search for: ahmedmunir007

Displaying 16 results from an estimated 16 matches for "ahmedmunir007".

2016 Oct 13
2
Openfile Issue
...remove the >> file 90-nproc.conf as well as add to the Asterisk init script: >> ulimit -s unlimited >> ulimit -n 65535 >> ulimit -Hn 65535 >> ulimit -u 65535 >> ulimit -Hu 65535 >> >> >> >> On Thu, Oct 13, 2016 at 9:59 AM, Ahmed Munir <ahmedmunir007 at gmail.com> >> wrote: >> >> > >> > See below; >> > >> > [root at abc asterisk]# cat /proc/50771/limits >> > Limit Soft Limit Hard Limit >> Units >> > Max cpu time unlimited...
2012 Aug 15
1
Send Fax from Asterisk
...end fax to destination number just like '.call' file. Does anyone worked on this scenario? If yes/no, please let me know at earliest. please check it. might be it will help > > http://ictfax.org/content/installation-guide > > On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir <ahmedmunir007 at gmail.com > >wrote: > > > Hi, > > > > I would like to know, anyone who worked in Email to Fax scenario? If so > > please share the idea for implementing it. > > > > As on other hand I configured Asterisk for inbound Fax which is working > > goo...
2011 Sep 02
0
No subject
...t;/code></div> </div> -dbc<br /> <br /> <blockquote cite=3D"mid:mailman.7.1348074001.2526.asterisk-users at lists.digi= um.com" type=3D"cite"> <pre>Message: 1 Date: Tue, 18 Sep 2012 15:41:46 -0400 From: Ahmed Munir <a href=3D"mailto:ahmedmunir007 at gmail.com" class=3D"moz-t= xt-link-rfc2396E">&lt;ahmedmunir007 at gmail.com&gt;</a> Subject: [asterisk-users] Trigger Asterisk after data inserted in mysql To: <a href=3D"mailto:asterisk-users at lists.digium.com" class=3D"moz-txt-lin= k-abbrevi...
2013 Jan 15
4
Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable
Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =
2012 Feb 01
2
Getting one way audio even NAT is configured
Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13
2016 Sep 27
2
Asterisk Radius CDR
I did radius client status testing with radius server, able to access the radius server. However, still getting radius CDR issue after setting debug level 8 even granting 666 access to radiusclient-ng config files. message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded! Please advise if I missed out anything. Date: Mon, 26 Sep 2016 12:09:34 +0200 > From:
2016 Sep 28
3
Asterisk Radius CDR
...gt; that file and recompiling. I think in my case it was as simple as a > > hostname not resolving. Once you're not working blind, you'll find what > is > > happening pretty quickly. > > > > Andrew > > > > On 28 September 2016 at 03:32, Ahmed Munir <ahmedmunir007 at gmail.com> > wrote: > > > > > I did radius client status testing with radius server, able to access > the > > > radius server. However, still getting radius CDR issue after setting > debug > > > level 8 even granting 666 access to radiusclient-ng conf...
2020 Jun 25
1
Asterisk Getting Crashed
Hi, Currently I'm experiencing crashes on Asterisk more recently, see messages below (crashed reason: segfault signal 6). abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT - dumping core asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fbd2c 00d170 (0) After running the backtrace for the
2015 Jan 06
0
Participant unable to hear other participants in ConfBridge
Hi All, The issue appearing at the random for confbridge module i.e. in some cases if a participant joins the confbridge, he/she unable to hear others which make him/her to hangup the call and redial the bridge again. By joining the bridge second time, participant able to hear the other participants. Any ideas which may causing this issue? As Asterisk version I'm using is 11.2.1. Is it a
2012 Jun 15
0
Getting Error: 3RD_T2_TIMEOUT while using T38 on Asterisk 10
Hi, I'm getting error: ' FAX session '9' is complete, result: 'FAILED' (FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution: '204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax more than 2 pages to Asterisk using T.38. First I set speed rate to 14400 which I was getting same error message while
2014 Mar 24
0
Getting T.38 issue
Hi, Few months back I configured Asterisk 11.6.0 for an outbound fax using T.38 protocol as listing down the flow below; Asterisk Fax server -> (IP) -> Cisco VGW ->(IP) -> Carrier The issue I'm currently getting when Asterisk receives warnings as listed below, it is overloading the Cisco VGW, therefore need to restart Asterisk service or sometimes reboot VGW to clear these
2013 Mar 29
0
Getting Unknown Error while configuring Asterisk with Linux HA
Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at
2019 Oct 22
2
Realtime PJSIP max_streams' issues
Hi, I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For configuration perspective, I'm pretty much done with it but here the real issue I'm currently facing i.e. when setting parameters max_audio_streams & max_video_streams to any positive greater than 0 integer value in realtime (DB) of any endpoints. After running command "pjsip show endpoint
2013 May 21
4
Asterisk Log rotate not working
Hi, Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily basis which was working perfect. Now in couple of months back, the logrotate feature is not working at all but simply appending the logs in 'messages' file. Listing down down the configuration for logrotate below; /var/log/asterisk/messages { missingok rotate 5 daily postrotate /usr/sbin/asterisk -rx 'logger
2013 Feb 27
3
Getting compilation error while installing Dhadi
Hi all, I'm getting compilation error as trying to install latest version of dahdi on CentOS box 5.9 which I now updated from 5.6. I also installed the dependencies but still not getting the clue to get install the driver. Listing down the errors below; CC [M]
2012 Jan 04
2
asterisk -> AGI (perl) -> sqlplus (oracle)
Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get