[asterisk-users] Strange issues with newly rebooted machine
Hi all, After being up and running for almost 2 years, we finally had to reboot one of our servers. Now, however, it's having problems. We're using real-time configuration for SIP peers and voicemail, via ODBC. But when I run sip show peers I don't get anything. Even when I load a known peer by name, nothing happens: sip show peer voice12 load This command just returns, with no output. It did occur to me that I might be having a problem with ODBC. However, when I show the status, I get good results: odbc show db ODBC DSN Settings - Name: db DSN:db Last connection attempt: 1969-12-31 17:00:00 Pooled: No Connected: In use I'm using 10.2.1. Also, I've noticed that tab command completion doesn't work on the Asterisk console. Any ideas what is wrong here? Mike Diehl. -- _ -- 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] Strange issues with newly rebooted machine
On 8/6/13 5:30 AM, Mike Diehl wrote: sip show peer voice12 load This command just returns, with no output. throwing out a random idea since it's early in the morning and you might be in a big jam... assuming the sip isnt working correctly at all (and its not just a console issue), after asterisk is started, perhaps try core set verbose 10, core set debug 10, module unload chan_sip.so, and module load chan_sip.so . if there are any errors loading the module it may be easy to spot them. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 - WHMCS Intergration
Gday Guys I was wondering if anybody might have some ideas, other then saying to get a programmer to code something up for me. I have seen it done before, and what i am after is a module for asterisk/freepbx that will communicate with WHMCS, What i would like to achieve from this, is a section on WHMCS client profile for a security pin, and when they call through for support, it asks them to enter in their customer ID and the PIN in their profile Any reply's on or off list would be appreciated, Even other alternatives will be considered :D Have a great evening guys, And Gals :D Cheers 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
Re: [asterisk-users] Voicemail variables on email subject
I checked the raw text of my voicemail messages today and I saw pretty much the same escape sequences for UTF-8 like you did, but I do not have any display problem. You could save the message locally and hand edit it (starting with uppercase UTF instead of lowercase utf). jg -- _ -- 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] Voicemail variables on email subject
I noticed that the problem occurs when I use the variables ${VM_DUR} and
${VM_CALLERID}. Only the subject of the message, if the body is not the
problem. Using UTF or utf the same problem occurs.
Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2013/8/6 jg webaccou...@jgoettgens.de
I checked the raw text of my voicemail messages today and I saw pretty
much the same escape sequences for UTF-8 like you did, but I do not have
any display problem. You could save the message locally and hand edit it
(starting with uppercase UTF instead of lowercase utf).
jg
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Re: [asterisk-users] Voicemail variables on email subject
I checked your original message, and I guess the expected string was a little bit different: 1504|12|Teste - Rafael 1570|0:16 I can't see anything wrong with quoted printable decoding. My best guess is still the email client and its settings. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail variables on email subject
On Monday 05 August 2013, Rafael dos Santos Saraiva wrote:
Hi
I have a problem w/ voicemail, the subject message is corruption when used
voicemail variables, e.g. :
voicemail.conf
emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR}
Return:
Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?=
Expected:
Subject: 1504|12|Teste - Rafael 1570|16
That looks about sane for a subject line in UTF-8 encoding (I haven't studied
it too closely). What mail client have you been using to retrieve the e-mail?
Is it UTF-8 aware?
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Answers come *after* questions.
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Re: [asterisk-users] Strange issues with newly rebooted machine
I appreciate your quick response. I issued the commands specified and got NO output! === CLI core set verbose 10 Verbosity was 25 and is now 10 CLI core set debug 10 Core debug was 25 and is now 10 CLI module unload chan_sip.so CLI module load chan_sip.so CLI === The reason we had to reboot the machine is that we changed it's physical location, but didn't change it's IP address. As part of the restart, I also took the opportunity to rebuild a RAID-1 array. Other than that, there have been no configuration changes since the last time this worked. Any other ideas? Mike On Tue, Aug 6, 2013 at 4:36 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 8/6/13 5:30 AM, Mike Diehl wrote: sip show peer voice12 load This command just returns, with no output. throwing out a random idea since it's early in the morning and you might be in a big jam... assuming the sip isnt working correctly at all (and its not just a console issue), after asterisk is started, perhaps try core set verbose 10, core set debug 10, module unload chan_sip.so, and module load chan_sip.so . if there are any errors loading the module it may be easy to spot them. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Strange issues with newly rebooted machine
