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Please help - NO outbound calls / inbound calls are fine - callcentric

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@Neromarketing18 wrote:

I have been struggling with inbound/outbound routes I finally have the inbound routes working and the outbound states “cal cannot be completed as dialed”

working with freepbx 14

peer details:

context=from-pstn-toheader
fromdomain=callcentric.com
fromuser=17778387121
host=callcentric.com
insecure=port,invite
secret=hello1
type=peer
defaultuser=17778387121
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

extensions_custom.conf
[incoming]
exten => s,1,Set(Var_FROM_DOMAIN=${CUT(CUT(SIP_HEADER(TO),@,2),>,1)})
exten => s,2,GotoIF($["${Var_FROM_DOMAIN}" = “callcentric.com”]?5:3)
exten => s,3,GotoIF($["${Var_FROM_DOMAIN}" = “ss.callcentric.com”]?5:4)
exten => s,4,GotoIF($["${Var_FROM_DOMAIN}" = “66.193.176.35”]?5:7)
exten => s,5,Set(Var_TO_DID=${CUT(CUT(SIP_HEADER(TO),@,1),:,2)})
exten => s,6,GotoIF($["${Var_TO_DID}" != “”]?ext-did,${Var_TO_DID},1:7)
; Users may edit the lines below to route incoming calls to other locations/contexts.
; If you don’t know what this means then you should likely skip the lines below and
; allow the script to run unmodified
exten => s,7,GoTo(from-pstn,s,1)
exten => h,8,Playback(ss-noservice)
exten => h,9,Macro(hangupcall)

[100]
context=to-callcentric
type=friend
defaultuser=100
secret=PASSWORD
host=dynamic

[101]
context=to-callcentric
type=friend
defaultuser=101
secret=PASSWORD
host=dynamic

[to-callcentric]
exten => _X.,1,Dial(SIP/${EXTEN}@callcentric)

any help would be greatly appreciated

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Misc destination sometimes no sound

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@nxswart wrote:

Hello!

Hope you guys can help me, I have a strange problem.
Where using FreePBX 2.11.0.43 on a ubuntu server.
We created a IVR and if the caller press button x trough a misc destination we call a external number.
In some cases we external number hears the caller but in some cases it doesn’t.
In all cases the caller hears sound from the the one on the external number.

Any idea what could cause this?

Hope to hear from you.

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Enable SRV

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@anty506 wrote:

How do I enable SRV in FreePBX for my Trunk? I’m not able to dial out unless we put A records in place.

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Tranferring calls does not complete

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@cooki3monst3r77 wrote:

When someone calls into the PBX, the call is answered and transferred to another extension. The user on the other extension hears the phone ring, and picks up the phone, but the call transfer doesn’t appear to complete. The caller that is being transferred only hears MOH forever, until they disconnect.

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VOIP call to any cellular sim

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@murphy_joy wrote:

Dear all
good morning
call from one extension to another extension is going very well but
i want to do call from SIP extension to any cellular sim.
i dont have any idea how to configure this scenraio.
i am using dinstar gsm gateway, android application is zoiper and freepbx 13.
please help me as soon as possible

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What language do you prefer to develop ARI

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@james wrote:

Hello:
we want to use ARI to develop some Apps, there are some ARI libraries from ARI Libraries.
we want to use PHP, but it looks PHP libraries not very active. If we develop Apps based on FreePBX, what language do you prefer? PHP or others?
thanks!

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TEL URI - SIP/2.0 416 Unsupported URI Scheme

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@mveselic wrote:

Hi,

is there support for TEL URI in FreePBX? I have PJSIP trunk to ITSP, everything is OK, except inbound DID routing not working.

<--- Transmitting SIP response (497 bytes) to UDP:10.252.64.110:5060 --->
SIP/2.0 416 Unsupported URI Scheme
Via: SIP/2.0/UDP 10.252.64.110:5060;received=10.252.64.110;branch=z9hG4bKm7b1zuqu1qrt1plqp1zomgzmqT36378
Record-Route: <sip:10.252.64.110:5060;transport=udp;lr>
Call-ID: asbcBW1630168621010181045472166@10.10.10.4
From: <sip:0xxxxxxxx@10.252.64.110;user=phone>;tag=sbc0404795406886-1539181816862-
To: "xxxx" <tel:+387xxxxxx>;tag=z9hG4bKm7b1zuqu1qrt1plqp1zomgzmqT36378
CSeq: 791762448 INVITE
Server: FPBX-14.0.3.19(13.22.0)
Content-Length:  0**strong text**

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Freepbx losing VLAN

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@mhessels wrote:

Hey everyone

I have a fairly new setup (2 months old) that I deployed. Over the course of two months I have had to go back several times because the phone system was not working ( no incoming or outgoing calls). When I log into the system the vlan is not responding. After I delete the interface and rebuild it, it usually works.

