Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1
On 03/17/2014 01:56 PM, Eric Wieling wrote: Often it is P-Asserted-ID, but depends on the carrier. You should be asking your carrier how to do this. Be careful, if the carrier doesn't like your CID spoofing they might bill the call to a default number on the account. Speaking as a carrier that allows this, we require the P-Asserted-Identity field. This is the example of a header that we insert with our SBC: P-Asserted-Identity: sip:2325551212@1.2.3.4 The phone number is the identifying marker to tell our Metaswitch the needed information to associate the call to the correct object for billing and call restriction purposes. The IP is the internal IP of our Metaswitch. It is the internal IP due to our MetaSwitch being behind our kamailio SBC. I suspect it is the destination which is rejecting the call because toll free numbers are not considered valid, not your carrier rejecting the call. As a carrier, I have never seen a case where a call (inbound or outbound) was rejected because the received caller ID string contained a toll free number. For me, as long as it passes the number validation step, we are good. And a toll free number looks like any other NAMPA number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic Sent: Monday, March 17, 2014 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1 In a multi-tenant environment, we are sending various CallerIDs outbound from asterisk based on who the user is. We have an insurance agency who would like to present a toll free callerid. This works.. unless they're calling a toll free number. In that case, occasionally, the call fails. However, should we send a correctly formatted npanxx of a local number, the call completes. We have been advised that we can send the billing telephone number as a separate header and the call will complete, all-the-while, presenting the toll free number as the caller id. Does anyone know of the correct header required to provide this functionality? -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] SPA112 Won't stay up
On 02/06/2014 09:25 AM, Mike Diehl wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. More then likely, the replacement router/modem had a different timeout and it was a luck of the draw that it worked. In routers that allow me to set the UDP timeout, I normally set the timeout to 90 seconds. Most routers that done offer a setting for this are usually set to 90 or 120 seconds Then I usually set my registration time on the ATA's to 60 seconds. The devices I seem to have most issues with are SonicWall routers. Jim Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] VM notification to multiple email recipients
Create an alias on the mail server rather then on each asterisk box. On 09/11/2013 11:44 AM, Mike Diehl wrote: OK, and to make things even more difficult, I store my voicemail and voicemail configuration in MySql. Looks like, for now, I will be creating aliases in /etc/aliases and sync'ing that across my servers Thank you for your suggestions. Mike. On Wed, Sep 11, 2013 at 12:14 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hi You can do this, http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/ If you are using asterisk 1.8 On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.comwrote: Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple addresses? Mike -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 Log rotate not working
On 5/21/2013 11:54 AM, Ahmed Munir wrote: Checked in /var/logs/ directory, all logs are not rotating by logrotate. Please advise how can I overcome this issue as I'm using CentoOS 5 Ahmed, Proper log rotation depends on a couple things working together correctly to get the job done. First, you need to make sure you have the space to rotate the logs. If you have compression enabled, logrotate creates a copy of the file(s) as it compresses them. You could be running out of space??? Next you need to verify that everything is in place, follow these steps to do so. Keep in mind that I have CentOS 6.4. So the packages might differ a little in the name and surely in the version numbering. 1) Verify logrotate is installed to your system. # yum install logrotate if it asks you to install it, do so. 2) Verify that crond is installed and running. Below is the output I get when searching yum to see if crond is installed. If your query returns nothing then crond is not installed. [root@jim etc]# yum list all | grep ^cron | grep @ cronie.x86_64 1.4.4-7.el6 @anaconda-CentOS-201303020151.x86_64/6.4 cronie-anacron.x86_64 1.4.4-7.el6 @anaconda-CentOS-201303020151.x86_64/6.4 crontabs.noarch 1.10-33.el6 @anaconda-CentOS-201303020151.x86_64/6.4 If crond is not installed, then you will need to install it. Once you have it installed, move on to the next step. 3) Make sure crond is setup to start at boot time. chkconfig crond on 4) Verify that logrotate is in one of the cron include folders. Mine is located in the cron.daily folder. [root@jim etc]# find /etc/*/logrotate /etc/cron.daily/logrotate If you don't find that the above file exists, you might need to re-install logrotate. Next I would've had you verify that you have a config file in /etc/logrotate.d/ for the asterisk log files. But it seems you already to. After all this, if it still isn't working, double check all the steps above. Let us know if this does or doesn't help. -- Jim Lucas -- _ -- 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 SIP deadlocks - update_provisional_keepalive
