similar to: Monitoring SIP Service (Jai Rangi)

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

Hi Joshua HTTP is used on in our setup on 127.0.0.1/mxml?<command> to send commands to the server, such as http://127.0.0.1/mxml?action=login&username=myuser&secret=thesecret to log in and then http://127.0.0.1/mxml?ActionID=123&Action=BlindTransfer&Channel=Channel&Context=local&Exten=123&Priority=1 etc. to control transfers, for example. ARI is not being

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Hi all I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - 32GB RAM. When I approach about 320 channels, I -sometimes- get thousands of these messages suddenly streamed in the CLI / Asterisk log: WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length queuing to Local/xxxxxxxxxx at local-0002dbea;2 WARNING[71993][C-00022dcc] channel.c: Exceptionally

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script:

Hi all I'm trying to limit the maximum concurrent calls on my Asterisk to try and mitigate another problem I posted about earlier. I've edited /etc/asterisk/asterisk.conf And uncommented this line, and put a value of 60 in there: maxcalls = 60 in an effort to limit my Asterisk to 60 simultaneous calls. I did a core reload in the CLI after doing that.

I'ld like to test SMSes between Asterisk and an analog phone on a GS-ATA-286 (SIP). I have spent many hours trying any exemple on voip-info or mailing-list, but no message got sent from the analog phone. My goal, at now, is to send SMSes locally from and to Asterisk. Does anyone have a working configuration ?

Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140110/18f3f1e2/attachment.html>

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

Hi all We've had a very strange failure on an Asterisk 1.8 install that has been running for about a year at a customer site. The physical hardware is fine, all other services off the Centos 6.5 server are running. Only Asterisk is not working... The first symptom was that no calls can be made over the SIP phones used with it, and no calls could be received over the SIP trunk connected to

Digital ocean offers ssd on all the virtual machines. Uptime is good. Jai Rangi Www.didforsale.com www.cebodtelecom.com www.cebod.com > On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere <jeff at jeff.net> wrote: > > > Amazon instances are shared resources. I wouldn't want to count on timing or disk throughput, and you can't just ask them to do "ssd" - its a

hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central.... roy

Well, I had exactly the same issue as you described. It turned out to be a piece of malicious software that was running on the server. The customer server was compromised due to a weak root password and only Asterisk process was the target of the malicious program that was embedded deep into the server. The exact details escape me, but I do remember that it took more than two days of tracing

Hi all I have a strange issue with 1.8.11.0 on a production Asterisk machine at our head office, and the same issue with a production machine at a branch office. Every now and then, on the head office machine, ODBC CEL and CDR logging will stop working. On examination in the CLI, Asterisk behaves as if the config files for ODBC in the /etc directory are just gone. Repeated tests have then

I am trying to do something in R and would appreciate a push into the right direction. I hope some of you experts can help. I have two distributions obtrained from 10000 datapoints each (about 10000 datapoints each, non-normal with multi-model shape (when eye-balling densities) but other then that I know little about its distribution). When plotting the two distributions together I can see that