similar to: Extension state can get stuck in 'Ringing' state

Hi, For some time now I have noticed that our RBS T1 (asterisk 1.4.35, Dahdi 2.3.0+2.3.0, TE410P) often has channels stuck in the state "Ringing", like this poor chap who got stuck on two calls in a row, apparently: [excerpt from "core show channels"] SIP/7157997-0000534b 7760308 at business:1 Ring Dial(Dahdi/g0/7760308) DAHDI/3-1 5130262 at from-pstn:1

Hi, I set verbose to 3, but I do not see any RINGING notification in the CLI. On the contrary, when the phone goes UNREACHABLE I get: [Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer '152' is now UNREACHABLE! Last qualify: 130 == Extension Changed 152[blf] new state Unavailable for Notify User 154 [Dec 13 21:11:08] NOTICE[9988]: chan_sip.c:20196

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

Hi, I have simulated ?Chan phone? driver according to my own driver code and I am able to make internal and external [trunk] Asterisk calls. Only issue I am facing is with hangup in ringing state of incoming call. (1) Make a call from external X-lite to FXS and FXS is in ringing state now (2) Disconnect the caller [X-lite] (3) X-lite sending cancel message to asterisk but hangup

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters:

>All, >I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and >are answerd by the auto attendant. When an extensions is entered I see >the Dial(SIP/100) on the console but the phone never rings... >I can pick up any extension and call 100 and it rings just fin>e and they >answer and everthing is fine. All 8 extensions can dial 9 for an analog line >and call

When I call from extension A on Box and to Extension A on Box B I get no ringing sound. Regards Paul Dracevich Wireless Technology Consultant Wayby Group Mobile +64 29 638 9675 Phone +64 9 623 2143 Fax +64 9 623 1380 email paul@vnet.cc website www.vnet.cc <file:///C:\Documents%20and%20Settings\paul\Application%20Data\Microsoft \Signatures\www.vnet.cc> "the freedom to communicate

Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, -- There are no Debian developers in any part of Hell, because the good karma incurred by being one takes you straight to the pearly gates. Of course, the frequent flame wars you put

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

Adds VmStateChangeEvent class and uses VmObserver to track/audit state changes for Vm objects. Callbacks in VmObserver also update the new total_uptime and total_uptime_timestamp attributes of Vm class, which allows easy calculation of time spent in running states for user resource use accounting. --- src/app/models/vm.rb | 34 +++++++++++ src/app/models/vm_observer.rb

Hi All... When I dial into my Asterisk box and then dial an extension, I here silence until the person picks up or until the call goes to voice mail. At one point I had Asterisk configured to play music during this time by adding the m to the extension, but music on hold does not work for me (I think mpg123 does not work on my box) so I turned off music on hold and removed the m. Now I just

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

I have a line in my house that goes to phones across the line BEFORE Asterisk. My wife uses them :) I plugged the line into an FXO port and pointed it at an extension, so I can pick it up as well. I can also pick it up with *8. One side effect is that once my wife picks up a call, the Asterisk extension continues to ring until the Zap card detects that it's not ringing any more. What

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls