Re: [asterisk-users] a=sendonly Music On Hold ignored
On Fri, Jul 22, 2011 at 8:07 PM, Matthew J. Roth mr...@imminc.com wrote: Luckily, there is an option to force Asterisk to ignore the SDP session version number and treat all SDP data as new data. Try adding ignoresdpversion=yes to the phone's configuration in sip.conf. COOL!!! OK, so when I added this line to the device's settings (somewhat complicated, as FreePBX uses a database for these fields), it didn't do anything, but then I read that I can add it in the general SIP seetings, and after I did that, I get a new message now: Music class default requested but no musiconhold loaded How can I solve this issue?? For the sake of future list readers, please respond to this post with [RESOLVED] appended to the subject line if this fixes your problem. I'll do that once it's fully resolved, and include all the steps to the solution. 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
Re: [asterisk-users] AsteriskNow install addons despite license conflict with FFA and G.729
Please help... On Thu, Jul 21, 2011 at 11:25 AM, Michael voip.quest...@gmail.com wrote: Any suggestions on how to install Asterisk addons despite the license conflict? BTW, on another system, we installed the addons first and then the paid licenses from digium and there was absolutely no problem running and installing both. It seems to happen only when Digium software is installed BEFORE the addons are. Thanks. On Tue, Jul 19, 2011 at 10:46 PM, Michael voip.quest...@gmail.com wrote: On Tue, Jul 19, 2011 at 9:49 PM, Jason Parker jpar...@digium.com wrote: Yes, but you don't have to use cdr_mysql to insert into a MySQL database. The cdr_odbc module works just fine for that. So what's the procedure required to set FreePBX CDRs active, under these conditions? How do I activate/install/set the cdr_odbc module? 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] dialplan pattern help
Thanks for the suggestion.
If I have to do this way i will check the AGI side...
Thanks
Armand
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Leif Madsen
Envoyé : samedi 23 juillet 2011 20:18
À : asterisk-users@lists.digium.com
Objet : Re: [asterisk-users] dialplan pattern help
On 11-07-23 10:30 AM, Armand Fumal wrote:
Hi all,
I need help for make a pattern for a special case that i can't find the
solution.
In my case I want to match these in one pattern:
This is the same ext that can come in 4 cases
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
42704701
exten = _X42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
042704701
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with +3242704701
exten = _XXX42704701,1,Macro(dialfax,${EXTEN:-8}); case with
3242704701
I have try _.42704701 but the parser stop to check after the point .:-(
So did you have any suggestion ?
Ya you can't match anything after the '.' in pattern matching.
I'm not sure the pattern matcher is really capable of doing what you want here.
The only way to do it really is to match less restrictively and perform a check
using dialplan applications/functions, and then if nothing is found, to fall
through.
Perhaps something like:
exten = _XXX,1,NoOp()
same = n,ExecIf($[${EXTEN:-8} = 42704701]?Macro(dialfax,${EXTEN:-8}))
same = n,Verbose(2,Did not match -- falling through)
same = n,Playback(invalid)
same = n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (the
ExecIf() application).
Leif Madsen.
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Re: [asterisk-users] Securing Asterisk
On 07/23/2011 11:39 PM, C F wrote: On Sat, Jul 23, 2011 at 1:38 PM, CDRvene...@gmail.com wrote: I beg to differ. Digium is hiding from the real world and somebody is Because you have no clue how to secure a box its someone elses fault? Of course! Does Call Detail Record need to repeat himself? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security questions
Configure a firewall to allow only known IP's.
Regards,
Mitesh Thakkar
+91 94279 07952
GTalk: mail.mthak...@gmail.com
On Sun, Jul 24, 2011 at 9:06 AM, C F shma...@gmail.com wrote:
It's not bad but it wont prevent flooding your box with register
attempts and spoofing a user agent is trivia at best.
On Sat, Jul 23, 2011 at 9:09 PM, Flavio Miranda
flaviormira...@hotmail.com wrote:
Hello everybody!
I'd like to heard from those with more experience in Security if the
following configuration is a good attempt to prevent hack:
exten = CALLER,2,Set(header=${SIP_HEADER(User-Agent)})
exten = CALLER,3,NoOp(Cabecalho ${header})
exten = CALLER,4,GotoIf($[${header}= My User Agent]?6:7)
Considering I have only one type of IP phone in my scenario.
I know, somebody with another IP phone will succeed in dial on my asterisk
but I think it will limit at one only kind of IP phone.
My question is , if there are some way to break it and use any kind of User
Agent despite this configuratio.
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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Re: [asterisk-users] One way calling on asterisk to cisco
I did duplicate cucm as cucm2. I was a bit confused as to what changed. However, it was the same results. I commented out the cucm1 instances so it was forced to use cucm2. however I still get the same results: == Using SIP RTP CoS mark 5 -- Executing [8000@myphones:1] Dial(SIP/2002-0006, SIP/cucm2) in new stack == Using SIP RTP CoS mark 5 -- Called cucm2 [Jul 23 00:57:50] NOTICE[31563]: chan_sip.c:19198 handle_response_invite: Failed to authenticate on INVITE to 'Macbook 2002 sip:2002@172.16.200.232;tag=as2fda8b5f' -- SIP/cucm2-0007 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/2002-0006' status is 'CONGESTION' Thanks, Mitch On Jul 24, 2011, at 5:02 AM, asterisk-users-requ...@lists.digium.com wrote: Message: 6 Date: Sat, 23 Jul 2011 13:04:32 -0500 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] One way calling on asterisk to cisco callmanager integration To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 019601cc4962$f9683300$ec389900$@debsinc.com Content-Type: text/plain; charset=us-ascii Try duplicating cucm as cucm2 with qualify=no and dialing on cucm2. -- _ -- 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] Error Code 101
Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal -- _ -- 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] Error Code 101
On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal Hi, The versions are as below :- asterisk18.x86_64 1.8.5.0-1_centos5 libpri-1.4.11.5-1_centos5.x86_64 WANPIPE Release: 3.5.20 Regards, Kaushal -- _ -- 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] Error Code 101
On Mon, Jul 25, 2011 at 5:08 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal Hi, The versions are as below :- asterisk18.x86_64 1.8.5.0-1_centos5 libpri-1.4.11.5-1_centos5.x86_64 WANPIPE Release: 3.5.20 Regards, Kaushal Hi, I have Package libpri-1.4.11.5-1_centos5.x86_64 already installed and latest version on CentOS 5.6, is there a rpm version of 1.4.12 for CentOS 5.6 as per http://downloads.asterisk.org/pub/telephony/libpri/ ? Regards, Kaushal -- _ -- 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
