similar to: Extension cant pickup calls but can transfer.

Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be transfered to the trunk that directs to the second server. How do I do to configure it this way?

Greetings. I'm running asterisk here (elastix) and I have a few extensions configured in it. I have here two different callgroup/pickgroup where the extensions are configured in, but it doesn't work when I try do pickup a call. Looking the config file (sip_additional.conf) I see they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear

Greetings. I've run into a trouble when trying to add a new Win7 machine on a domain. The domain is controlled by a server running Samba + LDAP (samba compiled with ldap support), on a Debian 5 OS at the local network. I've added the machine name to the LDAP three through phpldapadmin using the option "Samba3 Machine" on the related submenu and via terminal on samba. Then I

Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine....

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

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 list, i'm trying to do that iax channels can acces the pickup feature(normaly *8 dialing). But always the iax channel when dial *8, search for the extensi?n *8 on its context. I know i can program the *8 extension with the pickup applicati?n. But its doesn't works for me, becouse i need to pickup some calls comming from IVR's o Queues. And there de exten is no the same as the

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extension 100 when there is a call to it. From CLI, I can see it issue **100 to asterisk but failed

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

Hey there list ! I'm receiving negative feedback when people try to pickup another ringing phone by pressing *8 on there own Grandstream device. These are my setting that should make pickup possible : all my sip-clients (Grandstream) have this in their config (sip.conf) : callgroup=1 pickupgroup=1 canreinvite=no qualify=yes So they are all in the same pickupgroup... This the

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

What's the proper syntax for implementing directed call pickup? Running cvs-head from today (9/24/05 including Mark's fixes), and tried: exten => *99,1,Pickup(${EXTEN:3}) but that does not seem to work, and there isn't an example in the configs directory. 'show application pickup' suggests the above should work with our sip phones, but apparently I'm missing

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek