[asterisk-users] No more connections allowed
Hi I'm having the same issue as someone else who wrote into this list but feel I have more information to add and that this is possibly a new bug. We have multiple servers, most running 1.8.7.0 and one running 1.8.18.0 which is our next upgrade candidate (all servers running CentOS 5). We have a cron running on all the machines which runs every minute and gets data from the asterisk and pushes it to a monitoring system. It gets the information by running asterisk -rx 'command' This morning, on the machine running 1.8.18.0 (which is exactly the same version as the last person to report this problem) I can see in the logs that there isn't a Remote UNIX connection disconnected for every Remote UNIX connection Over time these stale/phantom connections build up and we hit the remote connections limit which renders asterisk completely unresponsive. This same cron has been running on the 1.8.7.0 machines for about half a year without this problem ever occurring. So, the 2 points I want to make are: Has something happened between 1.8.7.0 to 1.8.18.0 that means executing an asterisk -rx means the execution isn't completed cleanly from the point of view of the asterisk service? Should hitting the limit of the concurrent remote connection really render the service completely unresponsive (to the point that restarting using /etc/init.d/asterisk restart doesn't work)? Thanks Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] TLS
Le 06/02/2013 23:15, kepin sinatra a écrit : Hi, I tried it the implementation of TLS in asterisk 1.8.4.3 on ubuntu 10.04. I follow the tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. and I use blink as a softphone in ny client in windows. for regular communication process (without TLS) smoothly, but when it just follow the tutorial, it is always error on his softphone: transport error. Check that blink is configured for TLS. Also, when you start asterisk or sip reload check that message SSL certificate ok appears in your logs Other check: run tshark on the interface of your asterisk on port 5061 in tcp to check if the traffic of your softphone arrive to the good port. -- Daniel -- _ -- 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] Asterisk calls between 2 private networks
My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no The Digium on network B can register. I can see it when I do sip show peer xxx. When the phones are calling each other, the signaling is working. They ring. But when they pick up, there is no audio, in any way. Has anyone ever worked on the same configuration, and had success ? If yes, I'd love to hear your story and check your configuration. Thanks ! -- _ -- 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] Asterisk calls between 2 private networks
For the phone on the public network. you might need to set canreinvite=no. My guess is that if you listen really closely you would have about a quarter second of audio before it cuts out. Whenever I have had this happen it is because the packets didn't know how to reroute from the IP address of the Asterisk server to the IP address of the phone. My guess is that your network has the proper pathing to send the packets into the servers IP address but can't redirect them to the other IP addresses. If it works, you can leave canreinvite on for phones in the private network, but any that will register to the public network should have it set to no. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 08:39 AM Subject:[asterisk-users] Asterisk calls between 2 private networks Sent by:asterisk-users-boun...@lists.digium.com My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no The Digium on network B can register. I can see it when I do sip show peer xxx. When the phones are calling each other, the signaling is working. They ring. But when they pick up, there is no audio, in any way. Has anyone ever worked on the same configuration, and had success ? If yes, I'd love to hear your story and check your configuration. Thanks ! -- _ -- 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] Asterisk calls between 2 private networks
On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- AJS Answers come *after* questions. -- _ -- 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] Call abandoned from queue not showing in CDR (possible bug)
On Tue, 2013-01-29 at 08:32 -0600, Matthew Jordan wrote: On 01/29/2013 02:52 AM, Ishfaq Malik wrote: On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote: On 01/16/2013 05:31 AM, Ishfaq Malik wrote: On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote: Hi Everyone This issue has reared it's ugly head again for us. If a call comes into a queue and the caller abandons the call, the call does not show in the CDR. This is also the case for asterisk version 1.8.18 Does anyone have any ideas, or try to replicate it? Thanks in advance Ish Do you have unanswered=yes set in cdr.conf? CDRs in Queues can depend heavily on your dialplan, whether or not the call is Answered prior to it going into the Queue, etc. What is the state of the inbound channel when it goes into the Queue? unanswered=yes in the cdr.conf would have too many side effects for us (i.e. a single cdr entry for each channel rung). To me this behaviour seems inconsistent with that of Dial. If I use dial to call 3 peers and the caller abandons the call I will get a single CDR entry with disposition NO ANSWER. Now if I use Queue to call the same 3 peers that are members of that queue and abandon the call, I get no cdr entry at all. This to me seems wrong. Regards Ish Hi Ish - The behaviour of CDRs in Queue can be interesting at times, and doesn't always match the behaviour of what occurs through Dial. In this particular case, because Queue doesn't Answer a call automatically for you, a lack of an Answer prior to going into Queue means the 'unanswered' logic kicks in for the CDRs. Hence, if a caller abandons a call attempt and no agent ever answered it, Queue/CDR code treats the call as never having been answered and, if you don't have unanswered=yes in your cdr.conf, will not log an entry. Note that there are a few other quirks with CDRs in queues in this and related scenarios, particularly when some of the members are busy (see ASTERISK-17776). We discussed making changes to this behaviour in release branches (see https://reviewboard.asterisk.org/r/2064/), but decided against it due to the ripple effect changes in CDRs have on users. If you're running into similar behaviour, you may want to backport those changes to your version. Matt Hi Matt/anyone The only way I can get the desired behaviour is if I do a dial for one second before the queue is called. This gives me a No answer disposition if the caller abandons the call while in the queue. I tried ringing, answer and playback. The latter 2 always set the disposition to answered, even when the call is abandoned. So, is there any other application that answers the channel without setting the disposition like Dial does? Thanks Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Problem using ast_tls_cert script
Hello Kepin, I don's know if there's a difference, I changed order with the same result. Did you find a different script with CentOS? Elder On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra insanlaks...@gmail.comwrote: hi daniel, are you sure the command in debian and ubuntu same? On Wed, Feb 6, 2013 at 10:59 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O MyCompany -d /etc/asterisk/keys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -- _ -- 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] Asterisk calls between 2 private networks
AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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] Asterisk calls between 2 private networks
On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote: AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. Adding more points of failure and more devices to maintain without any real benefit is always the wrong thing to do. IAX is also flaky as hell. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Asterisk calls between 2 private networks
The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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] Asterisk calls between 2 private networks
I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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] Asterisk calls between 2 private networks
I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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] Asterisk calls between 2 private networks
Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a
Re: [asterisk-users] Asterisk calls between 2 private networks
Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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
Re: [asterisk-users] Asterisk calls between 2 private networks
I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 http://192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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
Re: [asterisk-users] Asterisk calls between 2 private networks
Did you set canreinvite=no in sip.conf on the phone in network B? A phone that can connect but loses audio is almost a sure sign that it is reinviting and your rtp packets are not making it to the phone. By turning canreinvite off, it will keep asterisk in the middle of your sessions and should give you the audio. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject:Re: [asterisk-users] Asterisk calls between 2 private networks Sent by:asterisk-users-boun...@lists.digium.com I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 http://192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that
Re: [asterisk-users] Problem using ast_tls_cert script
I'm not sure, but it looks like a command in centos and ubuntu are same ... i'am also trying to configure TLS on ubuntu but always error on the softphone blink: transport error. On Fri, Feb 8, 2013 at 12:23 AM, Daniel - Asterisk earohua...@gmail.comwrote: Hello Kepin, I don's know if there's a difference, I changed order with the same result. Did you find a different script with CentOS? Elder On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra insanlaks...@gmail.comwrote: hi daniel, are you sure the command in debian and ubuntu same? On Wed, Feb 6, 2013 at 10:59 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O MyCompany -d /etc/asterisk/keys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -- _ -- 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] Asterisk calls between 2 private networks
i think canreinvite is not part of Asterisk 1.8 anymore. Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other. On 2/7/13 1:15 PM, Kevin Larsen wrote: Did you set canreinvite=no in sip.conf on the phone in network B? A phone that can connect but loses audio is almost a sure sign that it is reinviting and your rtp packets are not making it to the phone. By turning canreinvite off, it will keep asterisk in the middle of your sessions and should give you the audio. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Sent by: asterisk-users-boun...@lists.digium.com I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the
Re: [asterisk-users] Asterisk calls between 2 private networks
And actually I did not have directmediadeny=0.0.0.0 But I had directmedia=no. So I will add the directmediadeny line, and will check it out again tonight. On 2/7/13 1:22 PM, Frank wrote: i think canreinvite is not part of Asterisk 1.8 anymore. Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other. On 2/7/13 1:15 PM, Kevin Larsen wrote: Did you set canreinvite=no in sip.conf on the phone in network B? A phone that can connect but loses audio is almost a sure sign that it is reinviting and your rtp packets are not making it to the phone. By turning canreinvite off, it will keep asterisk in the middle of your sessions and should give you the audio. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Sent by: asterisk-users-boun...@lists.digium.com I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone
Re: [asterisk-users] save the number of sip phone
Hi,
exten = _0614.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}_*${CALLER}*
.wav|av(0}V(0))
This should append the caller number in your recorded file name.
Ensure that you save the callerid in the variable before you're changing it
to MY_CALLERID
exten = _0614.,1,*Set(CALLER=${CALLERID(number)})*
exten = _0614.,n,Set(CALLERID(number)=MY_CALLERID)
exten = _0614.,n,MixMonitor(zap_g2_${EXTEN}_*${CALLER}*.wav|av(0}V(0))
exten = _0614.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0614.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0614.,n,Hangup();
Best Regards,
Sammy
On Thu, Feb 7, 2013 at 8:48 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
Hello list
i have asterisk 1.4 installed and i use MixMonitor to save the outbound
call like below
exten = _0614.,1,Set(CALLERID(number)=MY_CALLERID)
exten = _0614.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0614.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0614.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0614.,n,Hangup();
using this code i can save all outbound calls begin with 0614
in /var/spool/asterisk/monitor like this
(zap_g2_0614xx_1360219942.10291.wav)
i use 3 phones sip (222,223,224) and my question how to save the sip phone
with number like that (zap_g2_0614xx_224) for example
thanks and regards
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Re: [asterisk-users] TLS
when i start sip reload, doesn't appear about SSL certificate ok, i install asterisk with : ./configure --enable-xmldoc make menuselect make make install make samples make config ok, maybe i try using tshark later... yes, i'm sure blink is configured for TLS. and i've installed the certificate in client with trusted root certification. any ideas? thank for your attention... On Thu, Feb 7, 2013 at 8:39 PM, Administrator TOOTAI ad...@tootai.netwrote: Le 06/02/2013 23:15, kepin sinatra a écrit : Hi, I tried it the implementation of TLS in asterisk 1.8.4.3 on ubuntu 10.04. I follow the tutorial: https://wiki.asterisk.org/** wiki/display/AST/Secure+**Calling+Tutorialhttps://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. and I use blink as a softphone in ny client in windows. for regular communication process (without TLS) smoothly, but when it just follow the tutorial, it is always error on his softphone: transport error. Check that blink is configured for TLS. Also, when you start asterisk or sip reload check that message SSL certificate ok appears in your logs Other check: run tshark on the interface of your asterisk on port 5061 in tcp to check if the traffic of your softphone arrive to the good port. -- Daniel -- __**__**_ -- 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
Re: [asterisk-users] Problem using ast_tls_cert script
I did follow instructions in debian without problems, this issue arise when trying with Centos 5.8 and 5.9. On Debian 6.0.6 i wrote: ./ast_tls_cert -C 10.200.x.y -O Company -d /etc/asterisk/keys/ and I got ca.cert which is working on my Blink phones. If you have any news please let me know, Thank you! Elder On Thu, Feb 7, 2013 at 1:18 PM, kepin sinatra insanlaks...@gmail.comwrote: I'm not sure, but it looks like a command in centos and ubuntu are same ... i'am also trying to configure TLS on ubuntu but always error on the softphone blink: transport error. On Fri, Feb 8, 2013 at 12:23 AM, Daniel - Asterisk earohua...@gmail.comwrote: Hello Kepin, I don's know if there's a difference, I changed order with the same result. Did you find a different script with CentOS? Elder On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra insanlaks...@gmail.comwrote: hi daniel, are you sure the command in debian and ubuntu same? On Wed, Feb 6, 2013 at 10:59 PM, Daniel - Asterisk earohua...@gmail.com wrote: Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O MyCompany -d /etc/asterisk/keys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -- _ -- 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 -- _ -- 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] Asterisk calls between 2 private networks
