similar to: Unable to pickup an extension, trying everything

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

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

Hi, I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is configured for call pickup as recommended by Aastra. When the LED flashes, I press the corresponding button and the display tells me "Call not possible". In the CLI is see the follwoing output. Why isn't the call transfered to user2 and what does "No target channel found for 31 mean"? User2 can

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

Hello list, why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ?? [Jun 29 14:06:34] -- Executing [s at macro-vac:10] *GotoIfTime*("SIP/testcorp-00000036", "**|*|29-*|jun*?onvac") in new stack [Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*', assuming none [Jun 29 14:06:34] -- Executing [s at macro-vac:11]

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still

Hello, We have a situation where a call comes in, users are notified via an external process (curl request to web service), and we can't answer the call until a callee can call in and pickup the call. How can we implement this functionality? We tried using : [caller-inbound-leg] ; code to send the CALL_UUID information to users. exten =>

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br> <br> Using Asterisk 1.4.25.1<br> Using realtime sip_buddies<br> <br> I notice

Hello list, I have the following problem with MySQL-queries : it seems that the resultid and connid are not cleared ! [macro-GetMailboxFromSIPuserID] exten => s,1,MYSQL(Connect connid localhost xxx xxx xxx) exten => s,n,MYSQL(Query resultid ${connid} SELECT\ extensie FROM\ tbl_SIPaccounts\ WHERE\ ID="${ARG1}") exten => s,n,MYSQL(Fetch fetchid ${resultid} extensie) exten

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez Inviato: venerd? 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp

Hi! I'm using ael language and I need to pick up a call from outside to an internal number. for example: i'm 120 the phone 100 rings, it's a call from outside. now I pick up the call with: *8100 and I would expect to answer the call but the response is Declined the Puckup code is below: _*8X! => { Pickup(${EXTEN:2}); Hangup(); } the problem is if an another number ( 130) calls

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

Hi, I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap :) I was wondering if someone could point me at 3 things that I appear to have "lost"? 1) ZapEC(off) - Is there an equivalent dialplan command to request no EC on a channel before dialling in DAHDI? 2) rxfax(file.tiff) - I have found ReceiveFax(), but I am aware that much has happened in the faxing stakes