On Tue, Aug 6, 2013 at 10:47 AM, Mike Diehl mdiehlena...@gmail.com wrote: I appreciate your quick response. I issued the commands specified and got NO output! === CLI core set verbose 10 Verbosity was 25 and is now 10 CLI core set debug 10 Core debug was 25 and is now 10 CLI module unload chan_sip.so CLI module load chan_sip.so CLI === The reason we had to reboot the machine is that we changed it's physical location, but didn't change it's IP address. As part of the restart, I also took the opportunity to rebuild a RAID-1 array. Other than that, there have been no configuration changes since the last time this worked. Any other ideas? Are the phone still working? I've noticed that realtime registered peers don't always show when I do sip show peers or even sip show peer *name*. I usually only see the peer if I make a call to the peer or the peer makes a call first. Do you have rtcachefriends=yes in your sip.conf? -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange issues with newly rebooted machine
No, my phones aren't getting a response from the server. I can't even get any output from the server if I do: sip show peer name load This command usually loads the peer from the db and shows me it's configuration. In this case, I get nothing. I do have rtcachefriends=yes in my sip.conf. In fact, this server has a virtually identical configuration to one that is already running. (I sync the configurations using unison.) I don't THINK this is a configuration issue. Any ideas, though? Mike. On Tue, Aug 6, 2013 at 10:08 AM, Warren Selby wcse...@selbytech.com wrote: On Tue, Aug 6, 2013 at 10:47 AM, Mike Diehl mdiehlena...@gmail.com wrote: I appreciate your quick response. I issued the commands specified and got NO output! === CLI core set verbose 10 Verbosity was 25 and is now 10 CLI core set debug 10 Core debug was 25 and is now 10 CLI module unload chan_sip.so CLI module load chan_sip.so CLI === The reason we had to reboot the machine is that we changed it's physical location, but didn't change it's IP address. As part of the restart, I also took the opportunity to rebuild a RAID-1 array. Other than that, there have been no configuration changes since the last time this worked. Any other ideas? Are the phone still working? I've noticed that realtime registered peers don't always show when I do sip show peers or even sip show peer name. I usually only see the peer if I make a call to the peer or the peer makes a call first. Do you have rtcachefriends=yes in your sip.conf? -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Strange issues with newly rebooted machine
- Original Message - No, my phones aren't getting a response from the server. I can't even get any output from the server if I do: sip show peer name load This command usually loads the peer from the db and shows me it's configuration. In this case, I get nothing. I do have rtcachefriends=yes in my sip.conf. In fact, this server has a virtually identical configuration to one that is already running. (I sync the configurations using unison.) I don't THINK this is a configuration issue. Any ideas, though? It sounds like Asterisk is hung in general. Next step, stop asterisk altogether, edit your /etc/asterisk/logger.conf to output all to a logfile: full = notice,warning,error,debug,verbose,dtmf Then, do a 'tail -F /var/log/asterisk/full', and startup Asterisk. I'm guessing you'll be able to see some errors flow by, but more importantly, maybe the log will stop, showing you exactly what is hanging. Good luck! --Tim -- _ -- 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] Dial application b subroutine arguments not passing?
On Fri, Aug 2, 2013 at 3:05 PM, Mitch Claborn mitch...@claborn.net wrote:
On 08/02/2013 01:28 PM, Matthew Jordan wrote:
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
I'm trying to use the b subroutine of the Dial application so that
I can do some stuff with our internal applications that need to have
access to the called channel information. I can see that the
subroutine is being executed, but the arguments I pass don't see to
make it to the subroutine.
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,tc)
[dial-to-customer]
exten = s,1,NoOp(to-customer)
same =n,Wait(1)
same =n,Playback(custom/callmenow-**announce)
same =n,GoSub(sub-outbound_caller_**id,start,1)
same
=n,Dial(${TOLL}/${MMCUSTOMER_**NUMBER},,*b(dial-to-customer-**
sub,s,1,${MMCUSTOMER_NUMBER},$**{MEMBERINTERFACE},${**MEMBERNAME})*)
Use a '^' to delineate arguments pass to subroutines. This is actually
true for the U option as well. See:
https://wiki.asterisk.org/**wiki/display/AST/Pre-Dial+**Handlershttps://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
And:
https://wiki.asterisk.org/**wiki/display/AST/Asterisk+11+**
Application_Dialhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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That is not working for me either.
