similar to: Can asterisk unregister?

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

On Sun, Nov 02, 2014 at 11:06:13PM +0800, Amos Kong wrote: > On Fri, Oct 31, 2014 at 03:23:21PM +0800, Herbert Xu wrote: > > On Fri, Oct 31, 2014 at 10:28:00AM +1030, Rusty Russell wrote: > > > Herbert Xu <herbert at gondor.apana.org.au> writes: > > > > On Thu, Sep 18, 2014 at 08:37:45PM +0800, Amos Kong wrote: > > > >> From: Rusty Russell

On Sun, Nov 02, 2014 at 11:06:13PM +0800, Amos Kong wrote: > On Fri, Oct 31, 2014 at 03:23:21PM +0800, Herbert Xu wrote: > > On Fri, Oct 31, 2014 at 10:28:00AM +1030, Rusty Russell wrote: > > > Herbert Xu <herbert at gondor.apana.org.au> writes: > > > > On Thu, Sep 18, 2014 at 08:37:45PM +0800, Amos Kong wrote: > > > >> From: Rusty Russell

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

Hello dear list. A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, and we had the opportunity to unregister user by typing *-a number and -* again, ex * 99 *, and then the phone number/sip extension was unavailable, and all of the calls to that extension was redirected to the receptionist. When the user came back and wanted to register her sip account/extension, the

On Fri, Oct 31, 2014 at 03:23:21PM +0800, Herbert Xu wrote: > On Fri, Oct 31, 2014 at 10:28:00AM +1030, Rusty Russell wrote: > > Herbert Xu <herbert at gondor.apana.org.au> writes: > > > On Thu, Sep 18, 2014 at 08:37:45PM +0800, Amos Kong wrote: > > >> From: Rusty Russell <rusty at rustcorp.com.au> > > >> > > >> The previous patch

Amos Kong <akong at redhat.com> writes: > From: Rusty Russell <rusty at rustcorp.com.au> > > The previous patch added one potential problem: we can still be > reading from a hwrng when it's unregistered. Add a wait for zero > in the hwrng_unregister path. > > v4: add cleanup_done flag to insure that cleanup is done That's a bit weird. The usual pattern

On Fri, Jun 05, 2020 at 02:46:24PM -0700, Sean Christopherson wrote: > Unregister each queue's xdp_rxq during freeze, and re-register the new > instance during restore. All queues are released during free and > recreated during restore, i.e. the pre-freeze xdp_rxq will be lost. > > The bug is detected by WARNs in xdp_rxq_info_unreg() and > xdp_rxq_info_unreg_mem_model()

On Fri, Oct 31, 2014 at 10:28:00AM +1030, Rusty Russell wrote: > Herbert Xu <herbert at gondor.apana.org.au> writes: > > On Thu, Sep 18, 2014 at 08:37:45PM +0800, Amos Kong wrote: > >> From: Rusty Russell <rusty at rustcorp.com.au> > >> > >> The previous patch added one potential problem: we can still be > >> reading from a hwrng when

On Fri, Oct 31, 2014 at 10:28:00AM +1030, Rusty Russell wrote: > Herbert Xu <herbert at gondor.apana.org.au> writes: > > On Thu, Sep 18, 2014 at 08:37:45PM +0800, Amos Kong wrote: > >> From: Rusty Russell <rusty at rustcorp.com.au> > >> > >> The previous patch added one potential problem: we can still be > >> reading from a hwrng when

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

Hello, I found the problem and also the workaround. Clearly, since it was working with chan_sip it should not be dialplan problem, but sip stack problem. I have line=yes set up. After asterisk restart the old registration is not unregistered and new one is registered with different line value. Then incoming invites and qualify requests are sent to all the registrations and there the problem

hello to asterisk start WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum retries exceeded on call xxxxxxx...xxxxxxxxxx@192.168.0.1 for seqno 102 (Request) with a call from x-lite 38113@fwd.pulver.com WARNING {98311] : File chan_sip.c, line 2002 (__transmit_response): Unable to determine sequence number from '' and x-lite hang up the second warning is new since morning

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

hi! i am working on a tree which allows moving nodes/subtrees via drag&drop. since i am not a javascript expert i try to stick to the script.aculo.us library without modifying anything. the nodes are loaded dynamically from my rails-app and have a unique ID. when a node gets moved to another folder, it disappears at its original destination (via Element.remove()) and gets drawn as soon

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk

Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address

hi all, i have 2 x-lite version 3.0 softphones configured on extension 9000 and 9005. i have one call the other and then try and send an IM between them using the x-lite IM facility. the asterisk console shows the message... WARNING[27193]: chan_sip.c:11866 receive_message: Received message to "s"9005 at hhh> from "c"9000 at hhh>;tag=717de473, dropped it... when i