Re: [asterisk-users] Weird IPs in Fail2ban list
I was using 1.8.8.1 and now upgraded it to 1.8.9.1. Here is a problem I have with Asterisk logging if someone can point me to the right direction. With allowguest=no, Asterisk 1.8.9.1 doesn't create anything in the full log so my fail2ban can't ban the unregistered call attempt on my server. How can this be fixed so that there is an entry in the log file for the failed attempt so the IP gets banned? Best, On Fri, Feb 10, 2012 at 10:53 PM, Paul Belanger pabelan...@digium.comwrote: On 12-01-26 11:49 PM, asterisk jobs wrote: Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. What version of asterisk 1.8 are you using? I suspect this is a bug we recently fixed in 1.8.8.0+ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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-**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
[asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. Are there any benefits to one or the other method? Or what is the best way to do something like this now-a-days? I may even be able to install Asterisk 1.8x for this purpose if it makes my job easier. Any input is greatly appreciated. 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] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
Thanks for the input but using spool files or AMI or AGI is way different from each other and that is what I want to get an input on. I do have hands on with all methods like I noted but want to know what the trend is now-a-days and what is more robust and proven out of all three. Best, On Sat, Feb 11, 2012 at 8:12 AM, David Backeberg dbackeb...@gmail.comwrote: On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote: Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular problem. -- _ -- 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] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi Brynjolfur, Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Best, On Fri, Feb 10, 2012 at 4:10 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: Hi, I forgot to add that you are free to use my code, I’ll mail it later today. ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Brynjolfur Thorvardsson *Sendt:* 10. februar 2012 09:47 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? ** ** Hi, I’m working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It’s nothing fancy but does give me a graphical presentation of calls/15minute segments. ** ** Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn’t be a major problem. I’m working on adding the option of selecting date ranges, it’s all still a work in progress! ** ** Regards ** ** Binni ** ** *Fra:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] *På vegne af *asterisk jobs *Sendt:* 9. februar 2012 16:36 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? ** ** Hi everyone, ** ** I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. ** ** Is there such a tool? ** ** 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] Is there a php script to analyse and show call detail reports from Asterisk CDR?
No, that doesn't do the job I specifically asked and installation instructions are all over the place... Thanks though. On Fri, Feb 10, 2012 at 11:36 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Don't forget about the ever-popular Asterisk-stat and the newly revised cdr-stats projects, both web based, proven, and work fantastic: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 http://www.cdr-stats.org/ --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
Re: [asterisk-users] Question for the group
There is none. We are looking to develop our own currently and in the process of hunting down best developers. We have a great deal of experience with billing systems but doing a fron-end for this purpose just requires multiple developers. You can e-mail me in private if interested in a shared project. Best, On Fri, Feb 10, 2012 at 11:34 AM, James Wystead szilvertho...@gmail.comwrote: Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: * - Email to Fax and Fax to Email - Inward DID local and 800 services - Calling card SIP based and ANI authenticated * I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? Thanks G -- _ -- 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] Question for the group
What have you looked for yet? There are no commercial ones that do all that in one. On Fri, Feb 10, 2012 at 11:57 AM, James Wystead szilvertho...@gmail.comwrote: Yes, I like the look of that. Researching it too - the commercial one looks nice too, but I don't know if there is a budget. G On Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote: I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --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 -- _ -- 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] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
May I know how the compile RPM from Digium Repo gets to install DAHDI so easily on the VM? Can you please point me to how this compilation is done so I can have my own RPM of Asterisk with all options added on (e.g. ooh323, jabber, etc...) Thanks On Mon, Jan 16, 2012 at 1:57 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/16/2012 12:52 PM, Patrick Lists wrote: On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). They have changed; DAHDI is required for MeetMe/SLA/Page, but is not required for timing. In Asterisk 10, ConfBridge can be a suitable replacement for MeetMe for many users as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**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] Weird IPs in Fail2ban list
I can't see those IPs in the /var/log/asterisk/full. I can't event see
parts of the IP address as I try *grep -o 23.20.189 full. *That is still
nothing.
