Register Trunk for each FXS port on GXW4224?
@avayax wrote: I am setting up a trunk between a Grandstream GXW4224 FXS gateway and FreePBX.Everything works, but only if I register each FXS port separately on Asterisk with extension as the SIP...
View ArticleG722 an IAX2
@avayax wrote: Are there any concerns, problems with using G722 over IAX2 or is G711 the better choice here? Posts: 1 Participants: 1 Read full topic
View Article"WARNING,Friendly Scanner from IP address
@avayax wrote: I have SIP ports open to allow traffic from my SIP provider.I am getting Warning Friendly scanner messages from his IP address. What do they mean? -- Executing [s@from-trunk:3]...
View ArticleScripting auto cert install possible?
@mlaihk wrote: I am moving to Let's Encrypt and the Auto Let's Encrypt features in the freepbx 13 GUI is not use able for my application. So I have a server in charge to renew all the Let's Encrypt...
View ArticleConf. room fro every extension after update
@bazzacad wrote: My PBX had 22 module updates this more, so I apply them.After that it created a conf. room for each extension.It's done this once before & I deleted all of the Conf. rooms.Why...
View ArticleYank Back a Blind Transferred Call
@cyberdocwi wrote: Hello, I was asked by a user today on if they perform a blind transfer, and realize after the call "leaves", that they typed in the wrong extension, or perhaps suddenly remembered...
View ArticleNew install having parking lot problems, ParkPlus not installed but getting...
@woogieboogie wrote: Moved an office from Elastix to FreePBX. To be safe I didn't use any kind of backup or bulk extension imports and did all settings manually. After beginning to use the pbx,...
View ArticleHelp with Cisco 2811 as PRI to Sip Gateway
@voiptechny wrote: I am wondering if anybody here can help me with setting up the config file on a Cisco 2811 router to act as a PRI to Sip Gateway. I came across a few threads about this subject but...
View ArticleIVR DTMF not working "we have not received a valid response"
@avayax wrote: Getting we have not received a valid response on 80% of the time when I press a single digit to trigger an IVR option.4 digit direct extension dialing is fine, one digit not.What's...
View ArticleRecording Outbound Calls made via Originate - Context:from-internal
@FreeSWServers wrote: Hello! We currently have a web button that makes outbound calls via the follow code: Action: Originate Channel: Local/847 Context: from-internal Exten: 913478152765 Priority: 1...
View ArticleIs there a hold recall timer in the FreePBX?
@BCS_Inc_WORMA wrote: Customer places inbound call to line key on hold, states that the held call rings back too soon. Where can I change that timer? Posts: 1 Participants: 1 Read full topic
View ArticleClick to Dial
@dsabot wrote: I am looking for a solution that would allow me to click a phone number in a web browser and it would call that number. But rather than call the number using a softphone, it would...
View ArticleChanging vm-login file from "Comedian Mail, Mailbox" to "Mailbox"
@striker495 wrote: I have read a few post about changing the vm-login file to remove the "Comedian Mail" from the login of your mailbox. The issue that i'm hitting is that the file that looks to be...
View ArticleReports->AsteriskInfo->Peers show 192.168.1.2 for an extension - SOLVED
@solutions4smb wrote: I have a new installation 13.x with a single extension for an Aastra 6731i that is acting very weird. Extension shows registered on Aastra phone, PBX says UNREACHABLE but shows a...
View ArticleChan_sip.c:4263 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit...
@avayax wrote: Was configuring a SIP trunk to a grandstream gateway and I am now getting this constantly:chan_sip.c:4263 _sipreliable_xmit: Serious Network Trouble; _sipxmit returns error for pkt data...
View ArticleCalls from PSTN to Grandstream GXW4104 end with dial tone
@avayax wrote: Configuring a Grandstream GXW4104 PSTN Gateway with FreePBX and AT&T analog trunks.All is working except for inbound calls. They don't come into my inbound route but end with dial...
View ArticleBusy Lamp Field Howto
@strowgerswitch wrote: Noob here trying to get busy lamp field to work. Using Freepbx 13.0.190.15Does blf feature require end point manager. If not, what are the steps needed to activate blf in the...
View ArticleRinging ext 140
@ashthelefty wrote: Hi everyone, Having a bit of a weird problem here, People are trying to call EXT 140 on the freepbx server internally and they are getting put through to an outside number. I've...
View ArticleUnspecified sip users
@chasemixon wrote: A week ago, I changed an extension for one person's name to new person's name, after I saved it I saw an Unspecified sip connection in the Asterisk Info page and one sip user...
View ArticleUpgrading iSymphony fails
@antonny wrote: Hi all, I just upgraded two FreePBX systems to 10.13.66-18 using the upgrade script /usr/sbin/sysadmin_update_system. One system upgraded without error messages, on the second system...
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