similar to: how to check Asterisk SIP registration

Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via ->QueueAdd("4050", "Local/4053 at from-internal/n", 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The "show queue" command still displays 4053 as "In use". However, if 3210 calls 4050

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

On 18/07/2019 17:36, Kris Lou via samba wrote: > Might have something to do with this bug: > https://bugzilla.samba.org/show_bug.cgi?id=11482 > > You can find and make the relevant changes with ADSI Edit. > > The problem is that renaming a user on Linux works the opposite way to Windows and neither renames everything. If you rename on Linux with ldbrename, it renames dn, cn,

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

You're quite right. We'll need to see your misdn.conf file to check the settings. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 13:49 -->> To: asterisk-users at lists.digium.com -->> Subject:

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

Hi, I have a PRI euroisdn link between an Alcatel PBX and Asterisk. I'm having some trouble with overlap dialing. Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap". I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:36 -->> To: asterisk-users at lists.digium.com -->>

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

I am writing a cron script to check if certain extensions are online and if they aren't then Asterisk creates a couple of .call files to notify another set of extensions or external numbers. It works fine except for logging information. What I'm doing in the script is setting a "fake" caller ID (as it's generated by Asterisk, not by a user) and calling out real users. So

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as

I'm wondering if hdlc can be the culprit (not sure what it is and what it does). Should I set hdlc to yes in misdn.conf (I'm asking before testing because this is a production system)? misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ntkeepcalls=no bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What