# Uncategorized **URL:** https://community.asterisk.org/c/uncategorized/1.md [Latest](https://community.asterisk.org/latest.md) · [Categories](https://community.asterisk.org/categories.md) --- ## [Connect analog line to asterisk](https://community.asterisk.org/t/connect-analog-line-to-asterisk/66512)
**Author:** [@lb11](https://community.asterisk.org/u/lb11)\ **Replies:** 6\ **Last updated:** [April 29, 2016, 6:58am UTC](https://community.asterisk.org/t/connect-analog-line-to-asterisk/66512 "2016-04-29T06:58:15Z")
Hi, i'm using Asterisk for a long time now, but i want to connect my Netherlands analog phoneline to Asterisk. I mean, i want to connect my home analog line to Asterisk, so the users call via the analog line and not via … --- ## [\[SOLVED\] Centos 7 compatible init.d or systemd script for Asterisk 13](https://community.asterisk.org/t/solved-centos-7-compatible-init-d-or-systemd-script-for-asterisk-13/66359)
**Author:** [@Furious-George](https://community.asterisk.org/u/Furious-George)\ **Replies:** 6\ **Last updated:** [April 17, 2023, 2:35am UTC](https://community.asterisk.org/t/solved-centos-7-compatible-init-d-or-systemd-script-for-asterisk-13/66359 "2023-04-17T02:35:36Z")
It's just not working for Centos: # systemctl start asterisk Job for asterisk.service failed because a configured resource limit was exceeded. See "systemctl status asterisk.service" and "journalctl -xe" for details. \[… --- ## [Can I hide the clients number from users when using asterisk?](https://community.asterisk.org/t/can-i-hide-the-clients-number-from-users-when-using-asterisk/68198)
**Author:** [@henri12](https://community.asterisk.org/u/henri12)\ **Replies:** 17\ **Last updated:** [August 23, 2021, 7:40pm UTC](https://community.asterisk.org/t/can-i-hide-the-clients-number-from-users-when-using-asterisk/68198 "2021-08-23T19:40:46Z")
I have an asterisk system which I use to make outgoing calls. I've been trying to hide the number from users so that they click to call but without knowing the number they are calling. I have tried to base64 encode the n… --- ## [Can't Delete User/Ext](https://community.asterisk.org/t/cant-delete-user-ext/86907)
**Author:** [@rel82me](https://community.asterisk.org/u/rel82me)\ **Replies:** 1\ **Last updated:** [December 27, 2020, 12:04am UTC](https://community.asterisk.org/t/cant-delete-user-ext/86907 "2020-12-27T00:04:29Z")
Friends, I’m getting “Extension in use by another user” message when I try to recreate a deleted extension? Will I have to access the database and remove manually?? --- ## [Management Asterisk with Web panel or GUI](https://community.asterisk.org/t/management-asterisk-with-web-panel-or-gui/82902)
**Author:** [@rusakotov](https://community.asterisk.org/u/rusakotov)\ **Replies:** 1\ **Last updated:** [February 28, 2020, 4:56am UTC](https://community.asterisk.org/t/management-asterisk-with-web-panel-or-gui/82902 "2020-02-28T04:56:46Z")
Hello everybody I have set up Asterisk 16 on ubuntu 18.04 and I need to manage by web admin panel or GUI. But I can’t find any way. Please help me. Thanks --- ## [Memory issues after asterisk 15.7.2 update over the weekend](https://community.asterisk.org/t/memory-issues-after-asterisk-15-7-2-update-over-the-weekend/79091)
**Author:** [@JaredBusch](https://community.asterisk.org/u/JaredBusch)\ **Replies:** 4\ **Last updated:** [March 26, 2019, 1:37am UTC](https://community.asterisk.org/t/memory-issues-after-asterisk-15-7-2-update-over-the-weekend/79091 "2019-03-26T01:37:10Z")
I upgraded a couple FreePBX systems over the weekend from Asterisk 15.5.0 to 15.7.2 and had memory problems this morning. I reverted them to 15.5.0 (one via VM revert, the other via yum downgrade) and rebooted. No issue… --- ## [Welcome to the Asterisk Community](https://community.asterisk.org/t/welcome-to-the-asterisk-community/8)
**Author:** [@system](https://community.asterisk.org/u/system)\ **Replies:** 6\ **Last updated:** [January 24, 2019, 7:25am UTC](https://community.asterisk.org/t/welcome-to-the-asterisk-community/8 "2019-01-24T07:25:55Z")
Welcome to the new Asterisk Community! We've moved! Why? Because the old forums platform was long in the tooth, doesn't translate well to non-desktop devices, and isn't the easiest for you, the user, or for us, the … --- ## [Building dahdi on fedora22 and "certs/signing\_key.pem: No such file or directory"](https://community.asterisk.org/t/building-dahdi-on-fedora22-and-certs-signing-key-pem-no-such-file-or-directory/65540)
