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Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

...sion for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC handset and it is useless, and as far as I > can tell they cannot be re-flashed. > > Cheers, > Steve > > > > On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N <gopalakrishnan.an at gmail.com> > wrote: > > Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have > the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to > upload woth TFTP due to some reason it's getting failed. Do I need to load...

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ= et=3D"_blank">gopalakrishnan.an at gmail.com</a>&gt;</span> wrote:<br /> <blockquote class=3D"gmail_quote" style=3D"border-left: #c...

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Its really weird working with OpenSuse. I am not sure how others are using > with OpenSuse. Through Yast also I tried to install Asterisk package, it > didn't find. > > Now I am clueless to work with OpenSuse. > > > &gt...

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to upload woth TFTP due to some reason it's getting failed. Do I need to load 3pcc firmware or anyway to Configure from the phone itself or from the GUI? I have the SEPMAC.cnf.xml as well. Any suggestions would be appreciated. Regards .

...hat model of phone. > > I previously had experience of upgrading the Cisco build to the SIP build > on Cisco 7641 handsets, which have 2 similar builds, but none of the > techniques seemed to apply this time around. > > Cheers, > Steve > > > On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N <gopalakrishnan.an at gmail.com> > wrote: > > Can't I upload the 3PCC firmware ? available from the Cisco website? > > Actually it came with sip88xx.... firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com&gt...

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

Hi, How to create dialplan restriction for particular extensions.. -- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100917/a4bc96f6/attachment.htm

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

...ke to have a generic code, say if I specify the distribution with the parameters and simulation and sample size I would like to have the simulations done for the mentioned distribution and the statistics performed. I would appreciate any help in doing so? Thanks for your time. Mathangi Mathangi Gopalakrishnan Graduate student Dept of Mathematics and Statistics University of Maryland, Baltimore County Baltimore, MD

What happens when we increase the queue frame size in channels.c if ((queued_frames + new_frames > 128 || queued_voice_frames + new_voice_frames > 96)) { Be default it is 128 and 96 if i increase it to 256 and 192 what will happen? will it impact to default behavior? Regards, Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Am using Asterisk 11.2 in one location and 11.1 in another location. when I trunk between two servers, the status is unreachable. But with different server with 11.2 and 11.2 it works fine. I tried both IAX and SIP. the trunk in sip.conf what i have is, [serverb] type=friend username=serverb secret=serverb host=10.10.10.5 port=5060 context=default insecure=port,invite dtmfmode=rfc2833

Suddenly my asterisk restarted automatically and came up in seven seconds, While checking core dump I see some message related to snmp. No symbol table info available. #5 0x00007fc7e6249faa in agent_thread (arg=<value optimized out>) at snmp/agent.c:206 __PRETTY_FUNCTION__ = "agent_thread" #6 0x000000000056dd0b in dummy_start (data=<value optimized out>) at utils.c:1028

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be