Questions tagged [asterisk]

Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

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Loss of SIP Header Values ​on Transfer

Closed. This question is not about programming or software development. It is not currently accepting answers. This question does not appear to be about a specific programming problem, a software ...
Mustafa Furkan Bilen's user avatar
-1 votes
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51 views

OpenVPN with static route

I purchased SIP trunk from a provider, along with SIP trunk details they provided a .ovpn file stating that the SIP will only connect with the VPN on the same network. On VPN connection it creates ...
Mylsel's user avatar
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Issue with Asterisk SIP Trunk: Payload Must Originate from Specific IP in Multi-Cloud Setup

I'm facing an issue with an Asterisk SIP trunk setup involving multiple cloud environments and a Session Border Controller (SBC). Our service provider, Airtel, requires that the SIP payload originates ...
George Muddu's user avatar
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90 views

Create Asterisk config

I have to build Asterisk on my own on Debian 12. As part of the installation, Asterisk offers to generate a large amount of long config files in /etc/asterisk, incluing an example PBX scenario (e.g. ...
MrSnrub's user avatar
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How to apply multiple filters in asterisk 20.4.0 AMI interface

I just want to filter all event that contains "Event: Hangup and "Channel: PJSIP/ANCONA_UFFICIO" against an event like this: Event: Hangup Privilege: call,all Channel: PJSIP/...
silvered.dragon's user avatar
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77 views

Setting the Display Name with Asterisk MessageSend

I am setting up Asterisk to connect a few phones / sip-devices on my LAN. I am able to send messages to each device via the MessageSend() application in the extensions.conf file : [...] same => n,...
Janthelme's user avatar
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1 answer
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How to recover/reinstall Grandstream GXP phones firmware?

I bought a Grandstream IP phone recently and tried to go through first installation, but suddenly most of SIP settings and also settings of firmware update server disappeared and I even can't login to ...
Phree's user avatar
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unable to recieve sip call from external source on fusionpbx, error seems to be [WRONG_CALL_STATE] . how can i fix it

Newbie here, I havejustall installed Fusionpbx in my debian based 1gb ram 40gb ssd server.I did also create several extensions but the calls from external sources are getting hanged up. I looked up ...
Dhirendra Biswal's user avatar
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49 views

Asterisk server shows peer UNREACHABLE for Kamailio sipper server. How do I diagnose the issue?

Presently I have 4 sippers, 2 are working using OpenSer and 2 are not working using Kamailio. I have inherited these servers. There is an Asterisk server. When I run sip show peer SIP_SER against ...
Jay Ryan's user avatar
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224 views

Can't configure Inbound/Outbound on Issabel

I have PRI line connected to E1 card. I installed Issabel with Asterisk11 on it. Whenever I try to call the number given to us by the PRI provider, I find the number busy. I tried to configure inbound ...
idk's user avatar
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Postfix using asterisk user to send emails. Mapping did not work, are there any other options?

I am trying to set up a PBX server to send voicemails to emails, which is being triggered properly, however, the from email is not the correct one I want to use. Instead of using [email protected], ...
Ceyntas's user avatar
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89 views

Setup Kamailio to accept only register request with FQDN and block IP request

I have set up Kamailio and am quite new to the process, I'm able to register with FQDN and IP directly to Kamailio but my main setup is Kamailio as edge proxy to asterisk to allow register only with ...
Mohammad Sarshar's user avatar
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415 views

CDR not working with asterisk & odbc

I had ODBC working with asterisk to write to my CDR then recently needed to check my CDR to realise nothing had been written for about 6 months. I had a look at my config, nothing seamed to have ...
Ben's user avatar
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the freepbx queue.log union in asterisk one directory?

how to merge free pbx queue.log into one directory of asterisk ? I tried to combine them but in myphpadmin queue_name comes queuenumber and I don't want that.
nika gubelidze's user avatar
1 vote
1 answer
43 views

One way audio, bad echo on asterisk trunks with PAP2T ATAs

This is not a question, I solved it. I set up Asterisk trunks between offices, with encryption, some offices do not have fixed IP and login in to the central server. That all went well. Anybody needs ...
SeniorGeek's user avatar
1 vote
1 answer
166 views

Fail2Ban - Match Asterisk PJSIP Successful Authentication

I'm trying to create a fail2ban filter that will match successful authentications. An example log entry looks like this: [2023-05-25 18:41:00] VERBOSE[26149] res_pjsip/pjsip_options.c: Contact user/...
Nick Coons's user avatar
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Fault tolerant setup of Asterisk and FreePBX using a dedicated line

I am attempting to configure a fault tolerant setup of Asterisk with FreePBX. Going over the docs, I see I need to configure Pacemaker. Not a problem in itself as I've used PCS before. I have 2 ...
Hussain Akbar's user avatar
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32 views

How can Asterisk voicemail handle call holding?

