Re: [asterisk-users] PBX selection
t; than LAME > > MP3 encoder, but easier than the Linux kernel. If you altered > > `monop` from > > the BSDgames package to make the streets match your local edition of > > the game, > > you will have no problem whatsoever with building Asterisk. > > > > If you understand the process of what you are doing -- basically, > > setting up > > an automated process that will examine your server hardware and > > software > > configuration (configure), choosing which parts of Asterisk you > > want to > > include (make menuselect), compiling the selected human-readable > > Source Code > > into binary code that the computer can understand natively (make) > > and then > > moving the compiled binary code and configuration files from the > > Source Code > > folder to where the computer is expecting for them to be (make > > install) then > > you should not have too many problems. > > > > It is always preferrable to compile your own Asterisk to fit your > > hardware and > > include just the bits you want, rather than rely on anyone else's > > pre-compiled > > package. > > > > > 4. Which Asterisk version is recommended? > > > > The latest one. > > > > > And does Asterisk support Windows > > > ? > > > > You can certainly use Windows softphones to talk to Asterisk, but > > Asterisk > > itself requires a non-toy underlying operating system. Ubuntu and > > CentOS are > > the best-supported Linux distributions. Asterisk has also been seen > > working, > > to greater or lesser extents, on Solaris and the BSDs. But Linux was > > the > > original development environment (although one of the two original > > projects > > that ended up merging and becoming Asterisk, many years ago, was > > originally > > developed on FreeBSD), and is what most Asterisk telephonistas know. > > > > Any hardware which is capable of running Windows can, of course, run > > Linux; > > and usually better. > > > > -- > > JM or AJS > > > > Note: Originating address only accepts e-mail from list! If > > replying off- > > list, change address to asterisk1list at earthshod dot co dot uk . > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk community forum at: > > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <javascript:void(0);> | C 949-419-7634 <javascript:void(0);> | F 949-269-0449 / 949-232-1410 | jpra...@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Asterisk Platform
Eric, So far you have seen lot of suggestions. PBX in a Flash, Elastix, FreePBX, astlinux, Trixbox, ASteriskNOW. Different people like different flavors based on their choice, ease of use and needs. My 2 cents will be to pay more attention to security of the platform. I have seen customer's PBX being hacked all the time. MySQL injection through forms and urls is the most common hack. Some of these flavors package lot of different tool in the same PBX, which sounds really cool, but the challenge will be to find if all the tools and packages are secure or not. For example Elastix, PBX part might be secure, but the vtiger CRM might have a security hole it. Read the forums for each package and see if there is any issue. So being in industry for more than 7 years, If I were you, I will go with most secure open-source platform and modify GUI part based on my needs. *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jpra...@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Wed, Dec 23, 2015 at 5:31 AM, er ic <email.eherr9...@gmail.com> wrote: > 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 j...@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 virtual machine! 500 simultaneous recordings is a hefty load, and I would want to know that the underlying hardware is dedicated to the task. Sure you see lots of posts about hosting asterisk and/or freeswitch on EC2. I have done it myself and even have some clients doing it now *for proof of concept*. I've never heard of anyone using it for the kind of load you are talking about. I'm assuming with such a giant load you are making a decent profit. Buy some hefty hardware and do the architecture properly. You can rent half a rack at lots of high end datacenters for less than $1000/month. j On 03/07/2015 12:43 AM, Amit Patkar wrote: Hi Jeff Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. Thanks Regards, Amit Patkar On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote: Why use Amazon? With that kind of load I would want dedicated servers. Call Rackspace or Softlayer. j On 03/06/2015 11:59 AM, Amit Patkar wrote: Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that server with ssd is required as all 500+ calls needs to be recorded. Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AWS/EC2 server selection
Agreed, network will be bottleneck even with ssd on shared resource. For a stable env having a dedicated hosted server will be the best approach and cheaper too. Jai Rangi Www.didforsale.com www.cebodtelecom.com www.cebod.com On Mar 8, 2015, at 9:10 AM, Jeff LaCoursiere j...@jeff.net wrote: Still a shared resource. I don't see the benefit. Even beyond the shared resource bit, with the kind of IO you are likely to be pushing, you will want a decent NAS with lots of spindles and fibre channel to your hosts. j On 03/08/2015 10:51 AM, Jai Rangi wrote: 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 j...@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 virtual machine! 500 simultaneous recordings is a hefty load, and I would want to know that the underlying hardware is dedicated to the task. Sure you see lots of posts about hosting asterisk and/or freeswitch on EC2. I have done it myself and even have some clients doing it now *for proof of concept*. I've never heard of anyone using it for the kind of load you are talking about. I'm assuming with such a giant load you are making a decent profit. Buy some hefty hardware and do the architecture properly. You can rent half a rack at lots of high end datacenters for less than $1000/month. j On 03/07/2015 12:43 AM, Amit Patkar wrote: Hi Jeff Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. Thanks Regards, Amit Patkar On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote: Why use Amazon? With that kind of load I would want dedicated servers. Call Rackspace or Softlayer. j On 03/06/2015 11:59 AM, Amit Patkar wrote: Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that server with ssd is required as all 500+ calls needs to be recorded. Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attack on Sip server.
Anurag, Here is small script, that will check your logs and will block the IPs. http://www.didforsale.com/blog/is-your-asterisk-system-under-heavy-attack This is good if you dont expect any registration. If you do have some valid registration, you might want to add some counter to see how time IP need to fail or how many different users IP is trying to register on before blocking the IP. Jai Rangi www.didforslae.com On Fri, Jun 27, 2014 at 7:37 AM, Anurag Rana anuragrana31...@gmail.com wrote: Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the packets from attacker contains string 'VaxSIPUserAgent/3.1' ) but it failed. Packets are still flowing in. iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent --algo bm -j DROP Its something like this Registration from '30 sp:30@my_public_ip:5060 failed for '192.168.xxx.xxx:6373' - Wrong Password and there are approx 10 request per minute of this type. Please suggest some way to stop this. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
I can vouch for newfies, but its not asterisk and there is some learning curve, but comes with lots of features. -Jai www.didforsale.com SIP Trunking Simplified On Tue, Apr 22, 2014 at 2:54 PM, Nick Cameo sym...@gmail.com wrote: Hello Everyone, Thank you all for your response. The people I am doing it for run a non-profit charity, and are legally able to reach out to their customers. I will wire it up to the DNC however, for starters, I would like to get asterisk to: i) Iterate through a list of numbers ii) Play a pre-recorded message asking if they have waste they need picked up iii) If they press one, forward the call to mailbox The easier the better for us. I did see Wombat, newfies, and vicdial however, I can't go through with the installation process and find out there is some hidden clause, limited to 2 channels etc If I can do it with a simple dialplan as mentioned earlier, I think it's the best solution for starters. Kind Regards, Nick from Toronto -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise VoIP Trunk
Gopal, This should have been on asterisk-biz list. You can try didofrsale.com. We can offer your 10,000+ rate centers all under same tier. Contact us offline to discuss further contact-sa...@didforsale.com On Wed, Mar 5, 2014 at 10:33 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
Thank you every one, Yes google's translate is really good. http://zaf.github.io/asterisk-googletts/ But I dont like the fact that have to go over the wire every time. Looking for some thing to install on local server. -Jai On Fri, Jan 10, 2014 at 5:15 PM, Darryl Moore dar...@moores.ca wrote: http://translate.google.com/translate_tts?tl=enq=i always find google translate works well http://translate.google.com/translate_tts?tl=frq=je trouve toujours google translate fonctionne bien On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monitoring Asterisk 1.8
There is nagios plugin check_asterisk_channels Examples: Check channels/calls, with no concern about limits. check_asterisk_channels Check channels/calls. Issue a warning if there are more than 10 active channels, and a critical if there are more than 15 active channels. check_asterisk_channels -w 10 -c 15 Caveats: This plugin calls the asterisk executable directly, so make sure that the user executing this script has appropriate permissions! Usually the asterisk binary can only be run by the asterisk user or root. To grant the nagios user permissions to execute the script, try something like the following in your /etc/sudoers file: nagios ALL=(ALL) NOPASSWD: /path/to/plugins/directory/check_asterisk_channels You can easily edit this to add more monitoring Jai Rangi On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
Guys, Since I am attached to did for sale: My apology to every one who received the DIDForSale 2012 Achievement email and you hated it. As a asterisk user my question will be. If some xyz company send you a so called spam email, what made you think that you should spam the mailing lists. I am sure we all get lots if spam emails every day. If you really got some time and talent, why don't you write some good tips and tricks about asterisk. Long story short We have a link where you can unsubscribe your email for any further communication. http://www.didforsale.com/unsubscribe.php or Send me your email address I will personally take care of that and will remove your email. This will take less than 5 seconds. I am sure there will be lot of arguments on why you should that and all. I will refrain myself on any further unproductive communication. Happy new year to you all. On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote: +1 here. On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote: On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have gotten hit with this twice so far. in March and Today: Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13 UGH, when I asked in March where he got my email he said: Hi Chris, We got your contact from the Internet. Let me know the good time to talk about this in detail. Thank you, -Rohit Dhaka Obviously by harvesting these lists. I received 2 myself. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
Don, I have removed yours right away. Yes, I agree, But just like any company we have purchased/collected email from different source. Also just like any company we are not perfect, we make mistakes. -Jai Rangi On Wed, Jan 9, 2013 at 5:57 PM, Don Kelly d...@donkelly.biz wrote: Jai, ** ** It should not be necessary for me to remove my email address from your list. It should not be on there to start with—we do not have, and have never had, a relationship that justified you sending me email. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jai Rangi *Sent:* Wednesday, January 09, 2013 7:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] DIDForSale spam ** ** Guys, Since I am attached to did for sale: My apology to every one who received the DIDForSale 2012 Achievement email and you hated it. As a asterisk user my question will be. If some xyz company send you a so called spam email, what made you think that you should spam the mailing lists. I am sure we all get lots if spam emails every day. If you really got some time and talent, why don't you write some good tips and tricks about asterisk. Long story short We have a link where you can unsubscribe your email for any further communication. http://www.didforsale.com/unsubscribe.php or Send me your email address I will personally take care of that and will remove your email. This will take less than 5 seconds. I am sure there will be lot of arguments on why you should that and all. I will refrain myself on any further unproductive communication. Happy new year to you all. On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:*** * +1 here. On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote: On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have gotten hit with this twice so far. in March and Today: Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13 UGH, when I asked in March where he got my email he said: Hi Chris, We got your contact from the Internet. Let me know the good time to talk about this in detail. Thank you, -Rohit Dhaka Obviously by harvesting these lists. I received 2 myself. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable SIP Trunk Provider
Jake, We are DIDForSale support asterisk. We do IP based authentication and do not require registration. You can test our DIDs without paying anything. I am sending you the rates just to make it easy to compare apple to apple, no run around for pricing ;) . Let me know if I can have an opportunity to earn your business. We have been in business for 5 year and handle traffic for few of top calling card companies. So have very stable environment. Thanks for your interest in our service. Our Product offerings: We sell DIDs all over US. For the list of all the available rate centers please visit us at http://www.didforsale.com/moreinfo.php?help=ratecenter. Please contact Sangeeta at +19494568787 for details. We have inbound DIDs in 4 different configurations. 1) Flat Rate DID with 20 channels ($8.99 per DID per month + $5 Activation per DID)* #of DIDs Rate (USD)/DID 1-30 $8.99 31-100 $8.75 101-200 $8.50 201+ Contact us http://www.didforsale.com/contactus.php - NOTE: - Average Limit of 8000 inbound minutes, per DID per month. - Overage will be charged at $0.008 per minute. 2) DID with metered inbound (Pay as you go) $1 per DID + $5 activation per DID and 0.4 cents ($0.004) per minute for all incoming calls. 3) Toll free numbers $3 per month and 1.9 cents per minute. 4) Channelized Option Inbound: For high usage customers, like Calling Card, Call Centers, Conferences etc. #of Channel’s/Month Rate (USD)/Channel Rate (USD)/DID Rate (USD)/DID 1-30 $8.99 1-200 $0.99 31-100 $8.75 201-500 $0.85 101-200 $8.50 501-1000 $0.70 201+ Contact us http://www.didforsale.com/contactus.php 1000+http://www.didforsale.com/contactus.php Contact us. http://www.didforsale.com/contactus.php - Note: - DID/Channel ratio can not be more that 25. Example, for Every 25 DIDs you must have 1 channel. - Activation fee $1/DID. 5) Outbound is all metered The rates depends on the volume of commitment. For US rates start from 1.9 cents per minute. 6) Porting a Number We can port a number if it exists in the rate centers. It costs additional $10 fee. Before you buy our DID you can test our service for free. Free trial does not require you make any payment or purchase. Please follow these steps to reserve a test DID for free trial for 6 hours:- a) Signup to create an account on our website. Your login id is the email id that you created the account with. b) Login into your account and Click on Testing Center link on the Left hand side menu. Select the DID to test. To know what our customer are saying about us, please click on the link, http://www.didforsale.com/blog/2011-didforsale-customer-satisfaction-survey-resultshttp://www.didforsale.com/blog/?p=103 Please let me know in case you need any more information. Thanks Regards, www.didforsale.com On Fri, Mar 16, 2012 at 10:10 AM, white hat whitehat...@gmail.com wrote: I had many of the same problems with sip station. If you just need sip termination, Check out flow route. The service just seems to work properly for me, and they respond to tickets. You can open up new cases through their site. On Mar 15, 2012 11:48 AM, Jake Wicke j...@nxtphase.net wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