**Author:** [@gossamer](https://community.asterisk.org/u/gossamer)\ **Replies:** 4\ **Last updated:** [November 16, 2018, 8:26am UTC](https://community.asterisk.org/t/building-dahdi-on-fedora22-and-certs-signing-key-pem-no-such-file-or-directory/65540 "2018-11-16T08:26:50Z")
Hi, When downloading and building the latest dahdi-linux from git, it fails build due to a missing signing\_key.pem file, apparently included with the kernel? git clone git://git.asterisk.org/dahdi/linux dahdi-linux cd… --- ## [Audio goes blank for a few seconds during the conversation, and then comes back on](https://community.asterisk.org/t/audio-goes-blank-for-a-few-seconds-during-the-conversation-and-then-comes-back-on/67582)
**Author:** [@vijo.jose](https://community.asterisk.org/u/vijo.jose)\ **Replies:** 2\ **Last updated:** [August 3, 2016, 5:01am UTC](https://community.asterisk.org/t/audio-goes-blank-for-a-few-seconds-during-the-conversation-and-then-comes-back-on/67582 "2016-08-03T05:01:55Z")
audio goes blank for a few seconds during the conversation, and then comes back on in asterisk 1.4 with trunk as PRI line' with the sangoma PRI card. /etc/dahdi/system.conf Sangoma A101 port 1 \[slot:13 bus:3 span… --- ## [Match against multiple IP address in PJSIP?](https://community.asterisk.org/t/match-against-multiple-ip-address-in-pjsip/68139)
**Author:** [@phonefxg](https://community.asterisk.org/u/phonefxg)\ **Replies:** 27\ **Last updated:** [August 29, 2017, 1:50pm UTC](https://community.asterisk.org/t/match-against-multiple-ip-address-in-pjsip/68139 "2017-08-29T13:50:07Z")
Hi, Is there a way to match incoming calls to more than one ip adress for the same endpoint? My provider sometimes changes it's IP in the from user field and I need to allow another ip address for my endpoint. I have… --- ## [Web Request Missing Variable Is Null](https://community.asterisk.org/t/web-request-missing-variable-is-null/71096)
**Author:** [@strangewill](https://community.asterisk.org/u/strangewill)\ **Replies:** 2\ **Last updated:** [June 20, 2017, 12:48pm UTC](https://community.asterisk.org/t/web-request-missing-variable-is-null/71096 "2017-06-20T12:48:53Z")
This is actually a Switchvox question -- but I was directed to this being an Asterisk problem, so please excuse me if I'm making rough guesses at what is happening internally (I've configured bare Asterisk systems before… --- ## [Microsoft CRM Integration with AsteriskNow](https://community.asterisk.org/t/microsoft-crm-integration-with-asterisknow/67455)
**Author:** [@aegis](https://community.asterisk.org/u/aegis)\ **Replies:** 2\ **Last updated:** [July 21, 2016, 4:55pm UTC](https://community.asterisk.org/t/microsoft-crm-integration-with-asterisknow/67455 "2016-07-21T16:55:08Z")
Dear Team Is there any option/setting in AsteriskNOW application to link it with Microsoft CRM ? If yes pl let us know. --- ## [Need Ssh access to my Digium / Switch box system](https://community.asterisk.org/t/need-ssh-access-to-my-digium-switch-box-system/69630)
**Author:** [@PedroNL](https://community.asterisk.org/u/PedroNL)\ **Replies:** 2\ **Last updated:** [February 3, 2017, 3:20pm UTC](https://community.asterisk.org/t/need-ssh-access-to-my-digium-switch-box-system/69630 "2017-02-03T15:20:18Z")
I have a Digium Switchvox phone system is there a way for me to get SSH access to the system? --- ## [Modern 3G IP Radio PTT server](https://community.asterisk.org/t/modern-3g-ip-radio-ptt-server/68905)
**Author:** [@Bradlington](https://community.asterisk.org/u/Bradlington)\ **Replies:** 0\ **Last updated:** [November 28, 2016, 7:22pm UTC](https://community.asterisk.org/t/modern-3g-ip-radio-ptt-server/68905 "2016-11-28T19:22:05Z")
I would like to experiment to have and test to learn more about Asterix. The modern 3G IP Two way radios using POC /PTT over cellular. How does one start making and more importantly learning the architecture to run s… --- ## [Digium API - is there "hook up" event?](https://community.asterisk.org/t/digium-api-is-there-hook-up-event/68741)