Given an Asterisk PBX system version 20 or above. Given that a user calls another user, the call is missed and is sent to voicemail. While the caller is directed to voicemail, the caller receives ...
Charles Langlois's user avatar
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54 views

Atxfer does not transfer

I have two agent A and B. Agent A has a call online and wants to transfer the call to agent B. I already have axtfer => *2 configured in the features.conf file and I have alltransfer set to YES on ...
drop sip's user avatar
2 votes
1 answer
256 views

Linux ports become unavailable

We have some Ubuntu 18.04 servers with hard-to-diagnose problems that have some similarities. The servers run a hosted VOIP PBX service with various component applications. Kamailio occasionally ...
David Cunningham's user avatar
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1 answer
65 views

Authentication error in trunking between two asterisk

i'm creating a connection between two asterisk servers. I want only make calls from right side server to left side server, but i receive an authentication error. I can´t find the problem. The Error ...
Tomás Figueiredo's user avatar
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1 answer
215 views

Asteris No compatible codecs, not accepting this offer!

I have installed an asterisk it repeats this statement, again and again, my server is overloaded because of these errors. [Sep 19 18:34:23] NOTICE[1641][C-00000398]: chan_sip.c:10995 process_sdp: No ...
Huzoor Bux's user avatar
1 vote
0 answers
112 views

connecting two SIP phones from different LANs over ZeroTier openwrt

i make site to site conncetion by tp-link n600 openwrt zerotier and no problem , i can reach every thing in two sites and ping ok but i have in site 1 Issable and in site 2 sip phone ------------------...
Best Way's user avatar
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1 answer
994 views

Problems setting up a secure WSS connection for Sipml5

I have problems with setting up a webrts connection with sipml5 through an asterisk. When I check the status of https through the asterisk console, I get a response that it is up and running. But when ...
Ivan Dovgan's user avatar
1 vote
0 answers
175 views

My asterisk server can't read extensions.conf

I've installed asterisk on ubuntu 22 on virtualbox everything is ok when voip call comes to my server it says me : rejected because extension not found in context 'default'. my sip.conf file : [1060] ...
behruz montazeri's user avatar
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1 answer
3k views

Asterisk shows No matching endpoint found

I have planned to implement Asterisk SIP server for testing eMTA calls. I don't have eMTAs so I decided to start with Linux Soft client and then, when I will have an eMTA and physical access to the ...
Oleksandr Znachkov's user avatar
1 vote
1 answer
632 views

fail2ban not creating jails in CentOS

I'm trying to create a jail in Fail2Ban to protect a Asterisk PBX. I already did as suggested and made a copy of jail.conf named jail.local. In jail.local, I set up a asterisk jail as follows: [...
GD1's user avatar
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1 answer
99 views

How to enable queue_log on the embedded Asterisk of a Yeastar MyPBX?

I have an old PBX from Yeastar that contains an instance of Asterisk 1.6.2.6. How can I enable the queue_log setting and have it persist across reboots?
Joril's user avatar
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1 answer
497 views

Asterisk: pri to pri connection. Green light but a lot of hdlc errors

I want to connect(for learning, nothing serious), two asterisk server via ISDN PRI. The first server will act as "telco" so is pri_net, the second server is the cpe, so I will use pri_cpe. ...
elbarna's user avatar
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1 answer
118 views

asterisk - what's difference between format_* and codec_*?

When building Asterisk, there is two different entities related to audio formats - formats and codecs. What's difference? For example, I need a support only for alaw+ulaw+g.729 on voice traffic itself,...
ZZ Wave's user avatar
0 votes
1 answer
373 views

No RTP engine was found. Do you have one loaded? Asterisk-18.10.1

I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://...
xenon's user avatar
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0 answers
69 views

How to change pickupexten on asterisk 11

I'm looking for a way to change ‍pickupexten in Asterisk 11. Is there a way? The features.conf: pickupexten = *8 pickupgroup=1 pickupsound = beep pickupfailsound = beeperr I want to change *8 with ...
F.M's user avatar
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3 answers
3k views

Forwarding SIP headers with asterisk (PJSIP)