On 04/03/2013 08:15 PM, Duane Larson wrote: So it just happened again on both machines at the same time and I was running debug on both servers. I am running OpenSIPS and load balancing between both servers so I am guessing when the invite was sent to the first server it was frozen for some reason and then OpenSIPS sent the invite to the second server and that server was also frozen/deadlocked because of the SIP message. I noticed on both servers the last log that was posted with Asterisk deadlocked was the following Asterisk version 11.0.1 [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11805 instead Asterisk version 11.2.1 [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12423 instead In my last email I posted the debug from the Asterisk server with 11.0.1 version of code. Here is a post of the debug for the Asterisk server with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system A number of deadlocks were fixed in the current release of 11.3. Please read the change log to see if any fit your issue. http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.comwrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 disconnecting SIP Calls after 15 Minutes
On 3/21/2013 12:31 AM, Florian Wolters wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo Florian, As both an VoIP provider and phone system vendor, I had this same problem 2 years ago. In my situation, it turned out that it was nothing to do with either the Asterisk box or the provider. The problem was with a router that we had terminating our T1 connection. As an ISP we provide T1's to many customers and we provide the router as well. In this specific case, the customer purchased a data T1 connection with QoS (sip and rtp) then purchased our IP asterisk phone system with SIP trunks from us as well. The way we found this issue was by switching our the T1 router. Turns out that it fixed the problem. Exact same configuration was on each router. So we started scratching our heads... We then looked at the firmware of the two routers and found that they were different. We provide Cisco 26XX routers. Their are many places on the net talking about the 15 minute NAT timeout issue. If you are not using this device, well, maybe it has a similar bug. -- Jim Lucas -- _ -- 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] Google voice with no voice
On 1/21/2013 7:59 PM, Frank wrote: Actually, the funny thing is that it works randomly. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. In the past, I have had strange behaviors like this as well. Turned out to be a ARP race condition with my firewall with static IP assignments. As soon as the second device would ARP, I would loose connectivity with the first device. Check that you have no other device using the IP address that your D70 is using. Also, make sure that nothing else is competing with the Google Voice registration. -- Jim Lucas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On 01/09/2013 10:20 AM, Doug Lytle wrote: I received the same spam myself. No, I did not. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On 01/09/2013 10:54 AM, jon pounder wrote: On 01/09/2013 01:49 PM, Steve Edwards wrote: I was about to reply 'no' but thought to check my spam logs so now I reply 'yes.' I got a few of them actually. -- _ -- 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 What were the senders IP(s)? -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] Top Posting
On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] sip-user status
On 12/13/2012 11:39 PM, Hans Witvliet wrote: Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of the other two change to busy. I've seen code/dialplan snippets where you could change your status by dialling a specific extension, on which asterisk will react (and change some variables accordingly), but that is not what i'm looking for. It seems that kamaillo has build-in features to react on sip-simple changes. Can i perform the same trick with asterisk? if so, how? Hans. In * this is done via hints. The devices register with * that they want to be notified when the status of what they want to monitor changes. We, when * knows that it is doing something with the device, * changes the hint status of said device and then sends the notification of status change to the awaiting devices. -- Jim Lucas -- _ -- 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] Hacked by Microsoft?