**Author:** [@Mans7](https://community.asterisk.org/u/Mans7)\ **Replies:** 0\ **Last updated:** [November 14, 2016, 5:38am UTC](https://community.asterisk.org/t/digium-api-is-there-hook-up-event/68741 "2016-11-14T05:38:11Z")
Hello! I have Digium D50 and developing js application to control all necessary functions of phone. And I need now handle "hook up" event. As we know, the telephone has no lever under phone tube. It works on magnetic or … --- ## [Increase duration of DTMF tones on Digium G200 gateways](https://community.asterisk.org/t/increase-duration-of-dtmf-tones-on-digium-g200-gateways/68262)
**Author:** [@avayax](https://community.asterisk.org/u/avayax)\ **Replies:** 1\ **Last updated:** [October 6, 2016, 5:44pm UTC](https://community.asterisk.org/t/increase-duration-of-dtmf-tones-on-digium-g200-gateways/68262 "2016-10-06T17:44:51Z")
Is there a way to increase the duration of dtmf tones on Digium G200 gateways? --- ## [Timeout function of function Background isn't working](https://community.asterisk.org/t/timeout-function-of-function-background-isnt-working/68193)
**Author:** [@humber2](https://community.asterisk.org/u/humber2)\ **Replies:** 7\ **Last updated:** [October 3, 2016, 4:45pm UTC](https://community.asterisk.org/t/timeout-function-of-function-background-isnt-working/68193 "2016-10-03T16:45:39Z")
When We dial the number 211024067, we've three options: Pressing 1 - The cal will be recorded - It's working Pressing 2 - The call won't be recorded - It's working Pressing nothing - It's supposed to hve the call disc… --- ## [User Input and auto Dial](https://community.asterisk.org/t/user-input-and-auto-dial/68223)
**Author:** [@md.ehsan](https://community.asterisk.org/u/md.ehsan)\ **Replies:** 2\ **Last updated:** [October 1, 2016, 3:50pm UTC](https://community.asterisk.org/t/user-input-and-auto-dial/68223 "2016-10-01T15:50:07Z")
How can we accept input from caller e.g mobile no. id no. and name and save it to asterisk database and schedule a auto dial to this saved mobile no. and play a recorded message after called party answer the auto dial. … --- ## [Is there a way to get the sound level from a caller?](https://community.asterisk.org/t/is-there-a-way-to-get-the-sound-level-from-a-caller/68201)
**Author:** [@djc](https://community.asterisk.org/u/djc)\ **Replies:** 1\ **Last updated:** [September 28, 2016, 8:05pm UTC](https://community.asterisk.org/t/is-there-a-way-to-get-the-sound-level-from-a-caller/68201 "2016-09-28T20:05:13Z")
Just wondering if for the information you can get for a conference room of each caller (ID, Caller ID, mode, etc).... is there any ability to get the current audio level for a specific caller in the call (ie, I want to k… --- ## [Same line input and output](https://community.asterisk.org/t/same-line-input-and-output/68192)
**Author:** [@raquelita](https://community.asterisk.org/u/raquelita)\ **Replies:** 6\ **Last updated:** [September 29, 2016, 12:52am UTC](https://community.asterisk.org/t/same-line-input-and-output/68192 "2016-09-29T00:52:15Z")
Good afternoon, I installed Asterisk on a server and want to know if there is any way to set it up so you can use the same input line as one of the outputs. Thanks in advance A greeting --- ## [Asterisk SIP Integration with Avaya SES](https://community.asterisk.org/t/asterisk-sip-integration-with-avaya-ses/68167)
**Author:** [@arshad.hussain](https://community.asterisk.org/u/arshad.hussain)\ **Replies:** 15\ **Last updated:** [September 28, 2016, 4:12pm UTC](https://community.asterisk.org/t/asterisk-sip-integration-with-avaya-ses/68167 "2016-09-28T16:12:04Z")
Dear All expert, When i am trying outbound call from my asterisk to Avaya SES server between sip trunking between both system so i am facing the following issue. -- Called IPTEL/4545 -- Got SIP response 483 "Too Many … --- ## [Monitor picked call](https://community.asterisk.org/t/monitor-picked-call/68169)
**Author:** [@asterisker](https://community.asterisk.org/u/asterisker)\ **Replies:** 9\ **Last updated:** [September 27, 2016, 2:06pm UTC](https://community.asterisk.org/t/monitor-picked-call/68169 "2016-09-27T14:06:34Z")
Is it possible? Without one touch record, but automatically? --- ## [Handle\_response\_invite: Failed to authenticate on INVITE to '"](https://community.asterisk.org/t/handle-response-invite-failed-to-authenticate-on-invite-to/68138)
