [final]        5. Playback(no-valid-responce-transfering)    [pbx_config]
                    6. Goto(app-blackhole,hangup,1)               [pbx_config]
  Include =>        'ivr-6-custom'                                [pbx_config]

But clearly vqueuews exists and has a 1,1 extension so the Goto should work:

cricket-acd-new*CLI> dialplan show ext-vqueues
[ Context 'ext-vqueues' created by 'pbx_config' ]
  '1' =>            1. Set(VQ_AANNOUNCE=cricket/afterdisc_cd) [pbx_config]
                    2. Goto(ext-queues,6501,1)                    [pbx_config]
  '2' =>            1. Set(VQ_AANNOUNCE=cricket/afterdisc_ssf) [pbx_config]
                    2. Goto(ext-queues,6505,1)                    [pbx_config]
  '3' =>            1. Set(VQ_AANNOUNCE=cricket/afterdisc_esp_cd) [pbx_config]
                    2. Goto(ext-queues,6504,1)                    [pbx_config]
  '4' =>            1. Set(VQ_AANNOUNCE=cricket/afterdisc_esp_ssf) [pbx_config]
                    2. Goto(ext-queues,6506,1)                    [pbx_config]
  Include =>        'ext-vqueues-custom'                          [pbx_config]

Ideas?

-Eric

Posts: 1

Participants: 1

Read full topic

@rubyred wrote:

Hi All,

I am wondering if there is a way to forward calls that are from a specific CID to a specific extension. There doesn’t seem to be any way to do this in the inbound route and I’m not sure where else to look.

Thanks!

Posts: 5

Participants: 4

Read full topic