similar to: Queue - Multiple Transfer

Displaying 20 results from an estimated 1000 matches similar to: "Queue - Multiple Transfer"

2009 May 20
1
Queue and Dial operation - Common Variables?
Hi All, I am trying to implement ACD using Asterisk 1.2.18 and I've chosen AgentCallbackLogin for login purpose. One AGI is written which will actually get executed when agent dials '1001' (say) from his SIP phone and enters into the queue. Second AGI gets executed when the Dial operation is performed. I see the agi_uniqueid obtained from both AGI instances are different and I
2009 Jul 04
2
x86_64 EDAC throwing error
Hi All, We have installed CentOS 5.3 x86_64 in an HP DL585 server with AMD Opteron 64 bit processor and 16 GB RAM. The kernel version is 2.6.18-128.el5 . Now this has thrown an error message in /var/log/message, Jul 3 21:41:11 db1 kernel: EDAC k8 MC0: general bus error: participating processor(local node origin), time-out(no timeout) memory transaction type(generic read), mem or i/o(mem
2009 Mar 10
1
Asterisk and WebIntegration
Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with
2009 Jun 07
2
Call recording in - out
Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]
2009 Apr 22
5
Asterisk routine maintenance activities
Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090422/8cfeb434/attachment.htm
2009 Jun 29
2
Building Custom Kernel - CentOS 4.4
Hi All, I am having issue when trying to build a custom kernel in CentOS 4.4. Current Kernel Version is 2.6.9-42.ELsmp and the server is HP Proliant DL380 G3. I downloaded the source rpm and then installed it. Gave the command, rpmbuild -bp --target=i686 /usr/src/redhat/SPECS/kernel-2.6.spec. But it throws me an error after patch operations. The following is the error.
2011 Apr 09
1
custom iptables rules
Hi, Is there a way that we can add custom IP-Tables rules in a nat'd physical host? I need some custom rules mentioned in physical host to access some services in the guest systems. Any hints on this? Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2013 Jul 03
1
Recommended filesystem for GlusterFS bricks.
Hi, Which is the recommended filesystem to be used for the bricks in glusterFS. ?? XFS/EXT3/EXT4 etc .???? Thanks & Regards, Bobby Jacob Senior Technical Systems Engineer | eGroup P SAVE TREES. Please don't print this e-mail unless you really need to. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Dec 01
2
Error while copying/moving file
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2009 Jul 17
0
Queue member (Agent) does not Dial
Hi All, We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue (maqueue) structure for handling customer calls. There are 4 queue members (85744,85766,85511,84888). These 4 members are logged in using AgentCallbackLogin application. But at some point, one of the agent's SIP phone does not ring for an incoming call to this queue. I checked the agent status and it is not in paused
2009 May 06
0
Caller information in Web
Hi All, Is there a way that I can bring the callerID obtained from an incoming call to display in a web browser. I am using eyeBeam as softphone. Please help me with this. Thanks in advance. Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090506/7de6e24c/attachment.htm
2017 Oct 26
5
Re: Need to increase the rx and tx buffer size of my interface
Hi Ashish, I have tested with your xml in the first mail, and it works for rx_queue_size(see below). multiqueue need to work with vhost backend driver. And when you set "queues=1" it will ignored. Please check your qemu-kvm-rhev package, should be newer than qemu-kvm-rhev-2.9.0-16.el7_4.2 And the logs? tx_queue_size='512' will not work in the guest with direct type interface,
2017 Oct 26
2
Re: Need to increase the rx and tx buffer size of my interface
On 10/26/2017 10:38 AM, Ashish Kurian wrote: > Hi Yalan and Michal, > > Thank you for your response. So what I understand is that I can change > rx_queue size even if I use direct type interface and qemu driver as long > as the driver is virtio. Am I right? Yes. > If that is the case why am I getting > the error saying that > > error: XML document failed to validate
2017 Oct 26
2
Re: Need to increase the rx and tx buffer size of my interface
Hi Ashish, IMO, it is yes, no way to increase tx_queue_size for direct type interface ------- Best Regards, Yalan Zhang IRC: yalzhang Internal phone: 8389413 On Thu, Oct 26, 2017 at 3:38 PM, Ashish Kurian <ashishbnv@gmail.com> wrote: > Hi Yalan, > > In the previous email you mentioned "tx_queue_size='512' will not work in > the guest with direct type
2017 Oct 25
2
Re: Need to increase the rx and tx buffer size of my interface
Hi Michal, What I found was that when I restarted the machine and did a virsh edit command to see the xml config, I see that it is was not actually changed. This suggests why I saw 256 again after restarting. So now I tried again to edit the xml via virsh edit command and used the following to set the parameters. <driver name='qemu' txmode='iothread' ioeventfd='on'
2010 Nov 01
2
Best supported motherboard
Which among these three motherboard is the best supported in Linux (Centos/Ubuntu) ? Intel? Desktop Board DH55HC [ http://ark.intel.com/Product.aspx?id=42409 ] Intel? Desktop Board DP55WG [ http://ark.intel.com/Product.aspx?id=40078 ] Intel? Desktop Board DP55KG [ http://www.intel.com/products/desktop/motherboards/DP55KG/DP55KG-overview.htm] Also which will be a good processor , for using
2010 Jul 30
1
Samba PDC + BDC using LDAP
I am trying to implement Samba Domain Controller with replication using LDAP . My intention is to create two Samba DC servers with LDAP backend , then to set one as master and other as slave , then replicate LDAP database in master to slave I shutdown second samba server and only first samba DC was running . Then I added one XP client to first samba server , I can successfully logged in to
2017 Oct 25
2
Need to increase the rx and tx buffer size of my interface
Dear Users/Developers, I am using a KVM Ubuntu VM as a degrader to apply specific delays to incoming packets. As the delay for my packets can be higher than 7.5 seconds, there is not enough buffer on my interface to buffer all the packets. Therefore those overflowing packets are dropped in the machine and not forwarded. When I tried to use the command ethtool -G ens8 rx 512 to increase the
2017 Oct 26
1
Re: Need to increase the rx and tx buffer size of my interface
Hi Michal, I installed the latest version from the git and did run the libvirt from the new directory using /home/to/your/checkout/daemon/libvirtd But I still see that the version is 1.3.1. I have also stopped the already running libvirt daemon and ran the new daemon from the directory and it is showing 1.3.1 Best Regards, Ashish Kurian On Thu, Oct 26, 2017 at 10:50 AM, Ashish Kurian
2008 Sep 02
4
AgentCallbackLogin AddQueueMember
Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not