I am wondering what is wrong here. This is my regex filter file:
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
.* SIP/HOST-.* Playing 'ss-noservice.gsm' .*
Thanks,
On Fri, Jan 27, 2012 at 2:16 AM, Mikhail Lischuk mlisc...@itx.com.uawrote:
**
asterisk jobs писал 27.01.2012 06:49:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1DROP all -- 0.23.20.189 0.0.0.0/0
I also get things like, 0.0.5.2, etcFail2ban seems to be working
when I am testing. Are these numbers taken from the SIP packet or the
TCP/IP protocol source because they surely are not valid addresses.
Thanks
Did you find those IPs in Asterisk log?
If so - it isn't Fail2Ban problem, for it just parses logs and extracts
substring
--
With Best Regards
Mikhail Lischuk mlisc...@itx.com.ua
--
_
-- 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] Weird IPs in Fail2ban list
That was just another weird IP showing up. On Fri, Feb 10, 2012 at 4:50 PM, dotnetdub dotnet...@gmail.com wrote: On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote: Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. Thanks -- ___ 189.20.23.0 ? __ -- 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
[asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. Is there such a tool? 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
[asterisk-users] Weird IPs in Fail2ban list
Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. 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
[asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Hello, I can do simple, yum install asterisk18-* and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *You do not appear to have the source for the 2.6.32-4-pve kernel installed.* * * 1- Based on above error and Google search I have concluded that dahdi-linux module should be installed on mother node. So, I am puzzled. How does Digium yum repository achive this without acessing the mother node? 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Any feedback is much appreciated. 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] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Thanks for all the input guys. I am using Asterisk 1.8 for this purpose. 1- So, I do I still need Dahdi? And yes conference will be used. 2- Can you please detail on compiled already code? My mother node for OpenVz is probably different from what Digium uses to compile the source. How does this work? 3- How can I compile my own source code and then move it to my OpenVZ to work just the same? Thanks again On Mon, Jan 16, 2012 at 1:57 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/16/2012 12:52 PM, Patrick Lists wrote: On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). They have changed; DAHDI is required for MeetMe/SLA/Page, but is not required for timing. In Asterisk 10, ConfBridge can be a suitable replacement for MeetMe for many users as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**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] registration not authorized - stale nonce
We get the same error with this version. On Sun, Jan 1, 2012 at 6:13 PM, Matt Hamilton mistral9...@hotmail.comwrote: I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on the same subnet, no nat. The following is the flow of messages: 1. UAC sends the registration request 2. Asterisk responds with 401 Unauthorized with a new nonce 3. UAC sends a new digest with the nonce received from Asterisk 4. Asterisk authorizes UAC and sends OK This works as expected when there is no load on the Asterisk server. For testing asterisk under load, I use Sipp (runnig at a server on the same subnet) with a very basic scenario (call is placed, moh played, sipp hangs up after 20 secs). Max calls on the system at one time is 50. My problem happens when the above UAC tries to register while the Sipp test is running. Steps 1 and 2 above happen as expected, Asterisk sends a new nonce to the UAC, but at step 3, UAC sends the old digest (old nonce) back to Asterisk. Asterisk doesn't authorize the UAC with Correct auth, but based on stale nonce warning. UAC registration expiry is 60 seconds. As I mentioned, if the test is not running, everything seems to be OK. If I adjust the max calls to 10 in the test and reduce the load, the registrations go thru. Any ideas? Thanks, Matt -- _ -- 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] Looking for partners to develop Asterisk Call Centre Applications - A call to investors and programmers
Hi everyone, We are looking to develop our own call centre application (HTML5, real-time, shiny GUI, easy access, etc...) on top of Asterisk. We are tired of using the proprietary packages that currently exist due to no proper support, expensive licensing costs, ugly GUIs, and closed nature of the applications. If there are any developers out there or those who want to partner with the project with us by investing please contact me off-list. Due it's complex nature, we have came to the conclusion that it's best to share costs and feedback to come up with an amazing call centre product. We do have an interest to release this as open-source and that is why I am posting to group. Thanks Peter -- _ -- 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 to download sample video file for Asterisk 1.8x playback?
Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4 (ulaw)): No such file or directory The file of course exists and it's chowned to asterisk.asterisk. I think it's a file format issue. So, I appreciate it someone can give me a link to a file or maybe point me a universal convertor (open-source or linux based software) that can convert my videos to Asterisk readable format. 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
[asterisk-users] How can I decipher password in SIP Packet?
I am receiving requests to register to my Asterisk extensions. I have the full SIP packets. I also do see what extension is being tried to be registered. Is there ANY WAY to know what password is being attempted? I think the appropriate term would be decode the base64 response I get from the client. Here is what I get in the SIP packet from the client: * * *Authorization: Digest username=4456678, realm=asterisk, nonce=67461340, uri=sip:mailbox, response=5a9a5f2b527ca9687c8f75705e6a2d25, algorithm=MD5* Using a base64 decoder I get this:* *å¯Zåý›çnÜkÞ¼íÏ ïžôåîšÙݹ from the response above. Of course, that is not the plain password. So, is that encrypted? How can I can I decrypt it? Thanks, On Mon, Nov 28, 2011 at 12:48 AM, asterisk jobs asteriskcod...@gmail.comwrote: Hello, I am receiving requests to register to my Asterisk extensions. I have the full SIP packets. I also do see what extension is being tried to be registered. Is there ANY WAY to know what password is being attempted? Thanks, Peter -- _ -- 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] DHCP Option 43 and pfSense + Asterisk
Hi, Has anyone succeded using DHCP Option 43 and Aastra phones to set the configuration server from a pfSense router or any other router? Sorry, if not directly related to Asterisk but I am sure the collective knowledge will pay off. I am specifically wondering what the Number, Type and Value should be in Additional BOOTP/DHCP Options under pfSense 2.0 Thanks a lot -- _ -- 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] Max channel analyser from asteriskcdrdb?
Hello, Is there a php or any other program to analyse Asterisk CDR which is stored in asteriskcdrdb. I want to know outbound and inbound channels and not the internal calls channels as well which is what CDR Stats does currently. It doesn't differentiate between those. Someone might have done a custom script to find out their monthly inbound / outbound peak lines? I appreciate a guide on this. *FreePBX reporting and CDR Stats from Areski is not the answer to my issue. 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] How can I decipher password in SIP Packet?
As the Authorization header clearly states, this value is created using an MD5 Digest (hash). Since it is a digest function, it is not reversible. It is impossible to recover the password that was used during the calculation of the response value (although given enough time and CPU resources, it is possible go through a massive list of possibilities and try each one until you find one that matches). Thanks. Based on above, I am getting that Asterisk also runs MD5 algorithm on the password and then matches the two hash digests to see if they are good or not. Is that all happens? or is there an encryption involved as well? Chance of collision of 1^128? Regards, -- _ -- 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 Jobs Spring Special - Free Postings for Employers!
Greetings, Asterisk-Jobs.com has started the spring special for all it's employers. For the next 2 months all employer postings will be free on the site. We've also added a couple new features in the past months including Print to PDF for applicants, making it easier than ever to save and reply to employers postings. Please check out the site, and register for your free login, and job postings. http://www.asterisk-jobs.com Thanks, Asterisk Jobs Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk VOIP Jobs version 2 Launched!
Greetings VOIP Job Seekers, We wanted to let you know that we've completed the revamp of Asterisk-Jobs.com. There's not much there now after scrapping version 1.0 of the site, but we expect many postings to come soon. Keep an eye on the site for the latest in Asterisk and related VOIP employment. http://www.asterisk-jobs.com Thanks, Asterisk Jobs Staff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