I have looked into network settings, firewall settings and asterisk sip settings but I can’t find any definitive answer.

Could it be hardware?

Thanks for the help!

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Conditional Calling - Calling Operator

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@Ljrendon wrote:

I have several locations that want to be able to dial zero and have the call directed to the front office of the location zero is dialed from. Each location has its own caller ID so that way I can find a way to route the call to the correct Operator.
I am planning on creating ring groups for each location but then add a macro that essentially checks that if someone dials zero (0) and the caller ID is matched then the call is directed to a particular ring group.

This is what I have thought of but does not work. I am not experience I am adding this to the extensions_custom.conf file:

[macro-Operator]
;;User dialing zero is redirected to Operator A
exten => s,1,GotoIf($["${OUTNUM}"=“0” & “${CALLERID(number)}”=“XXXXXXXXXX”]?2:4)
exten => s,2,Dial(“Local/7151@from-internal”)
exten => s,3,MacroExit()
;;User dialing zero is redirected to Operator B
exten => s,4,GotoIf($["${OUTNUM}"=“0” & “${CALLERID(number)}”=“YYYYYYYYYY”]?5)
exten => s,5,Dial(SIP/7152)
exten => s,6,MacroExit()

Currently I dial zero but nothing happens. Is this possible? Any help or pointers are appreciated. Thank you.

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Character Limit for Screen Name on Mitel Phones

User Manager - Can't locate way to add items

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@JessicaRabbit wrote:

No add feature in user management (14.0.3.19)? Wiki mentions an “Add” button, none to be found under Users or Groups.

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Emergency 911 CID For Extensions Outside Primary Office

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@bluehive wrote:

Our office primary DID is configured within our SIP provider as the Emergency CID so if someone in our office at an extension dials 911, it transmits our primary DID as the CID which matches the e911 address on file with our SIP provider. That works just swell.

However we have 3 employees who work from home and have a SIP phone in their home office. Each of the 3 outside the office extensions has their own DID assigned to them from our SIP provider.

If one of those 3 employees calls 911 from their company SIP phone, we need to make sure the 911 operator is getting their home address and NOT our primary office address.

We have our SIP trunk setup within FreePBX to use our primary office DID as the CID for the trunk - +15554443333

I noticed in the (?) under General > CID Options for the trunk that it says

“Determines what CIDs will be allowed out this trunk. IMPORTANT: EMERGENCY CIDs defined on an extension/device will ALWAYS be used if this trunk is part of an EMERGENCY Route regardless of these settings.”

Since we have each of our offsite 3 employees extensions setup with their own DID, do we simply need to register their home addresses with our SIP provider for emergencies and once done, if they call 911 or 933, their extensions CID will be transmitted through the trunk in which their extensions are assigned to? Basically overriding the CID set at the trunk level. 911 will see their CID and the primary office CID. I just want to make sure I’m understanding what the (?) tip said correctly.

Thank you!

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New Errors i never saw before

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@josephchrz wrote:

Hello i got some new error watching in putty. when i tried to call in i get this error i see.