Re: [asterisk-users] Reliable SIP Trunk Provider
www.didforslae.com have wide range of products to fit low usage to very high usage. Dont want to put too much details here. Check it out let me know if interested, since you are using I will help you waive activation fee. -Jai On Thu, Mar 15, 2012 at 11:30 AM, Guy Gold g...@the-golds.us wrote: On Thu,Mar 15 12:10:PM, Eric Wieling wrote: I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. +1 for Vitelity , I like them for recognizing the fact that some people actually prefer to run pure Asterisk (no GUI) . Guy Gold -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resell VoIP Servcies
I am sorry. Meant to send to biz list. Thank you for correcting me. On Tue, Nov 22, 2011 at 5:57 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 11/22/2011 08:14 AM, Jai Rangi wrote: [removed commercial offer] You posted to the wrong list. The correct list for commercial business related discussion is asterisk-biz. Please do not spam the asterisk-users list again with your commercial offers. Regards, Patrick -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resell VoIP Servcies
Make money while helping others to enjoy great VoIP Services and huge savings on inbound SIP Trunking. There is no limit to how many friends and business partners you can refer. The more friends you refer, the more money you can make. Just have your friend send us an email that he was referred by you and he will save upto 10% of his 1st month bill spend and in addition you will get upto 13% of his 1st month spending. This offer expires on 12/31/11. Service purchased Between $25-$50, you get 5% and your referral get 5%. Service purchased Between $50-$100 you get 7% and your referral get 6%. Service purchased Between $100-$500 you get 10% and your referral get 8%. Service purchased Between $500-$1000 you get 13% and your referral get 10%. Credit applied on 1st month spending only. We will apply the credit either to your account and payment will be made by Paypal to your paypal account. You don't need to be our customer to refer our service. NOTE: Refer a friend can only be used to refer new customers (who have never purchased service from DIDforSale) and cannot be used for an existing customer, your direct family member, yourself or some one living at the same address. This would be considered a fraud and we reserve the right to refuse referral credit to you and your friend. Thank you, www.didforsale.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP
Re: [asterisk-users] One Way Audio
You can use this link too. http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale Keep the context as context=from-trunk. -Jai On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote: 209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172
Re: [asterisk-users] FW: Under heavy attack
Asterisk security has always been a big concern. I am sure most of asterisk pros have taken care of these type of attacks. For non pros I am sharing a shell script here. http://www.didforsale.com/blog/?p=253 If you care feel free is use it. -Jai On Tue, Nov 2, 2010 at 9:27 AM, Cary Fitch ca...@usawide.net wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Tuesday, November 02, 2010 10:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Under heavy attack I'm still on old-fashion copper-wire and have yet to experience the joy of SIP Trunk-ing and the type of issues discussed in this thread. My thought to share here is that outgoing calls should be easy for thoroughly authenticated users and impossible for others... Probably more can-o-worms than help. Sorry if this is so. nothing new here, this is just the digital equivalent of a wats line with a weak access code for outbound access. the difference is code guessing can be a lot more aggressive now, and finding the inbound path is simpler. == Each system needs to be configured according to its purpose and needs. Simply these are phone systems, not e-mail or web servers. You may want to be able to get mail from (almost) anywhere in the world, same for web services. But for a phone system you may have very different needs. One can visualize the differences between a national or international VOIP provider, a 4 person office in Little Rock, AR, a local SIP provider in Houston, TX and an international sales company with offices in Rome Italy. A small sip system used with an upstream VOIP provider should be invisible to 99.% of the world's population. (Excepting any other trusted peers.) If there was a wide spread peering network and an individual system wanted/needed to access and be accessed like email then it would be a different world. We could all be robo-call spammed just like email. :-( But leaving small systems open for attack from 99. percent of the world is just begging for trouble. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High Availability Asterisk PBX
Dns srv might be the solution for you. Jai www.didforsale.com --Original Message-- From: RESEARCH Sender: asterisk-users-boun...@lists.digium.com To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] High Availability Asterisk PBX Sent: Mar 14, 2010 10:27 PM Hi I have the following scenario A. A PBX on location A with network 192.168.1.1 with extension range 1XXX and connected to the PSTN Network via the E1 B. Another PBX on location B with network 172.30.18.1 with extension range 2XXX and connected to the PSTN Network via the E1 I need to configure the system and the endpoints such that when one system, says, A goes down, the system B assumes A responsibility. HALinux would have been my answer but this should work only on the same subnet Any advice Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jai Rangi 1949 419 7634 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
I think the key point is how many calls per second. That's what mysql is concerned about. Other than that it is just asterisk. Did you monitor the mysql, try log-slow-queries and set the time to 1 second. -Jai On Wed, Oct 21, 2009 at 12:57 PM, das sandesh sandesh...@gmail.com wrote: Hi Steve, Thanks for your reply. I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? Regards Sandesh. On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
The thing is, concurrent calls won't make any difference, it's the calls per second. And really you're unlikely to use too many queries per sec. Exactly and you can see the slow-log-queries if mysql is taking time. -Jai On Wed, Oct 21, 2009 at 3:51 PM, Matt Riddell li...@venturevoip.com wrote: On 22/10/09 10:57 AM, das sandesh wrote: Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? The thing is, concurrent calls won't make any difference, it's the calls per second. And really you're unlikely to use too many queries per sec. Seriously, use at least AGI (fastAGI would be better but AGI will at least give you a start). So: 1. Do you get the same delay if you use MySQL command line at the same time? 2. Do you have a programming language you know well enough to connect to MySQL in? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Monitoring
Nagios has a plugin check_sip that can be used for this. -Jai On Sat, Oct 17, 2009 at 5:30 PM, Dan Journo d...@keshercommunications.comwrote: 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 -- [image: See original image] *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 information on receiving IT support from £150 per month, please contact Kesher Communications. -- [image: sig] *Dan Journo **IT Business Consultant* Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.comhttp://www.keshercommunications.com/ Live Chat/Instant Support: Click Herehttp://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image002.gifimage001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some help/Suggestions for multiple invites from Asterisk