**Author:** [@humber2](https://community.asterisk.org/u/humber2)\ **Replies:** 12\ **Last updated:** [September 26, 2016, 12:17pm UTC](https://community.asterisk.org/t/handle-response-invite-failed-to-authenticate-on-invite-to/68138 "2016-09-26T12:17:31Z")
I can´t establish outgoing calls from an Asterisk to other using the SIP Trunk - GoVOIPGate. Both servers belong to me. The caller extension is the 28814 and the called one is the 37971. The error I've got is the followi… --- ## [Call file playback executes before the person answer the call](https://community.asterisk.org/t/call-file-playback-executes-before-the-person-answer-the-call/68092)
**Author:** [@pooriaG](https://community.asterisk.org/u/pooriaG)\ **Replies:** 6\ **Last updated:** [September 26, 2016, 12:11pm UTC](https://community.asterisk.org/t/call-file-playback-executes-before-the-person-answer-the-call/68092 "2016-09-26T12:11:25Z")
hi i have a call file as below. it sends the call to context ----Channel: DAHDI/g0/909133 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: pooria Extension: 21 \[pooria\] exten =\> 21,1,Answer() exten =\> 21,n,Wait(5) e… --- ## [Open web page specfic to call](https://community.asterisk.org/t/open-web-page-specfic-to-call/68134)
**Author:** [@hypuk](https://community.asterisk.org/u/hypuk)\ **Replies:** 2\ **Last updated:** [September 22, 2016, 1:26pm UTC](https://community.asterisk.org/t/open-web-page-specfic-to-call/68134 "2016-09-22T13:26:11Z")
Hi is it possible for Asterisk to open a specific web page when a call comes in relating to that call? EG if ABC Company calls, then Asterisk opens a web page specfic to ABC company? Any help would be much apprecia… --- ## [How to Budge into a call with an audio file every 30 seconds](https://community.asterisk.org/t/how-to-budge-into-a-call-with-an-audio-file-every-30-seconds/68095)
**Author:** [@kevinade](https://community.asterisk.org/u/kevinade)\ **Replies:** 9\ **Last updated:** [September 20, 2016, 5:26am UTC](https://community.asterisk.org/t/how-to-budge-into-a-call-with-an-audio-file-every-30-seconds/68095 "2016-09-20T05:26:51Z")
Hi All, I would like to know how to play an audio file every 30 seconds between participants in a call on Asterisk 13? Ideally I would like this to apply to every call on both the caller and the individual being call… --- ## [Level 3 Enterprise Voice](https://community.asterisk.org/t/level-3-enterprise-voice/68112)
**Author:** [@TommyFrass1](https://community.asterisk.org/u/TommyFrass1)\ **Replies:** 0\ **Last updated:** [September 20, 2016, 12:27am UTC](https://community.asterisk.org/t/level-3-enterprise-voice/68112 "2016-09-20T00:27:09Z")
Has anyone had the pleasure of setting this up in their PBX? I have been waiting on documentation from them but in the mean time I thought I would post and see if anyone has set this up and if so what changes might be re… --- ## [V5.2 signalling](https://community.asterisk.org/t/v5-2-signalling/68059)
**Author:** [@behnamfahimi](https://community.asterisk.org/u/behnamfahimi)\ **Replies:** 6\ **Last updated:** [September 16, 2016, 5:51pm UTC](https://community.asterisk.org/t/v5-2-signalling/68059 "2016-09-16T17:51:53Z")
Hi dears . I need Asterisk channel driver for v5.2 on dahdi/zaptel E1 interfaces. Thanks. --- ## [TCP with PJSIP - Contact Header in Outbound Invite Message](https://community.asterisk.org/t/tcp-with-pjsip-contact-header-in-outbound-invite-message/68065)
**Author:** [@sameersdhar](https://community.asterisk.org/u/sameersdhar)\ **Replies:** 3\ **Last updated:** [September 16, 2016, 12:19pm UTC](https://community.asterisk.org/t/tcp-with-pjsip-contact-header-in-outbound-invite-message/68065 "2016-09-16T12:19:28Z")
Hi, I am facing an issue where using TCP with an endpoint causes to have random port in the Contact header when it send out an invite to the endpoint. It should be what is it listening on (5060): Contact: It shou… --- ## [Shell script disobeying phpagi call flow](https://community.asterisk.org/t/shell-script-disobeying-phpagi-call-flow/68041)
**Author:** [@senjam](https://community.asterisk.org/u/senjam)\ **Replies:** 4\ **Last updated:** [September 14, 2016, 10:18pm UTC](https://community.asterisk.org/t/shell-script-disobeying-phpagi-call-flow/68041 "2016-09-14T22:18:14Z")
I have a system where the call lands on a call flow script which is a php script that uses phpagi. The call flow script calls a shell script which uses sox to concate two real time generated audio file by the call flow. … [Next page](https://community.asterisk.org/c/uncategorized/1.md?page=1)