On 8/02/2013, at 6:49 AM, Frank fr...@efirehouse.com wrote: I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. +1 There is nothing non standard about this but if asterisk sees the end ip address as 192.168.1.X then sip.conf will show thats a locanetl so nat support wont be required which may cause you issues With tcpdump on your asterisk box (or sip or rtp debug on on asterisk cli ) you should see where the rtp packets are going to and from when the call comes up and what sip packets are actually saying to each other But renumbering would help especially if you did want a vpn or other networking between the sites Cheers Duncan -- _ -- 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] asterisk 1.8.10.1 meetme
Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem? Thanks, -- _ -- 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] asterisk 1.8.10.1 meetme
motty cruz wrote: Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem? Let me just start by saying that MeetMe has been touched by a rather large number of patches in the 11 months and it's quite likely that your problem will be fixed if you upgrade. r373242 comes to mind in particular. Other than that though, it would be helpful if you added some additional information, such as what arguments are are running meetme with and what kinds of devices you are connecting with (SIP phones presumably?) -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://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
[asterisk-users] [FIXED] Asterisk calls between 2 private networks
Got it to work tonight. So once again this is my network: Network A: 192.168.1.x Network B: 192.168.1.x In between, the internet. Asterisk is in Network A. 1 Digium phone is in network A. Router from network A does NAT and forward (for now): - 5060 TCP/UDP to internal IP of asterisk - 10k-20k TCP/UDP to internal IP of asterisk -I know TCP is not needed, but I will remove little by little options tomorrow, since nothing was working before- Network B has 1 Digium phone, that registers to the public IP of network A. My SIP.CONF looks like that for now: [general] context=unauthenticated allowguest=no transport=udp dtmfmode=auto nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=network_a_public_ip_address directmedia=no [100] type=friend context=LocalSets host=dynamic disallow=all allow=ulaw host=dynamic secret=xxx mailbox=100@default [200] type=friend context=LocalSets host=dynamic disallow=all allow=ulaw secret=xxx mailbox=200@default nat=yes qualify=yes directmedia=no I added a file rtp.conf: [general] rtpstart=1 rtpend=10200 that's all folks ! -- _ -- 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] Where can get the latest manual our user guide
Hi, everybody, Where can I get the manual or user guide of latest asterisk version, 1.11.x? I want to know the syntax and usage of all the supported functions or something like that in the latest version. Thanks in advance Ding Peng -- _ -- 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] Where can get the latest manual our user guide
On Fri, Feb 8, 2013 at 11:05 AM, Ding Peng roc.dingp...@gmail.com wrote: Hi, everybody, Where can I get the manual or user guide of latest asterisk version, 1.11.x? I want to know the syntax and usage of all the supported functions or something like that in the latest version. Thanks in advance Ding Peng https://wiki.asterisk.org/wiki/display/AST/Home is the best place to start off with such stuffs. --Satish Barot Ahmedabad, India -- _ -- 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] Asterisk calls between 2 private networks
-Original Message- From: Carlos Alvarez car...@televolve.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Date: Thu, 7 Feb 2013 10:36:36 -0700 On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote: AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. Adding more points of failure and more devices to maintain without any real benefit is always the wrong thing to do. IAX is also flaky as hell. -- _ Carlos, with regards to your comment about IAX, where can i find your bug-report? -- _ -- 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
