search for: jprangi

Displaying 12 results from an estimated 12 matches for "jprangi".

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2014 Jan 10
4
Text to Speech Engine
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>
2008 Nov 06
2
Spam from DIDForSale <contact-sales@didforsale.com>
didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. Gordon
2007 Dec 13
0
Didnt get a frame from Channel and call gets
...ing?? ( E1 or FXO card) sip.conf, zapata.cons, zaptel.conf config details?? Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased ================== Message: 5 Date: Mon, 10 Dec 2007 15:26:52 -0800 From: "Jai Rangi" <jprangi at gmail.com> Subject: [asterisk-users] Didnt get a frame from Channel and call gets disconnected To: asterisk-users at lists.digium.com Message-ID: <eb007ec0712101526x493e0e25r9e09bcc1ebc9e9ee at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello...
2009 Nov 07
6
Location
Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm
2013 Jun 22
2
SIP Trunking Mantra (Origination)
Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP
2017 Apr 19
4
PBX selection
The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but
2015 Dec 23
7
Best Asterisk Platform
What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment
2013 Jan 09
13
DIDForSale spam
List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer
2015 May 19
0
Monitoring SIP Service
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx owners, How can we monitor asterisk, what happens if service stop responding. Here is a small howto on monitoring asterisk with nagios. I am sure there are plenty of options and suggestions, but this is one of them and has been working out very well for us for years. http://www.didforsale.com/monitor-sip-server Best, -Jai
2007 Dec 10
1
Didnt get a frame from Channel and call gets disconnected
Hello, Since last few days I have noticed some people complaining that their call gets disconnected while they are in the middle of the conversations. Looking in the log I found these error messages, Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:41 DEBUG[10410] channel.c: Didn't get a frame from channel:
2009 Sep 05
2
Need some help/Suggestions for multiple invites from Asterisk
Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222
2015 Mar 08
2
AWS/EC2 server selection
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