similar to: * works, but after a few seconds audio always stops.

Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking

Hi, I''m a very noob ror programer (i''ve started yesterday), but for my live, i''ve pass all the day going up & down, and in order to have a constant learning, i want to install ruby on rails, mysql, mysql-front, and RadRails on a 512mb PenDrive. I''ve followed these steps: 1) Download latest RubyOnRails windows installer. 2) Execute it settign the path to

I recently rebooted one of my machines and was greeted with a message about one of my ext3 partitions having some errors. I dutifully started e2fsck on the partition, but after a few hours of some useful messages about fixing inodes, it then started printing out lists of ascending numbers which I'm not sure how to interpret. The first time I noticed this, these numbers were in the 4000000

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

I'm trying to capture when a call is placed on and removed from being on hold through the AMI in Asterisk 12.3. In previous versions, the Hold event contained a 'Status' field which indicated if the call was going 'On' or 'Off' hold. However, in 12 not only am I not seeing the Status field, but I am not seeing any AMI Hold event that corresponds to removing the call

Hi There seems to be a few of these around, so was just looking for some recommendations. I need to be able to tag different objects, such as events and groups by a category. If anyoen can point me in the right direction on which one to use, or is aware of any reviews, please post them below thanks jon -- Posted via http://www.ruby-forum.com/.

Hi, I am getting loads of modprobe errors flying past on the screen during startup.. I never got these errors on any of the PC I installed 3.3 onto but now I have tried 3.4 on two seperate PC's and got these errors.. Anyoen else having this? Later..

i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

How can I detect WINE from my program, or detect Linux, in other wors how can I detect that OS is not Windows. BOOL WINAPI GetVersionEx(LPOSVERSIONINFO lpVersionInfo) returns that OS is Vista :D [Question]

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

Hi everybody We have a strange problem with several asterisk servers (Version 1.4.11) using PRI cards (tied to telco here in Belgium). Indeed we noticed that whenever a local user places an outgoing call through the PRI (and telco) to another IPBX (tied to telco using BRI or PRI), if the remote party places the call on hold, the caller hears the _local_ music on hold instead of the

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50