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UCP: Name in SMS history
@StephanK wrote: Hi, We have a lot of contacts in the UCP. In the SMS history we used to see the contact name, since the upgrade to FreePBX 13 we only see the number.(This is why we have all the...
View ArticleGrandstream Phones with the built in OpenVPN Client
@alpha202bd wrote: Greetings! Just wanted to check with the community here and see if anyone has had success with using any of the Grandstream GXP-21XX series phone's OpenVPN clients to connect to...
View ArticleUCP Contact Syncing with Exchange
@StephanK wrote: In the "manual" for the User Control Panel-Contact Directory it sais: Syncing with ExchangeYou can also optionally sync in your Contacts from Exchange. Your system administrator will...
View ArticleTLS and SRTP on FreePBX 13.0.190.2
@amin1356 wrote: Hi friends,I tried to implement TLS and SRTP on FreePBX Distro 10.13.66 (FreePBX 13.0.190.2) using the following help:wiki.freepbx.org/display/PHON/TLS+and+SRTP In "Chan SIP Settings"...
View ArticleYealink W52P One Way Audio on Outgoing DAHDI Calls
@jimf wrote: Hi. I've added two Yealink W52P phones to my system (FreePBX 13.0.190.2) today, but I'm having audio issues on outgoing DAHDI calls. With internal calls, audio works fine both ways. When...
View ArticleParking Pro
@Pandera wrote: Hi guysYesterday i bought Parking pro and i accidentally putted it on the wrong deployment I asked the sales team for assistance but they are just rode and not assisting me very much....
View ArticleMOH upload fail
@mvogel4949 wrote: I'm trying to upload a MOH file and keep getting the following error: Symfony\Component\Process\Exception\ProcessTimedOutExceptionThe process "/usr/sbin/asterisk -rx 'file convert...
View ArticleDial twice to complete outbound call on some phones
@thatchmaster wrote: Environment:* FreePBX 13* Avaya 9650C* Single network for phones/FreePBX A couple phones will fail the first outbound (internal or external) attempt with a fast busy. If the call...
View ArticlePEER SIP Trunk Failing
@comtech wrote: Asterisk: 13.7.2FreePBX: 13.0.188.9 I have my FreePBX appliance connected to my Avaya Session manager via SIP as a trunk for outgoing call processing. It works great, except I have had...
View ArticleOld fashion man using freepbx 2.9.0.12
@Tim75 wrote: I'm working with Freepbx 2.9.0.12 for many year and everything was going well until tuesday.For the firstime i got the following error message in the GUI: retrieve_conf failed, config...
View ArticleI manually copied SIP config files to system after I broke it
@dsbrown wrote: After troubleshooting NAT issues, I made a huge mistake and broke FreePBX. I compounded it by copying all the SIP configuration files that I manually backed up prior to my goof.As this...
View ArticleOutbound call recording
@alexcal wrote: Hi, I am unable to get outbound calls to record, inbound calls record without problem. call recording is enable in extensions and outbound route, also tried setting to force but to no...
View ArticleLumenVox/UniMRCP
@dave69s wrote: Has anyone integrated with LumenVox on their FreePBX system? I'm working on configuring the UniMRCP client, but it will not build because it cannot find the Asterisk source. Posts: 2...
View ArticleAutomating mp3 select during freepbx 13 install
@neuronetv wrote: following the instructions here:http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7I'm working on a script that will install freepbx automatically in centos 7. When...
View ArticleCant register SPA501G Cisco Phones
@newbefreepbxuser wrote: After some update of my provider, I can't register my phones (SPA501G) over my ipsec tunnels. They have been working for YEARS.I tried firmware update of ip phones....same...
View ArticleEndpoint Presense
@nickb wrote: I'm having an issue with a site where we've deployed about a dozen extensions, using the latest version of FreePBX. The endpoints are SPA514G, and I've used the OSS endpoint manager. The...
View ArticleVery odd one way audio connection
@mvogel4949 wrote: I have a remote phone off a system. I can call to other extensions with two way audio. Extensions that are local can call out and receive calls with audio. However, if I try to call...
View ArticleTrunk to Trunk Extension SIP 401 Unauthorized
@Summer wrote: Dear Sirs, PBX A - extensions with 1XX10.7.208.245 freepbx 6 with Asterisk 11.20.0||OpenVPN tunnel 10.7.218.3 gw|PBX B - extensions with 2XX10.7.218.245 Freepbx 10.13.66-11 Asterisk...
View ArticlePrepend digits to outbound DID
@LTAdmin wrote: I have an issue where I have to prepend 6 digits to my extensions for a certain SIP. So if I'm calling from 2579 I need 700417 to be added in front of that to be sent out the trunk so...
View ArticleRecommendations: SIP Trunking provider for Mexico DIDs
@scurry7 wrote: Does anyone have good or bad experience with porting Mexico DIDs to a provider? Which Mexico provider would you recommend/not recommend? Thanks in advance! Posts: 2 Participants: 2...
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