Re: [asterisk-users] Multiple IAX2 Trunks Load balancing
On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Sounds wonderful. S -- _ -- 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] Multiple IAX2 Trunks Load balancing
yeah -- searching how to perform this magic ... On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote: On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Sounds wonderful. S -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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] Multiple IAX2 Trunks Load balancing
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. Hi, Generally, incoming traffic (http/mail/sip/vpn/etc) can easily spread with the random functions with iptables. Perhaps also have a look at: http://www.linuxvirtualserver.org/VS-NAT.html But i feel that might not work in this case, if all your trunk-connections are already up-and-running. If so, you might contemplate an alternative approach. It sounds that you want to do the load balancing trick upon the application layer (eg. within asterisk) Perhaps it is possible to do the L.B. at the O.S. or network level, and let all trunks appear to asterisk to one single trunk. -- _ -- 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] Multiple IAX2 Trunks Load balancing
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. Hans said: Hi, Generally, incoming traffic (http/mail/sip/vpn/etc) can easily spread with the random functions with iptables. Perhaps also have a look at: http://www.linuxvirtualserver.org/VS-NAT.html But i feel that might not work in this case, if all your trunk-connections are already up-and-running. If so, you might contemplate an alternative approach. It sounds that you want to do the load balancing trick upon the application layer (eg. within asterisk) Perhaps it is possible to do the L.B. at the O.S. or network level, and let all trunks appear to asterisk to one single trunk. Don asks: What's the value of load balancing multiple IAX trunks between the same system pair? What resources are being balanced? --Don -- _ -- 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] Multiple IAX2 Trunks Load balancing
On Fri, 13 Dec 2013, Don Kelly wrote: What's the value of load balancing multiple IAX trunks between the same system pair? What resources are being balanced? 1) It's been said (but I've never experienced it) that IAX performs poorly at high channel levels. Something about all those packets being handled with a single socket (bound to port 4569). Maybe the OP is trying to distribute the load over a handful of sockets. 2) Maybe the OP is trying to distribute the load over different physical media. 3) Maybe the OP is just trying to develop a load balancing methodology. Multiple trunks between the same pair of hosts does seem odd. -- 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 -- 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] Multiple IAX2 Trunks Load balancing
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote: On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. Hans said: Perhaps it is possible to do the L.B. at the O.S. or network level, and let all trunks appear to asterisk to one single trunk. Don asks: What's the value of load balancing multiple IAX trunks between the same system pair? What resources are being balanced? ++ Perhaps the O.P. can explain about his intentions... In some situations it makes sense though: If you have to connect two servers, and use different kind of infrastructure / multiple providers... hw -- _ -- 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] IAX2 bridge failing
Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] IAX2 bridge failing
I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] IAX2 bridge failing
Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Thursday, December 12, 2013 5:08 PM To: Asterisk Users List Subject: [asterisk-users] IAX2 bridge failing I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong.One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's a clue? The dialplan seems to execute properly, and I can watch the destination system which answers the call and starts playing media (monkeys) which I don't hear. Any ideas on what is going on? Since this is IAX in and IAX out, NAT should not be an issue (even through there is NAT on both sides). Since media moves on the same UDP port as call setup, also proves should not be a network problem (I think) Can someone point me to a solution? Thanks! (IP's and ISP and phone number disguised) - Executing [s@macro-dialexternal:57] GotoIf(IAX2/S-14468, 1?dialnormal) in new stack -- Goto (macro-dialexternal,s,60) -- Executing [s@macro-dialexternal:60] Dial(IAX2/S-14468, IAX2/ISP123/1234567890|60|W) in new stack -- Called ISP123/1234567890 -- Call accepted by 201.191.37.138 (format ulaw) -- Format for call is ulaw -- IAX2/ISP123-2261 answered IAX2/S-14468 -- Channel 'IAX2/S-14468' ready to transfer -- Channel 'IAX2/ISP123-2261' ready to transfer -- Hungup 'IAX2/ISP123-2261' -- _ -- 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] [NEWBIE] Right dect to buy to use with asterisk
Hi, I made good experienes with Siemens Gigaset C610 IP. This model is about 90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is *not* possible with this phones. -Thorsten- Am 11.12.2013 11:30, schrieb Mario Giammarco: Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a standard set of features: - each dect phone has its extension so I can call it directly; - handover of a call with R key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which one should I buy? -- _ -- 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] Cisco IP Phones and Skinny in asterisk
If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. Where do u download the SIP firmware usually for your Cisco IP Phones? Search for cmterm-7941_7961-sip.8-3-1.zip I also have some other files here but I don't remember what was the reason for them :-( Martin Your kindly help is highly appreciated. Regards Bilal I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. --- Tato zpráva neobsahuje viry ani jiný škodlivý kód - avast! Antivirus je aktivní. http://www.avast.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] Multiple IAX2 Trunks Load balancing
Friends let me define the scenario please; Scenario: 2 asterisk servers (A B) are connected using 05 IAX2 trunks between them. The machine A is running asterisk Openvpn server in TUN mode (5 instances with difference IP addresses for clients). The machine B is running asterisk with 05 OpenVPN clients using 05 bandwidths. The IAX trunks are established between each pair of P-2-P ip address of machine A (The OPENVPN Server) machine B (The Openvpn client). Requirement: Required dial plan configuration at machine A for incoming calls from VoIP Switch/VOS which can forward the calls to IAX2 trunks in round robin fashion like Load Balancing. If any trunk goes down it starts forwarding the traffic to other available trunks when it gets UP the dialplan should perform as desired. Like L.B Fail-over scenarios. On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl wrote: On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote: On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. Hans said: Perhaps it is possible to do the L.B. at the O.S. or network level, and let all trunks appear to asterisk to one single trunk. Don asks: What's the value of load balancing multiple IAX trunks between the same system pair? What resources are being balanced? ++ Perhaps the O.P. can explain about his intentions... In some situations it makes sense though: If you have to connect two servers, and use different kind of infrastructure / multiple providers... hw -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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
