[asterisk-users] AMQP Support for Asterisk?
Are there any plans to implement AMQP directly in Asterisk or is it best to use a third party bridge like Mule? https://jira.amqp.org/confluence/display/AMQP/Advanced+Message+Queuing+Protocol -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] Crontab script to check health of Asterisk server?
Has anybody created a crontab script to check the health of an Asterisk server? The part I'm struggling with is some sort of IAX ping to test the connection to each provider without making a call. -HJC ___ --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] Question about dnsmgr
Asterisk 1.4.5 full log: [Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to move existing IAX connections when you see a DNS change or do you leave the existing connections the fnord alone on their current IP addresses and simply use the DNS change for new connections? -HJC ___ --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] IAX best practices
You will likely have latency issues - causing choppiness. Start with a traceroute to validate latency. Anybody tried IAX trunking on G.729 with jitter buffer internationaly? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
If the system is running away then I'd suggest looking deeper into it - is it opening a file and never closing it again, etc. Hard to track down unless you have a good knowlege of what's running, etc. If you think it might be asterisk itself, then check which files it has open. lsof -p `ps h -C asterisk -o pid | head -1` -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
If the system is running away then I'd suggest looking deeper into it - is it opening a file and never closing it again, etc. Hard to track down unless you have a good knowlege of what's running, etc. lsof -p `ps h -C asterisk -o pid | head -1` | grep -Fc '/dev/zap/timer' 120 You have to open your own timer device over one hundred times in the same process? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN As SIP Tunneling?
Luki [EMAIL PROTECTED] wrote: You don't gain anything QOS-wise by going through a tunnel, except hiding your traffic in case your ISP purposefully assigns lower priority to VoIP traffic and doesn't do it to OpenVPN/GRE/insert your favorite tunnel protocol traffic. It's a pity that OpenVPN doesn't have an option to hide as https requests (and handle the double-TCP problem internally) or even better yet gif uploads over http. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forgetting about client registration or Polycom phone forgetting to register?
I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. ... host=dynamic We've found much better results with the static IP here. Can you try this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] illegal VoIP in India
I ran across this article today: http://economictimes.indiatimes.com/articleshow/726843.cms Anybody know what the implications are for asterisk servers in and out of the country used by people in the country? Ummm Anybody offering VPN IAX services yet? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)
Going to the other extreme, what would it take to create an untappable and untraceable telephone service over the Internet? Asterisk is a good start, especially because the code can be examined (as long as G729 is avoided) and any law enforcement back doors removed. Now instead of trying to harden the wire protocols Asterisk uses, simply have it connect via VPN tunnels setup by other software. (Remove all the DNS calls from Asterisk also.) You could setup a tiny Linux box to automatically war-drive for unsecured hotspots, but then you'd need to bounce through trusted relay servers or overcome NAT in some way. Plus there is the problem of advertising your current IP address, but only to the people you'd like to call you. (Encrypted files on file sharing networks?) BTW: Nobody (within reach of the United States Military) should speak about such things after the detainee bill gets signed into law. ;-) -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 failover when out of licenses
[EMAIL PROTECTED] wrote: Can't help you on the licensing thing though. I guess no one wants to touch it since Digium's stance seems to be that you should have a license for each seat rather than a pool. That's not enough. You need one license per call, with no upper limit on the number of inbound calls your providers might deliver at any one moment. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!
Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country Then i just go to a click to call website and start an attack inserting this Special Phone Number like 1$/min. Even if i control call duration on asterisk, and also my recepcionist will notice this is a fraud call, i'll be charged for the 1st minute every time Why not exclude international and 809 outbound calls entirely and then bless specific countries as needed? You could include your phone number on your site so that people from other countries could call your center as needed. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Standard for transfer via IAX provider?