[2018-10-15 13:25:50] WARNING[14176][C-000000a0]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2018-10-15 13:25:50] WARNING[14176][C-000000a0]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2018-10-15 13:25:50] WARNING[14176][C-000000a0]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2018-10-15 13:25:50] WARNING[14176][C-000000a0]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2018-10-15 13:25:52] WARNING[2642][C-000000a0]: chan_sip.c:10116 process_sdp: Ignoring video stream offer because port number is zero
[2018-10-15 13:25:56] WARNING[3049][C-000000a0]: format_wav.c:146 check_header: Read failed (type)
[2018-10-15 13:25:56] WARNING[3049][C-000000a0]: file.c:399 fn_wrapper: Unable to open format wav
[2018-10-15 13:25:56] WARNING[3049][C-000000a0]: res_musiconhold.c:339 ast_moh_files_next: Unable to open file ‘/var/lib/asterisk/moh/.nomusic_reserved/silence’: No such file or directory
[2018-10-15 13:26:01] WARNING[14176][C-000000a0]: chan_sip.c:22159 func_header_read: This function can only be used on SIP channels.
[2018-10-15 13:26:01] WARNING[14176][C-000000a0]: chan_sip.c:22159 func_header_read: This function can only be used on SIP channels.
[2018-10-15 13:26:01] WARNING[14176][C-000000a0]: chan_sip.c:22159 func_header_read: This function can only be used on SIP channels.
[2018-10-15 13:26:01] WARNING[14176][C-000000a0]: chan_sip.c:22159 func_header_read: This function can only be used on SIP channels.
[2018-10-15 13:26:09] WARNING[2642][C-000000a0]: chan_sip.c:10116 process_sdp: Ignoring video stream offer because port number is zero
[2018-10-15 13:26:09] WARNING[14156][C-0000009f]: format_wav.c:146 check_header: Read failed (type)
[2018-10-15 13:26:09] WARNING[14156][C-0000009f]: file.c:399 fn_wrapper: Unable to open format wav
[2018-10-15 13:26:09] WARNING[14156][C-0000009f]: res_musiconhold.c:339 ast_moh_files_next: Unable to open file ‘/var/lib/asterisk/moh/.nomusic_reserved/silence’: No such file or directory
[2018-10-15 13:26:58] WARNING[14228][C-000000a3]: func_channel.c:538 func_channel_read: Unknown or unavailable item requested: ‘reversecharge’

Not sure what this is. Can someone please help me?

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GLIBC Update to 2.19?

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@mbristol wrote:

We have a customer that prefers we make daily CDR exports to a Dropbox folder. About a year ago we installed the Linux Dropbox client on our Sangoma 7 Distro machine and all has been working well. As of today, Dropbox no longer supports (among other things) glibc < 2.19. Our current updated distro shows glibc = 2.17.

Any idea on when and/or if the Sangoma repo will be updated to glibc 2.19?

Thanks!

Mike

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Two emails for 1 VM

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@dannyprecise wrote:

I have it set to delete the VM and attach it as an email attachment.

The user is getting two emails.

1 to let them know they received an email
1 to provide them with the attachment.

How can I turn off the notification of the email since they will get that with the attachment anyways

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GoİP8 VOİP goes down , after 30-40 minutef of reboot

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@caqa wrote:

)

hi I have problem with GoIp8 Voip status Goes Down after approximately 30-40 minutes after reboot.
And in freePBX registry is shown request sent .Only fix this I have to reboot GoiP8 and this fixing the problem only 30-40 minutes any solutions?

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Ports to be opened?

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@josephchrz wrote:

Hello i was wondering what ports do i need to be open. I have a tftp server on my freebpx and not sure what port needs to be for freepbx tcp or udp?

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Blacklist Upgrades - Comments

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@JessicaRabbit wrote:

The usefulness of black lists has suffered as spamers have turned to spoofing and rapid turnover of DIDs, but I think the availability of black and white list are still a good idea. To be really a useful tool, the implementation needs to go beyond the front end list checking and the ability to add items and needs to provide list management functions. To do this, the list database must be able to capture data like number of hits and last hit date. The list management interface then needs to be able to do things like sort by column and delete by criteria. Otherwise the list just gets bigger and there no basis for managing it.
I have posted these issues previously but I just thought I would refresh the topic for those who might be considering upgrades to the module.

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Call audio cutting in and out on grandstream fxo gateway

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@josephchrz wrote:

Hello everything seems to be working for almost a year now. The freepbx system is using a 4 channel fxo gateway with 4 hardware lines from the phone company. Just recently the audio keeps cutting in and out for unknown reasons. I don’t know how to check it and see what is wrong if it’s the freepbx server or the Grand stream box. all phones are Cisco sip ip phones. Doesn’t matter if it’s just one call or four calls it just keeps cutting in and out when talking and listening. Can someone please help me.

I’m not sure what other information to give please forgive me on that.

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Ring Groups inside a ring group?

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@edricksmith wrote:

I have a setup where they would like an inbound route to ring a group of phones for 30 seconds or so, then if no answer ring another group of phones.

Is this possible?

Scenario:

Main Business Phones ring 30 seconds
If no answer then ring Main Business Phones + Remote Business Phones

or

Main Business Phones ring 30 seconds
If no answer then Ring Remote Business Phones

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