similar to: dialplan =>how many concurrent calls

Hello friends I've problem with my Centos 4.3. On start some information: 1. serwer HP DL 380 x Intel(R) Xeon(TM) CPU 3.40GHz 2. RAM 4GB 3. HDD storage FC 4. kernel 2.6.9-34.0.2.ELsmp #1 SMP This serwer is working with another one (that some configuration) in RHEL cluster (locking by DLM), as a file serwer with samba-3.0.22. Samba have about 100 simultaneus sessions. Problem is with load

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

Hello guys, I've one SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter "limitonpeers" will limit at the trunk level, right ? Thanks in advance

Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking

I have Postfix/Dovecot/postfixadmin/MySQL with several virtual mailbox domains one of the domains is like aname.com.au, the user also now has aname.com, and, would like to 'mirror' most of the addresses to be user at aname.com, THOUGH, some are to remain as user2 at aname.com.au so, both user at aname.com as well as user at aname.com.au should be one user the users retrive emails as

Hey, I'm still waiting for the new file server hardware to arrive so I plan to enable user logins in the meantime without profiles and home directories. Reason for this is quite some pressure from people. The setup on the 2 DCs should be done a day or two. Could there be problems if I configure profiles and home directories some weeks after users logged in for the first time? What if e.g.

While SVCB/HTTPS provides a better solution for the browsing use case, I see other use cases where ALIAS/ANAME would be ideal, notably in apex RRs. So while fostering SVCB/HTTPS deployment is a good thing, I wouldn?t mind name server software implementing ALIAS. Including NSD, but I reckon it?s much more challenging to do due to NSD architecture than it was to implement it in PowerDNS. But if

In a function, say foo.glm for glm objects I want to use the name of the object as a label for some output, but *only* if a glm object was passed as an argument, not a call to glm() producing that object. How can I distinguish these two cases? For example, I can use the following to get the name of the argument: foo.glm <- function(object) { oname <- as.character(sys.call())[2]

Hi Guys.. I am making an anoucement machine that is not allowed to "answer" the call due to a billing issue. I found that Playback with "noanwser" is usefull in this case. $AGI->exec('Playback',"$message","noanswer")} But when i request some values to the user with get_data, i think there is an answer anywere. Is there a way to get_data

Hi Christof! AFAIK, PowerDNS is the only open source name server that supports ALIAS. There was an idea to standardize ALIAS as "ANAME" (https://datatracker.ietf.org/doc/draft-ietf-dnsop-aname/), but the idea was dropped in favor of SVCB/HTTPS record https://datatracker.ietf.org/doc/rfc9460/. So now we have to wait until all Browser vendors implement SVCB/HTTPS. Regards Klaus PS: If

smbclient (and smbtar) in version 2.2.7a (and prior) has problems with large files (> 4GB). The following patch (against 2.2.7a) fixes all known problems with this. This code has been checked into the CVS tree in all branches as well. -- ====================================================================== Herb Lewis Silicon Graphics Networking Engineer

Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: exten => s,1,answer exten => s,n,System(/tmp/check.sh) check.sh: check EPOCH time => do an IF for certain times => Allow mutiple calls in certain times and

Hello, I have a nasty loop that I have to do 11877 times. The only thing that slows it down really is this merge: xx1 = merge(dt,ua_rd,by.x=1,by.y= 'rt_date',all.x=T) Any ideas on how to speed it up? The output can't change materially (it works), but I'd like it to go faster. I'm looking at getting around the loop (not shown), but I'm trying to speed up the merge first.

I have a macro and when I call it I have something like this: exten => s,1,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) exten => s,n,Set(GROUP()=MYGROUP) ;Set Group exten => s,n,NoOp(Group List: ${GROUP_LIST()}) exten => s,n,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) The GROUP_COUNT returns zero before the call to GROUP but also returns 0 after the call to GROUP. If I

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels

Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to these SIP trunks with a round robin distribution pattern. I was thinking of using a group count

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the