I'm trying to forward a specific incoming header to the other leg of the call, but can't figure out how to pass the value of the header in the incoming leg to the pre-dial handler [addheaders] exten =&...
Pownyan's user avatar
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0 votes
1 answer
225 views

command that cannot work in background (asterisk)

am trying to run script in background which contain command such like "asterisk -rvd >> xyz.log", when I run it directly it work well but when run it with any background way (service, ...
Centos's user avatar
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0 answers
554 views

asterisk not recognizing answer from sip trunk

I have an Asterisk server (15.5, FreePBX) with three SIP trunks from different providers configured, two of them are working fine while the third for every call keep sendind the invite despite the ...
Spuria's user avatar
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0 answers
268 views

Asterisk automatic hang up when I try to call

I just set up a server with Asterisk set up but when I try to make a VoIP call the line automatically drops... I use CommPeak as a SIP provider and Bria for the interface Here's my sip.conf: [general] ...
dolor3sh4ze's user avatar
1 vote
1 answer
916 views

VoIP one-way audio, only when call initiated from one side

I am setting up some new switches and VLANs and I am getting trouble with our pre-existing Asterisk VoIP set-up. Most calls work ok. Some get just one-way audio. I tried to narrow it down to this ...
pgr's user avatar
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1 answer
580 views

Asterisk/FreePBX queues no longer working

Incoming calls go straight through to failover regardless what other settings are. I have deleted the queue and rebuilt it, nothing seams to fix it... Asterisk 16 Freepbx 15 Debian 10 There is a ...
Ben's user avatar
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0 votes
1 answer
371 views

adding modules to installed asterisk - alsa.so not found

I have set load= chan_alsa.so and i got this error ERROR[77064] loader.c: Error loading module 'alsa.so': /usr/lib/asterisk/modules/alsa.so: cannot open shared object file: No such file or directory ...
mordechai's user avatar
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0 answers
181 views

Asterisk - Position Announcement to First Caller

When Asterisk announces the position of a caller in the queue, it plays queue-thereare, followed by the caller's position, followed by queue-callswaiting. We have custom messages for each of these ...
Nick Coons's user avatar
1 vote
1 answer
381 views

tel: URI support in Freeswitch(for outgoing calls)

I have integrated a Freeswitch instance with an external SIP server and calls are working without any issues. But now I need to change the SIP server which only supports tel: URI. In the case of ...
uts9's user avatar
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0 votes
0 answers
671 views

Audio issue in public network with Asterisk 17

I have just installed and configured Asterisk 17 in a desktop PC running Ubuntu 18.4 My Asterisk and one of the clients using Zoiper Softphone are behind NAT. Another Client is an iPhone running on 4G ...
Econ_newbie's user avatar
2 votes
1 answer
709 views

Asterisk: "in call" queue member status, while actually no calls

My asterisk won't direct inbound calls to certain queue members, because it thinks that they are busy receiving another call. But they are not. They are waiting for a call (register and waiting), or ...
Roman's user avatar
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1 answer
1k views

asterisk -vvvvvrx 'core show channels verbose'

The output of the command asterisk -vvvvvrx 'core show channels verbose' shows total number of calls processed. That number is an indication of the calls processed since the service is running or last ...
Rahatur's user avatar
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1 vote
2 answers
2k views

Issabel-Asterisk- How to handle multi-digit inputs on IVR

I have an Issable server running and integrated with Cisco Call Manager correctly. I am using Issabel server as my IVR system to answer calls without any direct extension callings. My Scenario is as ...
Zareh Kasparian's user avatar
-1 votes
1 answer
761 views

how to run multiple instances of asterisk on one server

hope everybody is safe during challenging time i have a task to automate our demo installation process. Our software is mainly a GUI that operates with asterisk via AGI and stores all data to MySQL / ...
d3m0n's user avatar
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2 votes
1 answer
289 views

Large Jitter Spikes on a Small Percentage of VoIP calls

On about 5% of our customer's calls, we see large jitter spikes and high delta #s that have caused an noticeable audible impact on call quality. (Stuttering/Breakups/Robotic Audio). We know this from ...
AVoIPm8's user avatar
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0 votes
1 answer
1k views

FreePBX / Asterisk: use inbound routes to block spammers/hackers

My FreePBX / Asterisk configuration was recently forced into allowing both anonymous inbound calls and SIP guests. So of course we're now getting blasted with spam/hack attempts. They show up in the ...
JimB's user avatar
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0 votes
1 answer
1k views

Asterisk: Using Queue() in the h extension

I'm trying to get users to be able to record a message, hangup and have the call continue and dial Queues and playback the recording. I've gotten most of the way there, but right now when I call ...
Mattisdada's user avatar
1 vote
0 answers
293 views

Initial one way audio with SIP trunk

I have an Asterisk server sitting on my network behind a pfSense firewall, it has two trunks, one for my household provided by my ISP using PJSIP and the other for my business provided by a third ...
Geoffrey's user avatar

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