On 11/28/2012 9:03 PM, jon pounder wrote: On 11/28/2012 11:52 PM, Steve Totaro wrote: You're not serious right ? That is just the center of the country since no better location is available. On Wed, Nov 28, 2012 at 7:45 PM, J Gao j...@veecall.com wrote: This morning someone tried to make sip call through my Asterisk. My server just drop these calls and record them in CDR with IP address: Now I noticed something interesting: The hacker's IP address: 168.63.67.239 whois gave me: NetRange: 168.61.0.0 - 168.63.255.255 CIDR: 168.61.0.0/16, 168.62.0.0/15 OriginAS: NetName:MSFT-EP NetHandle: NET-168-61-0-0-1 Parent: NET-168-0-0-0-0 NetType:Direct Assignment RegDate:2011-06-22 Updated:2012-10-16 Ref:http://whois.arin.net/rest/net/NET-168-61-0-0-1 hmmm Did I just hacked by Micro$oft? Gao http://iplocation.truevue.org/168.63.67.239.html I would put it in the North East. In or around New York. With some questionable routing towards the end of its journey. $ traceroute 168.63.67.239 traceroute to 168.63.67.239 (168.63.67.239), 64 hops max, 40 byte packets 1 49.b167.bendtel.net (66.39.167.49) 0.402 ms 0.345 ms 0.320 ms 2 g0-0-0.c1.sea1.bendtel.net (66.39.191.30) 9.896 ms 9.862 ms 9.919 ms 3 six2.microsoft.com (206.81.80.68) 436.893 ms 297.630 ms 211.67 ms 4 ge-1-3-0-57.wst-64cb-1b.ntwk.msn.net (207.46.46.39) 9.850 ms 9.917 ms 9.909 ms 5 xe-0-2-1-0.co1-96c-1a.ntwk.msn.net (207.46.45.216) 14.10 ms 14.37 ms 13.984 ms 6 ge-7-2-0-0.co1-64c-1b.ntwk.msn.net (207.46.40.166) 14.938 ms 15.28 ms 15.75 ms 7 ge-2-0-0-0.nyc-64cb-1a.ntwk.msn.net (207.46.40.91) 83.664 ms 83.821 ms 83.744 ms 8 207.46.45.231 (207.46.45.231) 172.135 ms 160.999 ms 159.25 ms 9 xe-3-0-0-0.db3-96c-1b.ntwk.msn.net (207.46.42.33) 160.677 ms 158.852 ms 158.812 ms 10 10.22.179.127 (10.22.179.127) 160.594 ms 10.22.178.195 (10.22.178.195) 157.664 ms 10.175.44.3 (10.175.44.3) 160.500 ms 11 10.175.46.247 (10.175.46.247) 159.802 ms 159.636 ms 10.175.46.201 (10.175.46.201) 158.802 ms 12 *^C -- Jim Lucas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 10/25/2012 01:21 PM, Justin Killen wrote: I'm looking for an fxs- sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- 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 How about this for a setup: 4 port T1 cards (1) Digium TE405P (PCI)~$600 (used) or (1) Digium TE420 (PCI-e 1x)~$1300 (used) and then (4) Adtran Total Access 624 (TA624)~$75 (used) 24 port channel bank We use the TA624's CPE all the time. They are very hard to kill. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] A new hack?
On 11/26/2011 5:00 PM, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful to others. Why is that? Even if they don't compromise an account they are still using your bandwidth and resources on your machine. How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? Also, since both methods involve the use of iptables, where exactly is the bandwidth savings? -- Jim Lucas -- _ -- 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] Binding asterisk to two static IPs
On 10/12/2011 3:55 PM, ge...@riseup.net wrote: After reading your original message, this is clear, yes. Sorry for being sloppy. np ;) Anyone else? Would be really really great... I solved it by having two physical connections to my network. PBX E0 IP 192.168.100.36 NM 255.255.255.0 GW 192.168.100.1 E1 IP 192.168.101.254 NM 255.255.255.0 GW n/a All the phones reside withing the 192.168.101.0/24 network. I still have bindaddr=0.0.0.0 so I can talk to my provider and my phones. But on two different interfaces. That forces the communication to always come from the correct source IP addr. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ http://www.bendsource.com/ C - (541) 408-5189 O - (541) 323-9113 H - (541) 323-4219 -- _ -- 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
