Re: [asterisk-users] SDP Issue

LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - > Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, > one of tho

Re: [asterisk-users] SDP Issue

Hi Kevin, will grab a sip trace today and post it up. -- Thanks, Phil - Original Message - > On 01/23/2012 09:48 AM, --[ UxBoD ]-- wrote: > > Hi, > > > > I am attempting to make a SIP call between an Asterisk 10 server > > and an > > Asterisk 1.8 syst

[asterisk-users] SDP Issue

Hi, I am attempting to make a SIP call between an Asterisk 10 server and an Asterisk 1.8 system but when it goes to VM and the first prompt plays the line drops and I see on the V10 console: [Jan 23 15:47:04] WARNING[7859]: chan_sip.c:8944 process_sdp: Insufficient information for SDP (m= no

Re: [asterisk-users] Voicemail weirdness after upgrade

If in a multi-tenant environment be aware of https://issues.asterisk.org/jira/browse/ASTERISK-17198 as VMs cannot be forwarded :( -- Thanks, Phil - Original Message - > > Paul Schenkeveld wrote: > > exten => 5551234,n,Voicemail(1234,su) > > I'm still running 1.4 (slowly configuring

[asterisk-users] IAX2 woes

Hello all, I attempted to make a couple of outbound calls this morning and always got the busy tone. I checked the Asterisk console and was greeted with: [Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) == Everyone

Re: [asterisk-users] AMI and Dialplan

Please ignore as this was a user error! -- Thanks, Phil - Original Message - > Hello all, > This may sound an odd question but if you initiate a call using AMI > does it adhere to what has been defined in the dial plan or do we > have to write the logic into the AMI call ? > -- > Th

[asterisk-users] AMI and Dialplan

Hello all, This may sound an odd question but if you initiate a call using AMI does it adhere to what has been defined in the dial plan or do we have to write the logic into the AMI call ? -- Thanks, Phil -- _ -- Bandwidth

Re: [asterisk-users] VoiceMail and IMAP

if you "need" the space, use quotes. > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ > > UxBoD ]-- > > Sent: Monday, December 12, 2011 11

Re: [asterisk-users] VoiceMail and IMAP

un...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ > UxBoD ]-- > Sent: Monday, December 12, 2011 11:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] VoiceMail and IMAP > > Hmmm, just tried lea

Re: [asterisk-users] VoiceMail and IMAP

the backend IMAP storage. > -- > Thanks, Phil > > - Original Message - > > On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: > > > Hello all, > > > > > > I have recently upgraded to version 1.8.7.2 and have started to > > > see > > &

Re: [asterisk-users] VoiceMail and IMAP

1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - > On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: > > Hello all, > > > > I have recently upgraded to version 1.8.7.2 and have started to see > > the following errors in

[asterisk-users] VoiceMail and IMAP

Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: [ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: SELECT failed [ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: must be in SELECTED state They are not ha

Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

Could we have more hours in the day to play with all the goodiness ? cannot keep up with everything at the moment :) -- Thanks, Phil - Original Message - > I know what you mean - I'd rather have a working x-beta1 that a > failing x.0 > > -Original Message- > From: asterisk-users

[asterisk-users] [OT] Yealink T26/28/38 and Open-VPN

Hi, Sorry for an OT post but striking out a bit at the moment trying to get a response from Yealink R&D. Has anybody successfully managed to get a Yealink phone to work across Open-VPN when using tlsauth ? We really do hope that it is possible due to the benefits tlsauth offers against DoS. -

Re: [asterisk-users] Trouble with *8 Pickup

Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a completely different dialplan! This was on Asterisk 1.6 though with Snom phones. -- Thanks, Phil - Original Message - >

Re: [asterisk-users] CEL and MySQL

cel_odbc.conf and then use adapative odbc I think. -- Thanks, Phil - Original Message - > Is anyone using CEL with a MySQL backed at all? > > I've found a table schema but I'm guessing I need some sort of > cel_mysql.conf and don't even have a sample for that. > > Can anyone give me any

Re: [asterisk-users] X86_64 Compilation Issue

Thank you Dave. -- Thanks, Phil - Original Message - > On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote: > > Hi, > > > > compiling up a new installation of Asterisk 1.8.5 on CentOS 6 > > X86_64 and > > am seeing the following when running the make: > >

[asterisk-users] X86_64 Compilation Issue

Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/l

Re: [asterisk-users] Securing Asterisk

Simple answer to all this is to install http://lync.microsoft.com/ ... good luck ;) -- Thanks, Phil - Original Message - > Kevin P. Fleming wrote: > > > > 'alwaysauthreject' in not imcompliant with any RFCs; the RFCs > > define > > response codes that *can* be used to indicate (for exam

Re: [asterisk-users] file2ban

If you are using OSSEC here are some rules: local-asterisk-denied Asterisk Potentially Under Attack 1 Asterisk Under Brute Force Attack and for the local_decoder: NOTICE[\d+] \S+: Registration from ^\S+ failed for '(\d+.\d+.\d+.\d+)' srcip OSSEC can then use Active

Re: [asterisk-users] Functions not autoloading

lease ? > > -- > > Thanks, Phil > > > > - Original Message - > > > On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote: > > > > Since upgrading to 1.8.5.0 I have had to add into modules.conf: > > > > > > > > load => fun