Thank you for your response, But we do get response from client (Step 2,3,4), the call is good, audio DTMF everything works, except CDR is wrong; always 30-60 seconds more for each call. 2 0.042380 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson o...@edvina.net wrote: 5 sep 2009 kl. 04.58 skrev Jai Rangi: 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 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23 sip%3a1978525...@192.168.4.23 , with session description 2 0.042380 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 5 0.046752 192.168.3.222 - 192.168.4.23 SIP Request: ACK sip:1978525...@192.168.4.23:5060 So far so good, call is established and audio conversations starts. But next my asterisk is sending Invite again and again and again, 6 0.047036 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 7 0.266230 192.168.3.222 - 192.168.4.23 RTP Payload type=ITU-T G.729, SSRC=905761218, Seq=56540, Time=0 8 1.046087 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 9 2.046091 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 10 4.046102 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description I disconnected the call, Receive BYe from Vendor, Asterisk acknowledge Bye and does not send Bye to the client. Few more invites from Asterisk to the client machine. 11 8.046123 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 12 16.046179 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description After a 30 second wait, asterisk receive Bye from Client. 13 24.253811 192.168.4.23 - 192.168.3.222 SIP Request: BYE sip:6056929...@192.168.3.222 sip%3a6056929...@192.168.3.222 14 24.253975 192.168.3.222 - 192.168.4.23 SIP Status: 200 OK 15 32.046319 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 16 32.085897 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 17 32.090654 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 18 32.092666 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 19 32.593335 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 20 33.607552 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description I am using canreinvite=yes, (Must use that to avoid media going through my asterisk server. I dont have any issue if asterisk send call to another asterisk box. Can some one please shed some light why asterisk is sending multiple invites. There's no response from the client phone. No 100 trying, no 180 ringing or 200 OK. We have to retransmit a few times and then just give up. Your client needs to wake up and start responding. Since the client was not responding, there never was a call to the client and no need to send a BYE. /O --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some help/Suggestions for multiple invites from Asterisk
But this is my questions why it is sending invites again in 6-10 when the call is already established. -Jai On Sat, Sep 5, 2009 at 3:22 AM, Olle E. Johansson o...@edvina.net wrote: 5 sep 2009 kl. 09.06 skrev Jai Rangi: Thank you for your response, But we do get response from client (Step 2,3,4), the call is good, audio DTMF everything works, except CDR is wrong; always 30-60 seconds more for each call. In step 6-10, there's no reply from the client, unless you missed something. Turn on SIP debug and you'll see that Asterisk will time out and give up about the call. /O 2 0.042380 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson o...@edvina.net wrote: 5 sep 2009 kl. 04.58 skrev Jai Rangi: 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 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23 sip%3a1978525...@192.168.4.23 , with session description 2 0.042380 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 5 0.046752 192.168.3.222 - 192.168.4.23 SIP Request: ACK sip:1978525...@192.168.4.23:5060 So far so good, call is established and audio conversations starts. But next my asterisk is sending Invite again and again and again, 6 0.047036 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 7 0.266230 192.168.3.222 - 192.168.4.23 RTP Payload type=ITU-T G.729, SSRC=905761218, Seq=56540, Time=0 8 1.046087 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 9 2.046091 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 10 4.046102 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description I disconnected the call, Receive BYe from Vendor, Asterisk acknowledge Bye and does not send Bye to the client. Few more invites from Asterisk to the client machine. 11 8.046123 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 12 16.046179 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description After a 30 second wait, asterisk receive Bye from Client. 13 24.253811 192.168.4.23 - 192.168.3.222 SIP Request: BYE sip:6056929...@192.168.3.222 sip%3a6056929...@192.168.3.222 14 24.253975 192.168.3.222 - 192.168.4.23 SIP Status: 200 OK 15 32.046319 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060 , with session description 16 32.085897 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 17 32.090654 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 18 32.092666 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 19 32.593335 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 20 33.607552 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description I am using canreinvite=yes, (Must use that to avoid media going through my asterisk server. I dont have any issue if asterisk send call to another asterisk box. Can some one please shed some light why asterisk is sending multiple invites. There's no response from the client phone. No 100 trying, no 180 ringing or 200 OK. We have to retransmit a few times and then just give up. Your client needs to wake up and start responding. Since the client was not responding, there never was a call to the client and no need to send a BYE. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
[asterisk-users] 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 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23 sip%3a1978525...@192.168.4.23, with session description 2 0.042380 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 5 0.046752 192.168.3.222 - 192.168.4.23 SIP Request: ACK sip:1978525...@192.168.4.23:5060 So far so good, call is established and audio conversations starts. But next my asterisk is sending Invite again and again and again, 6 0.047036 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description 7 0.266230 192.168.3.222 - 192.168.4.23 RTP Payload type=ITU-T G.729, SSRC=905761218, Seq=56540, Time=0 8 1.046087 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description 9 2.046091 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description 10 4.046102 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description I disconnected the call, Receive BYe from Vendor, Asterisk acknowledge Bye and does not send Bye to the client. Few more invites from Asterisk to the client machine. 11 8.046123 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description 12 16.046179 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description After a 30 second wait, asterisk receive Bye from Client. 13 24.253811 192.168.4.23 - 192.168.3.222 SIP Request: BYE sip:6056929...@192.168.3.222 sip%3a6056929...@192.168.3.222 14 24.253975 192.168.3.222 - 192.168.4.23 SIP Status: 200 OK 15 32.046319 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE sip:1978525...@192.168.4.23:5060, with session description 16 32.085897 192.168.4.23 - 192.168.3.222 SIP Status: 100 Trying 17 32.090654 192.168.4.23 - 192.168.3.222 SIP Status: 183 Session Progress 18 32.092666 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 19 32.593335 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description 20 33.607552 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK, with session description I am using canreinvite=yes, (Must use that to avoid media going through my asterisk server. I dont have any issue if asterisk send call to another asterisk box. Can some one please shed some light why asterisk is sending multiple invites. Best, -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIDForSale July Special (No Activation on new DID Purchases)
*All, To meet the target for the month, we are running a special promotion. $5 activation fee waived for all new DID purchases.* Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5 activation fees will be WAIVED* for all the DIDs purchased before July 20 2009. There is no limit on the number of DIDs you can buy and the offer is valid for all customers on new purchases only. We have inbound DIDs in 2 different configurations. 1) DID with unmetered inbound and 20 channels ($8.99 per DID). Additional channels can be purchased at $1 per additional channel. 2) DID with metered inbound are for $1 per DID and $0.004 (0.4 cents) per minute for all incoming calls. What our customer are saying about us, Please click on the link, http://www.didforsale.com/blog/?p=103 Thank you, www.didforsale.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale July Special (No Activation on new DID Purchases)
Sincere Apologies-- Send the mail to wrong list, Meant to send to asterisk-biz list. -J On Wed, Jul 8, 2009 at 11:35 PM, Jai Rangi jpra...@gmail.com wrote: *All, To meet the target for the month, we are running a special promotion. $5 activation fee waived for all new DID purchases.* Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5 activation fees will be WAIVED* for all the DIDs purchased before July 20 2009. There is no limit on the number of DIDs you can buy and the offer is valid for all customers on new purchases only. We have inbound DIDs in 2 different configurations. 1) DID with unmetered inbound and 20 channels ($8.99 per DID). Additional channels can be purchased at $1 per additional channel. 2) DID with metered inbound are for $1 per DID and $0.004 (0.4 cents) per minute for all incoming calls. What our customer are saying about us, Please click on the link, http://www.didforsale.com/blog/?p=103 Thank you, www.didforsale.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test
Yes, its working :) Jai Rangi ww.didforsale.com On Thu, Apr 30, 2009 at 12:12 PM, James A. Shigley j...@answeringserv.comwrote: Had an inbound email server issue, just double checking it is working again. James Shigley *Monroe Telephone Answering Service* 409-981-9213** Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein I know a little of everything, but a lot of nothing ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID's in a specific rate center
Vikas, www.didforsale.com can get you the DIDs, please contact me off list. Jai Rangi jpra...@didforsale.com On Wed, Feb 25, 2009 at 1:35 PM, Vikas topg...@gmail.com wrote: Since it's not clear from this thread of conversation, do you need 100 unique DIDs? I apologize for not being more clear. I need 100 DID's. I already have channels which allow me to set the outgoing caller id. Depending on which extension is making the call I will be sending out the unique caller id. So that the person receiving the call can call back directly to the caller id that they received on their phone instead of going through the IVR hell. Vikas On Wed, Feb 25, 2009 at 3:13 PM, M Hulber asterisk-ad...@hulber.com wrote: Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
** I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. ** We have some documentation and I can contribute that. Also we can provide the physical resources (Domain, Web hosting, bandwidth, storage, database etc). Ofcourse need a team with designated responsibilities. -Jai Rangi www.didforsale.com On Tue, Jan 27, 2009 at 1:16 PM, Robert Broyles rob...@poornam.com wrote: Jared Smith wrote: On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote: I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. Asterisk is truly an open-source community, and that pertains to documentation as well. The quality and quantity of the documentation depends heavily on contribution from the community at large. Digium has and will continue to put resources towards Asterisk documentation, but every contribution from the community at large helps. I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. (Unless one is willing to buy or read O'Reilly's Book -http://www.asteriskdocs.org - which quickly will be outdated again.) Alas, you've mentioned the one thing that both makes me happy and sad at the same time. Happy that people find it useful, and that O'Reilly was kind enough to let us publish it under a Creative Commons license (and put the PDF on the web for free!)... and sad that it takes so much time and effort to keep up to date. (And just for the record, the time that the other authors and I spend on writing the O'Reilly book is our own personal time -- I'm not working on it during company time!) This was an excellent read. I'm sad to say that I was one that didn't purchase the book, but made good use of the PDF. I was hoping to win one of the books during your sessions at AstriCon this past year. Too bad. :-( I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. If you're interested and serious about writing, join the asterisk-docs mailing list and let's try to get something started. I've been beating the documentation drum for almost seven years now, and I'd love to see the -docs mailing list come back to life. I'll be checking this out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] General Asterisk SIP/IAX provider question
Shane, You can try, www.didforsale.com. We allow free testing with no purchase required. See what others are saying, http://www.didforsale.com/blog/?p=103 -Jai On Mon, Jan 26, 2009 at 11:12 AM, Thomas Mullins tsmull...@wise.k12.va.uswrote: My coworker and I have built an Asterisk box. Everything went well, now we are ready to hook the box to a SIP/IAX provider. Does anyone have recommendation on choosing a vendor? We are located in Virginia. Thanks Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FWD and Asterisk
ngrep port 5060 or tcpdum port 5060 By default asterisk runs on port 5060, that way you can see if your getting the signal or not. Jai Rangi Buy SIP DID www.didforsale.com free Trial now purchase required On Tue, Jan 13, 2009 at 1:13 PM, David @ULC ucoms2...@gmail.com wrote: I also tried but cant see any call landing up in asterisk. Btw, how to find out whether a call is landing in Asterisk or not ? [123] type=peer qualify=no port=5060 nat=no insecure=very this is very important host=voiper.ipkall.com dtmfmode=rfc2833 context=from-pstn canreinvite=no ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FWD and Asterisk
Sorry for the typo, tcpdump port 5060 ngrep you can download the rpm (google) easy to install http://rpm.pbone.net/index.php3/stat/4/idpl/1127130/com/ngrep-1.38-1.i386.rpm.html rpm -ivh ngrep-1.38-1.i386.rpmftp://ftp.pbone.net/mirror/ftp.sourceforge.net/pub/sourceforge/n/ng/ngrep/ngrep-1.38-1.i386.rpm Is you sip configuration right? cant tell without looking at it. Jai Rangi Buy SIP DID www.didforsale.com free Trial no purchase required On Tue, Jan 13, 2009 at 1:44 PM, David @ULC ucoms2...@gmail.com wrote: [r...@vicidialnow ~]# ngrep port 5060 -bash: ngrep: command not found [r...@vicidialnow ~]# tcpdum port 5060 -bash: tcpdum: command not found [r...@vicidialnow ~]# Also, is my SIP configuration is correct ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What are the various models of DID providers
Alex, I must say wow, great explanation. It was a wonderful reading. Best, -Jai On Tue, Jan 13, 2009 at 1:49 AM, Alex Balashov abalas...@evaristesys.comwrote: Hi Randulo, I think this topic is probably more appropriate for asterisk-biz, as was the aforementioned rant about one particular DID provider. But, whatever - it is what it is. I assume that by DID providers you are referring to origination - that is, picking up calls on PSTN numbers and converting them to VoIP media and signaling and sending them to someone who wants to get numbers that ordinary PSTN users can call on a VoIP system of some kind. The reason for the disambiguation is that many DID providers also provide termination - that is, the delivery of calls from VoIP into the PSTN. There are also many companies that specialise in only origination or termination. The two are closely related from a technical perspective but are characterised by rather different economics. At the end of the day--on a technical and a regulatory level--telephone numbers can only belong to a carrier. A carrier is a network operator that is interconnected with other carriers and operates some form of switch, and usually interfaces via SS7 (or CSS7, as it is known outside North America) to the other carriers that they connect to. (Aside/digression about carriers: Of course, there are different types of carriers, depending on the jurisdiction. In the US, there are - broadly speaking - two different types: incumbents and competitive carriers involved in local service. Incumbents are either Bell system entities that were divested from the former ATT monopoly in 1984 when ATT was ordered to break itself up by the federal government, or various local-yokel independent telephone companies that were never acquired by ATT during the 20th century (as well as various types of conglomerates that have bought some of these independents before, or since divestiture). The latter type of incumbent is usually in small towns and/or rural areas, whereas the former is prevalent in metropolitan areas. The defining feature of an incumbent is that it tends to own the physical plant related to local telephone service delivery in a given area -- copper, fiber, central offices (telephone exchanges), remote terminals, junction boxes, conduit, and so on. That's why it's an incumbent. Examples of incumbents in the US include the former BellSouth (now ATT), Ameritech, Qwest, Southwestern Bell (now ATT), Verizon, GTE (now Verizon), and so on. Independent incumbents include something like Ellijay Telephone Company here in Georgia, or Windstream (formerly Alltel). This space has undergone a dizzying array of consolidation in the postmillenial years, so keeping accurate track of who is who even for pedagogical purposes is difficult. The Telecommunications Act of 1996 created local loop competition in the US and introduced the category of competitive carrier, or a CLEC (Competitive Local Exchange Carrier). These are carriers that can interconnect with the incumbent (and in fact, the incumbent is legally required to interconnect with them) and have the right to lease certain parts of the incumbent's infrastructure at regulated rates in order to provide subscriber services - this pricing and resale discipline is known as UNE (Unbundled Network Element) in the parlance. For example, a CLEC here in Atlanta in former BellSouth territory (now ATT) connects their network to BellSouth and can rent the copper going back to my residence from BellSouth and generate all the services, features and routing from its own equipment and use BellSouth's plant to reach me over the last mile. CLECs can do other things as well; they have various rights-of-way that let them build private networks across conduits in public spaces, they can lease dark fiber laid by electrical and gas utilities, etc. But the defining feature of a CLEC is that they don't own the existing physical plant in place before, although they are welcome to overlay their own - in fact, that was very much the point of the Telecommunications Act. Most CLECs are small, but some are quite large and have a regional, national and even international footprint. Examples of the large ones include Level3, Global Crossing, XO, McLeod USA, Paetec, Nuvox, etc. -- these network operators all have CLEC status in many different incumbents' operating areas, if not necessarily all of them. Some CLECs neither do UNE nor really build networks nor lease anything, but exist for some specialised purpose to reap some economic or logistical advantage, like supporting the back side of a VoIP product or providing dedicated private transport between various large interconnection / peering points. There are many different niches for the sort of thing that they are. Nor does a CLEC have to have an imposing physical presence; it is quite possible, with the right equipment, to
Re: [asterisk-users] cdr mysql error
Increase the timeout in my.cnf in mysql. -Jai Buy unmetered VoIP DIDs www.didforsale.com Free Trail On Mon, Nov 24, 2008 at 11:10 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Need help on mysql cdr, i keep on seeing this log on the console. but my db is up and i see the calls being logged on the cdr table. is there a timeout when there is no activity? can i remove the timeout if there is any? thanks [Nov 25 13:22:37] ERROR[21026]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. [Nov 25 14:20:32] ERROR[21061]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER/Asterisk interworking mailing list.
Good work, I am sure this will be endorsed by many and will be useful for lots of small VoIP user who are ready to expand. Only problem I have seen is that people who have done (deployed) this type of integration does not share complete solution mainly because of compititive disadvantage. But keeping the information at one place will definitely help. I am also working on a 'howto' on integrating Asterisk with Ser that will describe step by step instructions on the deployment of asterisk. I have tons of many things in my plate but targeting to finish within next week or so. -Jai Buy unmetered VoIP DID from DidForSale.com On Wed, Nov 5, 2008 at 9:04 AM, Alex Balashov [EMAIL PROTECTED]wrote: Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in duplicated effort and lack of specialised response. This is mainly due, I think, to the fact that detailed Asterisk experience - while common - is not a prerequisite for working with the SER products, while for Asterisk people SER can often be a next step in scalability and VoIP service delivery platform enhancement that they are just getting into. And so on. There's pollution in the respective discursive spaces; a lot of Asterisk people posting to the SER lists ask a lot of Asterisk-specific questions in addition to any they may have about SER which can be construed as potentially off-topic by some members, and the opposite is true on the Asterisk lists when detailed, involved discussion about SER occurs. We need to capture that discussion that exists at the overlap and is specifically concerned with making these two systems work together, requiring somewhat detailed and esoteric understanding of both and a community of user support and knowledge that focuses on both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: [EMAIL PROTECTED] It's the same GNU Mailman stuff you are already used to. While it could be argued that this cross-product discussion is valuable to retain in both communities, I think there is considerable benefit to creating a specialised mailing list that focuses specifically on this integration path and the unique interoperation and configuration issues it creates. I think it would be good to get some of this discussion off of the SER and Asterisk-specific mailing lists where it has somewhat marginal relevance at times and refocus it. If you agree and are interested in this topic, you are invited to join the list. One last note: The SER/OpenSER community has been in a state of flux recently, with OpenSER undergoing a name change to Kamailio and subsequently seeing a fork. The incumbent Kamailio project is now in the process of merging with the original SER project. The choice of nomenclature for list is not meant to imply an endorsement of or affinity for the IPTel SER project per se. It is just that right now it serves the aim of terseness to use a common denominator, to refer to this family of projects as the SER ecosystem. Whether you are a SER, OpenSER, Kamailio, or OpenSIPS user, you are part of that SER ecosystem. That is why the list is named what it is. Thank you, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
SIP-only accounting is good enough most of the time. Does not work in production environment. Specially when you are charging per second or per minute. Works only if some one is offering unmetered only service or just doing it for fun. If it metered service like calling cards, termination or metered DID etc, then this can be really bad. My 2 cents. -Jai Buy unmetered SIP DID www.didforsale.com On Wed, Oct 29, 2008 at 3:56 PM, Alex Balashov [EMAIL PROTECTED]wrote: Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting. But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is good enough most of the time. Nuno Marques wrote: Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Really? Yes, Specially when your service is metered, I don't know how some once justify good enough billing. Dealing with 500 customer calling every day for billing inquiries can turn out to be much more expensive then all other expenses. Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. No Need to be so contemptuous. On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED]wrote: Jai Rangi wrote: SIP-only accounting is good enough most of the time. Does not work in production environment. Really? Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. By the way, there are, of course mitigating strategies to minimise risk. Dialog-stateful modules can end the dialog after a certain timeout, you can send periodic re-invites with an SDP offer to probe the endpoints, etc. It is far wiser than introducing a point of failure, a source of latency, and a source of huge bandwidth and processing cost into the call path when you don't need it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound DID + voice ports needed for vote monitoring project
Hello Dave, We can offer you. What area DID you are looking for. Jai Buy SIP DID, www.didforsale.com On Thu, Oct 23, 2008 at 2:20 PM, David Troy [EMAIL PROTECTED] wrote: Hey folks, I am involved with a group that is going to use Twitter, SMS, iPhone, and Asterisk to field-monitor the upcoming US elections. The group is pretty large scale and you can find out more here: http://votereport.pbwiki.com We need some help with SIP telephony infrastructure. Specifically, we need approximately 200 inbound SIP ports, driven by just one US DID. We have a beefy asterisk box located in NYC and can take delivery of this traffic via the public internet comfortably. Is there a carrier on the list who can provide this kind of capacity between now and November 4 pro-bono, for the good of the US democratic process? Please contact me off-list if this sounds like something you can do. You would receive press and publicity as a partner in return. Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip and nat
John, Client Behind a NAT should not be problem. What are your issues? If you post your scenario and more details about your problem only then some can help you better. Jai Buy SIP DID at www.didforsale.com On Wed, Oct 22, 2008 at 12:24 AM, Johanna NIRINA [EMAIL PROTECTED]wrote: hi there, I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm wonder what is the best way to resolving the Asterisk/NAT problem : some clients are behind a NAT. anyone could help me? thanks johanna _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Problem
Check the permissions for the directory. Jai http://www.didforsale.com . On Sun, Oct 19, 2008 at 1:19 PM, Ahmed Torintino [EMAIL PROTECTED]wrote: i have done that as follow [EMAIL PROTECTED] asterisk]# service asterisk start Starting asterisk: [ OK ] [EMAIL PROTECTED] asterisk]# asterisk [EMAIL PROTECTED] asterisk]# asterisk [EMAIL PROTECTED] asterisk]# asterisk -vr Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) but i think the problem is because of those couple files /var/run/asterisk.ctland/var/run/asterisk.pid Thanks -- Date: Sun, 19 Oct 2008 16:05:34 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Problem When installing Asterisk, did you issue the command make config after make samples ?? If so, try issuing service asterisk start on RedHat or /etc/init.d/asterisk start on Debian. Regards, Juan On Sun, Oct 19, 2008 at 3:50 PM, Ahmed torinto [EMAIL PROTECTED]wrote: After installing a new box and asterisk. i have got these errors [EMAIL PROTECTED] ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory [EMAIL PROTECTED] ~]# asterisk -vr Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) I didn't find a folder called asterisk in the directory /var/run [EMAIL PROTECTED] ~]# cd /var/run/ [EMAIL PROTECTED] run]# ls acpid.socket dbus iptraf messagebus.pid ntpd.pid sendmail.pid sudo utmp atd.pid dhclient-eth0.pid klogd.pid mysqld ppp sm-client.pid syslogd.pid winbindd console haldaemon.pid mdadm netreport rpc.statd.pid spamassassin tog-pegasus xfs.pid crond.pid httpd.pid mdmpd nscdsaslauthd sshd.pid usb xinetd.pid [EMAIL PROTECTED] run]# [EMAIL PROTECTED] run]# asterisk -cv Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory Unable to bind socket to /var/run/asterisk/asterisk.ctl: No such file or directory == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.28, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Parsing '/etc/asterisk/dnsmgr.conf': Found Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found [EMAIL PROTECTED] run]# How can i solve it please? -- Get news, entertainment and everything you care about at Live.com. Check it out! http://www.live.com/getstarted.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 -- Explore the seven wonders of the world Learn more!http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Improving the voice Quality,
All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? Any thoughts? -Thank you, -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delaytos=0x08 high throughput tos=0x04 high reliabilitytos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
All, Just an update on this. This turned out to be a bug in Cisco firewall. We ended up in upgrading the Firmware on the firewall. One thing I want to add, this was first time we used the fail over unit during peak time. In the whole process (failover, upgrade and failover back to active unit) was completely seamless. Did not had any down time, there was just a pause for just 1 second in the audio. I was very impressed. -Jai On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote: Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Failover System
For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED]wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Failover System
Yes Redfone will do the T1 failover. Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) Jai *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 4:00 PM, Steve Totaro [EMAIL PROTECTED] wrote: You can use two OpenSer boxen with heartbeat and the dispatch module for load balancing if you need it, and failover, in front of a couple of Asterisk boxen connected to a Redfone device (TDMoE). Thanks, Steve Totaro On Wed, Oct 1, 2008 at 5:40 PM, Jai Rangi [EMAIL PROTECTED] wrote: For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED] wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test call generator
Are you looking for inbound or outbound. I can get you free inbound test DID. LMK Jai www.didforesale.com On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -- Sent from Gmail for mobile | mobile.google.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
We are using few dell 1950, it been two year and never had any issue, Jai www.didforsale.com *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Thu, Sep 25, 2008 at 3:19 PM, Alex Balashov [EMAIL PROTECTED]wrote: Philipp Kempgen wrote: Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. At least Dell doesn't seem to play nice with Debian. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
Hitesh, If you dont have experience with Linux I would recommend you to use Trixbox, that will come with all the required packages and will do everythign for you. Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can buy DIDs that can come to your asterisk over the internet. Jai www.didforsale.com *Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com; On Fri, Sep 19, 2008 at 9:18 AM, logan [EMAIL PROTECTED] wrote: Hello Ram, Thanks for the response. As I said there are too many options out there :). Could you help me in settling down on one? Something that will work with the phone lines in India is just fine for me. I don't have any or much Linux experience, but willing to play around, so any compatible distro will do for me. So once again: Which Linux distro is best with Asterisk? Which hardphone is the easiest to setup? Which fxo/fxs card I should go for? Thanks a lot guys. Best Regards, Hitesh On Thu, Sep 18, 2008 at 10:33 PM, ram [EMAIL PROTECTED] wrote: On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
Hitesh, Not sure if I understand your question. Let me try to explain again, There are two thing in Asterisk, Origination and Termination. Origination: You have a DID (Virtual Phone line say from LA), people call that Virtual line from anywhere in the world, and you will receive that call to your asterisk server. based on yoru dialplan rules Asterisk sends that call to your VoIP Phone registered on your asterisk. (Or regular phone connected through VoIP ATA Ex GrandStream ATA). Termination: Your VoIP phone registered on Asterisk wants call some number anywhere in the world. Now your asterisk needs a termination provider, who will receive call from your asterisk and will terminate that call to the destination number. Or you have two phones registered on your asterisk one in India, one is Aanada, they can call each other without any origination or termination provider. The point is that origination and termination can be done through FXO OR FXS card OR you can tie up with some one like www.didforsale.com who can do this over the internet. 2nd one is always more cheaper, more options, much easy to configure and troubleshoot. Hope this will help you, Jai www.didforsale.com http://www.didforsale.com/ Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com; On Fri, Sep 19, 2008 at 12:00 PM, logan [EMAIL PROTECTED] wrote: Hi Jai, If I understand correctly then the DID will enable to call me on the hardphone connected to the Asterisk. Will it also enable me to call out using the PSTN line at my home in India from Canada? Thanks. Best REgards, Hitesh On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi [EMAIL PROTECTED] wrote: Hitesh, If you dont have experience with Linux I would recommend you to use Trixbox, that will come with all the required packages and will do everythign for you. Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can buy DIDs that can come to your asterisk over the internet. Jai www.didforsale.com *Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com; On Fri, Sep 19, 2008 at 9:18 AM, logan [EMAIL PROTECTED] wrote: Hello Ram, Thanks for the response. As I said there are too many options out there :). Could you help me in settling down on one? Something that will work with the phone lines in India is just fine for me. I don't have any or much Linux experience, but willing to play around, so any compatible distro will do for me. So once again: Which Linux distro is best with Asterisk? Which hardphone is the easiest to setup? Which fxo/fxs card I should go for? Thanks a lot guys. Best Regards, Hitesh On Thu, Sep 18, 2008 at 10:33 PM, ram [EMAIL PROTECTED] wrote: On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] Pre-paid Billing
Another idea can be have the customers to opt-in for auto-refill if they want to use multiple call feature. Usually this does not have be a high number, just autorefill the account if the balance goes down $1. Jai www.didforsale.com *Buy DID at low cost http://www.didforsale.com; On Thu, Sep 18, 2008 at 8:14 AM, Jim Boykin [EMAIL PROTECTED] wrote: Thanks guys for inputs...not allowing multiple call is not an option - essentional thats the problem we try to solve :) Since we have our own CDR module, we can avoid external process. What are the evens to listen for? Other ideas will also be appreciated. On Thu, Sep 18, 2008 at 8:23 PM, Alex Balashov [EMAIL PROTECTED] wrote: Igor Zamocky wrote: Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) You don't have to update the balance every second - increments of something like 10 seconds will do. And you can have one synchronous process - not many - that listens to Manager events and updates the call times and balances accordingly. An outside process can also trigger a Hangup event causing the call to be hung up if credit is exhausted, or too low. Then, you can define a minimum formula for the balance required to admit a new call. Something like a minute of credit being required after subtracting the usage of all existing simultaneous calls in progress at the next projected utilisation polling interval. But you are essentially correct. Things are, of course, far easier if you just don't allow multiple simultaneous calls. :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy Load Asterisk Array
We also have the similar setup, 2 ser server with heartbeat doing the load balance and 4 asterisk servers handling the media. Of course the data is on MySQL Cluster. Jai Rangi www.bingotelecom.com On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote: I have used the OpenSer dispatcher module to load the calls (hash by caller id) to a group of asterisk boxes (In my case, 2 servers). The Asterisk boxes both use ARA and MySQL Master/Master replication. In a case like yours, I think you can use MySQL cluster, and you can still use Dispatcher to balance the load. On Mon, Jul 21, 2008 at 5:22 PM, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and LVS
Has anyone used or thought of using Asterisk server farm behind LVS. -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider recommendation in USA
Vivek, What do you need, DID or Termination? BTW We are in California. Send me you Contact info and we can discuss more about your needs. -Jai On Thu, Mar 6, 2008 at 10:25 AM, Vivek Shrivastava [EMAIL PROTECTED] wrote: Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Most Stable version of Asterisk