same
=n,Dial(${TOLL}/${MMCUSTOMER_**NUMBER},,b(dial-to-customer-**sub^s^1^fred^$george^$arrrgh))
You are missing a set of parentheses in your invocation:
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,b(dial-to-customer-sub^s^1(fred^$george^$arrrgh)))
See:
https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
Richard
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Re: [asterisk-users] Strange issues with newly rebooted machine
We got it fixed! Our co-lo is in the process of doing a network reconfiguration/relocation and had changed their MTU to 1400 during the transition. Once we did the same, everything started to work. Thank you all for your time and quick responses. Mike. On Tue, Aug 6, 2013 at 10:44 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - No, my phones aren't getting a response from the server. I can't even get any output from the server if I do: sip show peer name load This command usually loads the peer from the db and shows me it's configuration. In this case, I get nothing. I do have rtcachefriends=yes in my sip.conf. In fact, this server has a virtually identical configuration to one that is already running. (I sync the configurations using unison.) I don't THINK this is a configuration issue. Any ideas, though? It sounds like Asterisk is hung in general. Next step, stop asterisk altogether, edit your /etc/asterisk/logger.conf to output all to a logfile: full = notice,warning,error,debug,verbose,dtmf Then, do a 'tail -F /var/log/asterisk/full', and startup Asterisk. I'm guessing you'll be able to see some errors flow by, but more importantly, maybe the log will stop, showing you exactly what is hanging. Good luck! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paltel subscribers as called parties for SIP attacks (was: Malicious traffic comming from 37.75.210.90)
For what it's worth, I see similar traffic regularly from: orange.ps hadara.ps ovh.net iweb.ca scalabledns.com securedservers.com wholesaleinternet.com hostnoc.net rackspace.com hetzner.de all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Common numbers are: 972592871970 972597562803 972592170729 972595936848 972599532957 972592170729 972592539831 972592910519 972592577022 972592648299 972599146173 972592264761 972592600109 972598285108 972592910519 972599463826 972597072204 972599327923 972595813485 972598642462 972598431470 972598372537 972597248231 972598431470 … Now some of these numbers have been short-lived, others have been in use more than 2 years, like 972597562803 which seems to be sloppy tradecraft. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? http://en.wikipedia.org/wiki/Paltel Well, if the Paltel subscriber were actually abroad… say in the US or Algeria or the Philippines, but he didn't want to risk the longest arm of the call being intercepted by Echelon or similar means, then he'd find an ISP in the country which he knew that subscriber to currently be in, and scan its CIDR blocks for insecure SIP PBX's to use to contact the mobile user… relying on domestic privacy protections to inhibit spying on internal traffic to that country. Perhaps Hadara (or a Hamas cell operating within Hadara) has moved from psyops to more overt means: http://blogs.norman.com/2012/security-research/cyberattack-against-israeli-and-palestinian-targets-for-a-year I'm surprised that DHS hasn't taken more interest in this. Or perhaps they already have, and are operating deliberately insecure PBX's as honeypots. Coming soon to your AGPS+ coordinates: a Predator drone… In any case, with all the SIP (and other) abuse I've received from Hadara.ps, they've never once acknowledged a complaint I've sent in… which seems to be tacit approval of the practice. I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Given the number of inhabitants in Gaza, it seems like a statistical improbability. Certainly not random distribution. On Jan 6, 2013, at 4:36 PM, Nick Khamis sym...@gmail.com wrote: Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use 972599779558 N. -- _ -- 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] Paltel subscribers as called parties for SIP attacks
FWIW, we routinely see dodgy traffic from: ovh.net hetzner.de But since those are 2 of the larger short-term contract dedicated server vendors, I'm not surprised about that. It's so frequent that I don't even bother reporting it any more - when an abuse report is acted upon and the server shut down, another pops up to take its place. all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Likewise here. Well, not all, but a sizeable percentage of it. We're based in the UK. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? Are you perhaps reading too much into it? There are insecure servers and computers all over the internet. These are (ab)used and co-opted into botnets which are in turn used to compromise SIP servers. I suspect that it's probably a financial goal (free calls, or substantial termination payouts) rather than a political goal the perpetrators are seeking. I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Perhaps the termination payout on those numbers is particularly good, and/or regulation/investigation into abuse isn't so good? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Paltel subscribers as called parties for SIP attacks
On Aug 6, 2013, at 2:59 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: FWIW, we routinely see dodgy traffic from: ovh.net hetzner.de But since those are 2 of the larger short-term contract dedicated server vendors, I'm not surprised about that. It's so frequent that I don't even bother reporting it any more - when an abuse report is acted upon and the server shut down, another pops up to take its place. all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Likewise here. Well, not all, but a sizeable percentage of it. We're based in the UK. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? Are you perhaps reading too much into it? There are insecure servers and computers all over the internet. These are (ab)used and co-opted into botnets which are in turn used to compromise SIP servers. I suspect that it's probably a financial goal (free calls, or substantial termination payouts) rather than a political goal the perpetrators are seeking. Assuming that were true, then the financial goal would be uniformly distributed since other countries would have subscribers motivated by the same set of conditions. But the high concentration of requests going to a specific region mean that there's another factor at play. And it's axiomatic in intelligence that there are no coincidences. ;-) I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Perhaps the termination payout on those numbers is particularly good, and/or regulation/investigation into abuse isn't so good? Kind regards, Chris Ok, let's say it's higher than any other country. Then what? Once the art of hacking PBX's for free calls is perfected, shouldn't it trickle down into other markets where the reward is less, but someone else has already done the hard part for you? That 4 years later the overwhelming majority of calls continue to be destined to Paltel indicates that there are motivators unique to this region. -Philip -- _ -- 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