Is there any standard way to signal to an IAX provider that I want them to conference in another Asterisk box located elsewhere and then hand off the call to the remote center after a short period of three-way talk? My problem is that I don't want to take a double hit for latency back and forth from the United States. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS
Ola Lidholm [EMAIL PROTECTED] wrote: To sort of resolve this I had to install a local name-server on the machine that contains the addresses asterisk tries to resolve (changing to using IP-addresses did not fix the issue for me either). I would prefer an option in asterisk that tells it to not resolv more than once on each address. Have you tried setting timeout, attempts and rotate in resolv.conf? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls over VPN
We did a setup of 70 sites connected back to a central Asterisk box, and it worked very well over an MPLS VPN. regards, PaulH AsteriskIT www.asteriskit.com.au The best part about VPN is that it makes it harder for the ISPs to track and mess with. ;-) -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
[EMAIL PROTECTED] wrote: On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident. are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
Matt Florell [EMAIL PROTECTED] wrote: My backtraces never actually mention play_sound, but the crashes only happen right after app_conference attempts to play out DTMF tines with the playing function. This is because Malloc isn't crashing when the mistake is made. It crashes later because of the out of bounds write or double free has corrupted its memory structures. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
Brian Capouch [EMAIL PROTECTED] wrote: Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggested (limiting codecs, primarily). Something is weird there in that for some it appears to work perfectly, for others not at all. . . This is starting to stink a lot more like a memory overrun error than a double free error. This looks like a job for Electric Fence. http://www.die.net/doc/linux/man/man3/efence.3.html -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Optimization and Load Balancing
Mitch Jackson [EMAIL PROTECTED] wrote: Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. ... All calls recorded to disk ... 72 Ploycom 301P SIP phones using ulaw codec If you run this command vmstat 5 Does it show lots of processes that are blocked, waiting for the disk? If I had to do this I would have a battery backed writeback RAID controller. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
jeff oconnell [EMAIL PROTECTED] wrote: but while i'm in the code, i'll also take a look and see if i can figure out what your memory issues are... When I've tried it, app_conference always crashed within the hour. I think that the entire Asterisk server, including app_conference, needs to be compiled with one of the debugging malloc libraries because it might be that you are returning something to Asterisk in such a way that it does either a double free or a free of non-mallocated memory. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Encrypting the Conversation
Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use a data T-1?
I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren If this data T-1 just goes to the Internet then you would use it just like any other network connection at your cisco router. If this data T-1 goes between two sites of yours then you could use it either as a dedicated route between network cards on each end (that connect to cisco or other brand routers) or a voice route between two Asterisk servers with voice T-1 cards. The choice would be between capacity for say G729 trunks over a data link or latency as voice T-1s. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 100 lines + system config
Hello, We are planning to biuld a 100 lines PBX based on asterisk. How do you decide on the system config, e.i motherboard, cpu , how much ram , etc ? Is there any thumb rule ? What is your duty cycle? By that I mean, how often is each line going to be used? Will you have 100 phones plugged in but only a dozen calls going on at one time or will you be running a predictive dialer with 80 agents talking to contacts while another 150 calls are being placed predictively? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plainvoip problem.
Mike Lynchfield [EMAIL PROTECTED] wrote: could it be IPP VS digium implementation ? Actually it seems to have been a NAT issue and adding canreinvite=no (as suggested by another person offlist) fixed it up. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Plainvoip problem.
When calling through Plainvoip from my Asterisk at Home box I get the following log entries. Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert # Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by 66.199.240.2 (format g729) Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 Jun 8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped sounds Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered SIP/503-6d4c Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 What I hear on the phone is one ring and then nothing. This has only been in the past few days. Has anybody else had a problem like this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plainvoip problem.
Do you have the g729 codec? On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 ... Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 Yes, and that works fine when talking with the phone itself, as you see the connection to the phone is g729. Then it changes from g729 to g729? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I guess my server capacity is ok
All I know is that it is very processor intensive and either not using it or just passing it through is your best bet. I will be working alot with G729 in the near future and will post my findings but until then I am just relying on the dimensioning page on the wiki. Thanks, Steve Totaro Which DSP based boards does Asterisk support for G729 and are any of these more cost effective than piling on Pentiums? BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference sources?
The CVS server for app_conference seems to be down. Can somebody mail me a recent copy of the sources please? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference DTMFs?
We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hook into authentication
to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? This would be only mildly useful on the same subnet and completely useless over the internet. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