Thank you Everyone, I tried 1.4.15 on one and I am monitoring it. I am using asterisk with a2billing On thing I already have noticed is that, a2billing did not bill few calls. Not sure why. -Jai On Dec 12, 2007 9:10 AM, shadowym [EMAIL PROTECTED] wrote: 1.4.15 on CentOS 5.1 is running smooth as silk for me. *From:* Jai Rangi [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 11, 2007 2:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Most Stable version of Asterisk Hello, I tried to install the asterisk 1.4.15 and I am not able to start it. I get the segmentation fault error. What might be wrong, where I can look for a clue. Also could some one PLEASE suggest the most stable version of asterisk. -Jai ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Didnt get a frame from Channel and call gets disconnected
Anyone, could you please suggest the latest stable release for asterisk. -Jai On Dec 10, 2007 9:08 PM, Jai Rangi [EMAIL PROTECTED] wrote: I am planning to upgrade my asterisk to Asterisk 1.4.15http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.15.tar.gz Zaptel 1.4.7http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.7.tar.gz Libpri 1.4.2http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/libpri/releases/libpri-1.4.2.tar.gz Addons 1.4.5http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.5.tar.gz Currently I have Asterisk 1.2.12 zaptel-1.2.9.1-98.fc5.at libpri-1.2.3-1.369 Addons 1.2.4http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.5.tar.gz Few Questions before I do that. Are the current asterisk/zaptel/libpri version are stable and safe to use in the production environment. Will I be able use my current configuration files? Is there any major change in the way directories are structured. I have configured FreePBX for web management of few components of Asterisk. Will the be any problem, if I don't plan to upgrade free PBX (Because we have customized few components and don't want to do that again). My Current asterisk is configured with MySql. Is there any change in the asterisk tables and databases structures. I will appreciate any feedback. Thank you, -Jai On Dec 10, 2007 4:27 PM, Jai Rangi [EMAIL PROTECTED] wrote: Thank you Jared, I had the same feeling. But my servers are in production, doing great except this problem. So i was hoping if someone had that same issue and if there is/was an easy fix for this. -Jai On Dec 10, 2007 4:09 PM, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-12-10 at 15:26 -0800, Jai Rangi wrote: Is this the right place to post this error message and expect for the solution. I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me some hints to get rid of this problem. I doubt you'll get much response, unless you try again with a newer version of Asterisk. Asterisk 1.2.12 is quite old (and the 1.2 series is no longer receiving bug fixes), so I wouldn't expect that the core developers would spend much time trying to track it down. If you can reproduce the problem in a recent 1.4 version, I'm sure they'd be happy to look at it. --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Most Stable version of Asterisk
Hello, I tried to install the asterisk 1.4.15 and I am not able to start it. I get the segmentation fault error. What might be wrong, where I can look for a clue. Also could some one PLEASE suggest the most stable version of asterisk. -Jai ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Most Stable version of Asterisk
And that version name/number is ??? :) -Jai On Dec 11, 2007 4:17 PM, C F [EMAIL PROTECTED] wrote: In my experience the most stable asterisk is the one that runs and runs and never crashes. On 12/11/07, Jai Rangi [EMAIL PROTECTED] wrote: Hello, I tried to install the asterisk 1.4.15 and I am not able to start it. I get the segmentation fault error. What might be wrong, where I can look for a clue. Also could some one PLEASE suggest the most stable version of asterisk. -Jai ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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: SIP/5060-b7a03560 Dec 10 11:26:41 DEBUG[10410] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:53 DEBUG[10415] channel.c: Didn't get a frame from channel: SIP/5060-b7a0e2a8 Dec 10 11:26:53 DEBUG[10415] channel.c: Bridge stops bridging channels SIP/5060-b7a0e2a8 and SIP/Vendor-089d35b8 Dec 10 12:06:45 DEBUG[17210] channel.c: Didn't get a frame from channel: SIP/5060-b7a03560 Dec 10 12:06:45 DEBUG[17210] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/2219.206.2.291-089d8768 Dec 10 12:40:15 DEBUG[23089] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:40:15 DEBUG[23089] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 12:57:48 DEBUG[25800] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:57:48 DEBUG[25800] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 12:58:05 DEBUG[25809] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:58:05 DEBUG[25809] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:10:36 DEBUG[5927] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 14:10:36 DEBUG[5927] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:11:28 DEBUG[5961] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:11:34 DEBUG[5961] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 14:11:34 DEBUG[5961] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Is this the right place to post this error message and expect for the solution. I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me some hints to get rid of this problem. Thank you, -JP ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Didnt get a frame from Channel and call gets disconnected
Thank you Jared, I had the same feeling. But my servers are in production, doing great except this problem. So i was hoping if someone had that same issue and if there is/was an easy fix for this. -Jai On Dec 10, 2007 4:09 PM, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-12-10 at 15:26 -0800, Jai Rangi wrote: Is this the right place to post this error message and expect for the solution. I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me some hints to get rid of this problem. I doubt you'll get much response, unless you try again with a newer version of Asterisk. Asterisk 1.2.12 is quite old (and the 1.2 series is no longer receiving bug fixes), so I wouldn't expect that the core developers would spend much time trying to track it down. If you can reproduce the problem in a recent 1.4 version, I'm sure they'd be happy to look at it. --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Didnt get a frame from Channel and call gets disconnected
I am planning to upgrade my asterisk to Asterisk 1.4.15http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.15.tar.gz Zaptel 1.4.7http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.7.tar.gz Libpri 1.4.2http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/libpri/releases/libpri-1.4.2.tar.gz Addons 1.4.5http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.5.tar.gz Currently I have Asterisk 1.2.12 zaptel-1.2.9.1-98.fc5.at libpri-1.2.3-1.369 Addons 1.2.4http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.5.tar.gz Few Questions before I do that. Are the current asterisk/zaptel/libpri version are stable and safe to use in the production environment. Will I be able use my current configuration files? Is there any major change in the way directories are structured. I have configured FreePBX for web management of few components of Asterisk. Will the be any problem, if I don't plan to upgrade free PBX (Because we have customized few components and don't want to do that again). My Current asterisk is configured with MySql. Is there any change in the asterisk tables and databases structures. I will appreciate any feedback. Thank you, -Jai On Dec 10, 2007 4:27 PM, Jai Rangi [EMAIL PROTECTED] wrote: Thank you Jared, I had the same feeling. But my servers are in production, doing great except this problem. So i was hoping if someone had that same issue and if there is/was an easy fix for this. -Jai On Dec 10, 2007 4:09 PM, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-12-10 at 15:26 -0800, Jai Rangi wrote: Is this the right place to post this error message and expect for the solution. I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me some hints to get rid of this problem. I doubt you'll get much response, unless you try again with a newer version of Asterisk. Asterisk 1.2.12 is quite old (and the 1.2 series is no longer receiving bug fixes), so I wouldn't expect that the core developers would spend much time trying to track it down. If you can reproduce the problem in a recent 1.4 version, I'm sure they'd be happy to look at it. --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
