similar to: call Hold event asterisk

hi When i tried to install centos 7, my nvidia ethernet port not detected during installation whereas all of other distros working perfectly without any problem... Pl help.. my rig is AMD Athlon x2 64 processor and Nvidia chipset.. output of lspci -nn | grep -i net 00:0a.0 Ethernet controller [0200]: NVIDIA Corporation MCP77 Ethernet [10de:0760] (rev a2) with regards sathish.

Hi, I am newbie to rails , I need access facebook page information like wall, subscription,picture etc. I have apps in rails 3, it has list of artist users he upload songs, videos within my application, when the public (No Need to Login) can hear the list of songs,videos. i decide to fetch the facebook/twitter basic fanpage information and shown to the public in the popups. i

my d-link ethernet pci card not detected in centos 7 even after i have installed kmod forcedth driver.. output of lscpi -nn 01:07.0 Ethernet controller [0200]: VIA Technologies, Inc. VT6105/VT6106S [Rhine-III] [1106:3106] (rev 8b) pl help with regards.. sathish

Hi, I need to install R packages in several nodes (10+) in AWS. I wont be able to open R shell in each and do install.packages("foo") This will be done using a configuration management tool like Puppet and it'll be easier if i can do an apt-get installation of R packages automatically. I found a list of R debian packages here: http://cran.cnr.berkeley.edu/bin/linux/ubuntu/lucid/

Hi, I need to install R packages in several nodes (10+) in AWS. I wont be able to open R shell in each and do install.packages("foo") This will be done using a configuration management tool like Puppet and it'll be easier if i can do an apt-get installation of R packages automatically. I found a list of R debian packages here: http://cran.cnr.berkeley.edu/bin/linux/ubuntu/lucid/

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

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

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

Hello everybody and merry xmas. I have a problem with sip phones calling each other inside the same network (no nat, no firewall). You can make and receive calls and pick them up, but you cannot hear anything on any side of the call. But if you press hold/unhold or you transfer the call, then everything works as expected. Ths SIP phones I've tried are Swissvoice IP10s and kphone, it

zpool status shows a few checksum errors against 1 device in a raidz1 3 disk array and no read or write errors against that device. The pool marked as degraded. Is there a difference if you clear the errors for the pool before you scrub versus scrubing then clearing the errors? I''m not sure if the clearing errors prior to a scrub will replicate out any bad blocks that were identified as

Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just

Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) & Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is configured on MV-370. The MV-370 configuration is: Mobile to Lan Table : 0 * 192.168.1.1 Lan to Mobile Table: 0 * # SIP Setting: Display Name: Portech User Name: 1005

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

I''ve been trying to get a handle on AJAX today and am ready to call "BUG!" I started with the code from Curt''s "AJAX on Rails". It used <ul="my_list"> and <li> and worked fine. Except I need a table instead of a list. So I did the simplest thing that I thought could possibly work. In the view, I changed the :position attribute from

Hi All, We have an email receiver component to receive mails. We couldn’t get the bounced email id from “failure delivery notice” email. Scenario: In email receiver, we need to process the bounced emails separately as in it needs to be tracked and take a action to stop sending emails to particular email-id if it is hard bounce, and if it’s a soft bounce, we have to make a count for it and take

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 Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk> wrote: > 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

Hi All, Anybody know about to using AMI to agent answer call that placed in QUEUE. Asterisk Version : 1.8.9.1 Queue Name : agent-support Agent logged in Extension : SIP/8001,SIP/8002 ... strategy : ringall 1) How to intimate or send information to available agent to inside queue incoming call ringing. 2) How to agent answer incoming call that placed in queue. 3) AMI for agent Hold/Unhold call.

Hi, I have been blessed with the "Numeric 2000 Plus" UPS, suspected to be of 2KVA rating. This UPS does not figure in the list of UPS recognised by NUT. However I get lucky with megatec driver. Here is the log... ----------------------------------------------------------------------------------------------------- Network UPS Tools - Megatec protocol driver 1.5 (2.0.5) Carlos Rodrigues

The Asterisk Development Team would like to announce the release of Asterisk 17.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: