similar to: Parking lot problems

Is there anyway of storing an incoming calls CallerID on a parked call and having it restored when someone picks up the parked call? I've tried storing the CID as a global variable and restoring it in my dialplan, and while NoOp shows it working, the phone ignores it and uses the parking lot extension for callerid instead. I believe this is because the phone is calling out instead of a call

Asterisk Version: 1.8.1.1 Problem: Multiple Parking Lots Issue: Not redirecting to the right parking lot. Always uses the first parking lot from "parkedcalls show" or "features show" Asterisk Working Version: 1.6.1 Steps Taken: In features.conf added: [parkinglot_test] context => parkedcalls-test parkext => 700 parkpos => 701-710 parkingtime => 120 findslot

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

Hello everybody I have an asterisk with an integration of alcatel pbx, by sip trunk, all calls are fine, but tha calls calls that originate from a analog line, the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx. When a asteris user call to analog line the call is ok. Everyone, has been that problem? I change asterisk version 1.4.21 to

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

I have 4 asterisk servers. They all have local phones on their local network they manage for SIP based conversations. We then have IAX between them all for inter-asterisk connections. This setup has worked well for nearly 2 years now, minor problems here and there but overall very nice. Recently we acquired some Polycom video conference units. I was able to setup our main server to host all

Hi, I've been trying to set up multiple parking lots using multiple tenants on version 1.8.x (tried all versions including 1.8.4RC2), however calls only park on one parking lot (the top parking lot of the command 'parkedcalls show'). Everything works fine when running on version 1.6.2.17. Currently have a bug opened, but haven't got any updates for over a month.

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should