Call recording not working on a conference bridge
@infotek wrote: I have two conference bridges one will record and the other will not. The one that will record: 26639998The one that will record has "Record Conference" = no.[2016-05-27 15:32:51]...
View ArticleHow do you *use* Follow Me
@kbocek wrote: I've got the Follow Me module installed and activated on a couple of extensions. I see two different wiki pages on configuring it. I see *21 for activating and deactivating it. But once...
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Gui Whoops error on VoiceMail Admin
@avayax wrote: Getting a Whoops Errors on the GUI when accessing VoiceMail Admin.When I try to remove or copy that msg0034.gsm file (also .txt, .wav and .WAV) from command line, I get a "Input/output...
View ArticleOutbound trunk xs4all not working
@basd82 wrote: I have problem i have sip connection from xs4all. All inbound calls are working without any problem.But what is not working is the Outgoing trunk i get errors. My trunk settings look...
View ArticleInbound Routes, DID field expects numbers, Provider sends letters
@travisdietrich wrote: Hello, I either have something set wrong, or my provider is doing something non-standard. Up until just now, I've had no issues with my providers calls. However, up until now I...
View ArticleThought i could help with setup a Spa8800 with pjsip
@Henry_power wrote: Thought i could help ******** FreePBXGeneral : Trunk Name = spa8k8line1........ "same as User ID:" Outbound CallerID : 7111........ "same as Dial Plan 8:" DID value PJSIP Settings:...
View ArticleHow to send dtmf in disa
@voiptec wrote: hello i want to ask questioni have axsas number when i dial this number i put 9 and the numberand i want my freepbx call the number automatik 9 and the number when i press it in...
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Is suffixing number with # on UCP Follow me list necessary?
@avayax wrote: The description for the follow me list in UCP says I need to suffix a number on a remote system with pound #. Now all of my numbers on remote systems work with and without #, so when is...
View ArticleWhy/When did Polcom phone AutoAnswer change again?
@wpns wrote: So I've got: voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="ringAutoAnswer" voIpProt.SIP.alertInfo.2.value="Alert-Info: Auto Answer"...
View ArticlePacket transmission time
@Novais wrote: I have been having an issue with dropped calls with errors showing in the logs of "timeout on non-critical invite" which then drops the call. The provider has asked that we check on the...
View ArticleOne provider, 1 trunk programmed, 2 trunks online
@Novais wrote: Our PBX is programmed with one trunk from one provider with 6 channels but when looking at the dashboard it always shows 2 trunks online. Is this inbound and outbound or.... The other...
View ArticleTrying to do trunk to trunk transfer
@pantsonfire wrote: First post, little FreePBX experience, thanks for your patience. I will try to describe what I am attempting. I have two Asterisk servers connected with an AIX2 trunk. Server B is...
View ArticleMail not sended from dialplan
@maxi4 wrote: i use a hook in dialplan to send email in non-answer case: [root@freepbx asterisk]# cat extensions_override_freepbx.conf [sub-missedcallnotify] exten =>...
View ArticleCall Recording for All calls
@psdk wrote: Hi, Is it possible to enable call recording for all calls ? I know that we can enable call recording for extensions or using call recording module and define a routing for trunks and...
View ArticleInbound Route for Unknown Caller ID
@jimf wrote: Hello I'd like to create an inbound route for an unknown/blank caller ID (defined in call event logging as Unknown <>) to be sent to a particular destination. I'm having difficulty...
View ArticleVoice mail waiting indicator keep blinking even though no new message
@sumitk wrote: We are using Grandstream GXP1625 IP phones with FreePBX 12.0.76.2.Voice mail waiting indicator keep blinking even though no new message but after deleting all old message it stop...
View ArticleAsterisk 11.22.0 - major Queue Bug causes Segfault
@GSnover wrote: https://issues.asterisk.org/jira/browse/ASTERISK-25888 If you have a lot of Queues, avoid this version until the issue is fixed - switched a Cluster over the weekend and as soon as...
View ArticleHow to make a call coming in on a specific DID show on the phone differently
@ilovetiffany wrote: We are setting up a FreePBX system using the latest build and Grandstream GXP2170 phones. We have two companies and we have to answer the phone differently depending on which DID...
View ArticlePage group with Softphone?
@tom5870 wrote: I am attempting to use a Softphone (written by our company using an SDK), and I setup the extension to be part of the paging group. When I dial 8000, the phone that dial gets a tone...
View ArticleCan I pass parameters on Asterisk ARI using appArgs
@rsvmt wrote: I'm writing a Stasis app on Node.js, can I pass parameters to the stasis app using the POST /channels?app=AppName&appArgs=xxxxxxx=30&api_key=abc:122 ? If so how can I parameters...
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