Re: [asterisk-users] Securing Asterisk

That is pretty interesting. I am writing a similar tool but using OSSEC to identify the attacks and then share the data between nodes using Memcached and AnyEvent. Both Asterisk and Apache, or any other server that can run OSSEC, will be able to feed into the shared ban database. -- Thanks, Phi

Re: [asterisk-users] Functions not autoloading

functions exist in /usr/lib/asterisk/modules. > > How could I help to debug this please ? > -- > Thanks, Phil > > - Original Message ----- > > On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote: > > > Since upgrading to 1.8.5.0 I have had to add into modules.con

Re: [asterisk-users] Functions not autoloading

- Original Message - > On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote: > > Since upgrading to 1.8.5.0 I have had to add into modules.conf: > > > > load => func_callerid.so > > load => func_cdr.so > > > > otherwise they do not get loaded even though I have se

[asterisk-users] Functions not autoloading

Since upgrading to 1.8.5.0 I have had to add into modules.conf: load => func_callerid.so load => func_cdr.so otherwise they do not get loaded even though I have set autoload=yes. Is this something you would expect as it is different behavior to 1.8.3.0 and I do not see any issues in /var/log/as

[asterisk-users] Converged phones; akin to SwitchVox ?

Been looking at SwitchVox and how it handles mobility using virtual extensions. Does somebody have any examples on how this can be achieved with Asterisk ? I have Bria on my Android and it would be nice if I could get my office phone and/or cell to ring. -- Thanks, Phil -- ___

Re: [asterisk-users] chan_gtalk load error

- Original Message - > - Original Message - > > From: "--[ UxBoD ]--" > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Monday, July 18, 2011 11:42:25 AM > > Subject: [asterisk-users] chan_gta

[asterisk-users] chan_gtalk load error

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be

Re: [asterisk-users] Macro issue under 1.8.5

- Original Message - > - Original Message - > > On Sat, 16 Jul 2011 11:01:07 +0100 (BST) > > "--[ UxBoD ]--" wrote: > > > > > - Original Message - > > > > On 11-07-15 02:18 PM, Doug Lytle wrote: > > > > &g

Re: [asterisk-users] Macro issue under 1.8.5

- Original Message - > On Sat, 16 Jul 2011 11:01:07 +0100 (BST) > "--[ UxBoD ]--" wrote: > > > - Original Message - > > > On 11-07-15 02:18 PM, Doug Lytle wrote: > > > > --[ UxBoD ]-- wrote: > > > >> I back levele

Re: [asterisk-users] Macro issue under 1.8.5

- Original Message - > On 11-07-15 02:18 PM, Doug Lytle wrote: > > --[ UxBoD ]-- wrote: > >> I back leveled to 1.8.3 and that works fine. What am I missing as > >> app_macro has been installed okay? > > > > Macro was depreciated in 1.6 and most li

[asterisk-users] Macro issue under 1.8.5

Have just tried to test an upgrade to 1.8.5 and when making an outbound call I get: [Jul 15 18:48:52] WARNING[21038]: pbx.c:4071 pbx_extension_helper: No application 'Macro' for extension (, XX, 1) I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has been ins

[asterisk-users] Next Asterisk 1.8 Release

Hi, When is the next release planned for as very keen to get it into Production but require the call pickup fix. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

[asterisk-users] Unusual message

Hi, Needed to test follow-me this evening on Asterisk 1.6.2.17 and received the following message: == Spawn extension (international-US, 0114407590XX, 5) exited non-zero on 'Local /0114407590XX@aXX-a62a;2' -- no live channels left. exiting. I have not seen that before. What does it

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

- Original Message - > On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote: > > Are you not seeing issues with *8 call pick up then ? > > Nope, I double checked it after seeing someone saying they had issues > with it and it is fine on the installation I have. >

[asterisk-users] [SOT] Virtualising Asterisk

I know a lot has changed over the past couple of years, and even monthly, and that Asterisk running within a virtualised environment is very happy indeed. If one would only be using SIP/IAX would Xen/KVM be the best solution ? / or perhaps VServer/LXC maybe advantageous due to binary hashing. Yo

Re: [asterisk-users] Configuring Voicemail in Asterisk 1.8

- Original Message - > Dear; > > Where I can find a new documentation for Asterisk 1.8? > > Where is the wrong in that line? I see it is as 1.8 version ! > > 500 => 1234,Operator,opera...@gama.com > > Regards > Bilal > > > --- > > > You are using an old format for spe

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

- Original Message - > On Thu, 2011-05-05 at 14:13 +, satish patel wrote: > > Hi All, > > > > Just wondering is it safe to use asterisk 1.8 latest branch on > > production ? > > > > http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision > > 317100 > > > > -S > We've been run

Re: [asterisk-users] [OT] Yealink Phones

- Original Message - > > I've just started deploying these (well the T28P model) after years > of > Snom issues and they look pretty good (although the documentation is > execrable; if you thought the Snom stuff was obtuse Yealink have got > them knocked into a cocked hat!). > > Anyway, f

Re: [asterisk-users] Asterisk 1.8.3

- Original Message - > On 11-04-07 08:20 AM, Satish Patel wrote: > > Is it ture 1.8.3 is unstable? We are planning to put this in > > production. > > Please suggest me what should I do? > > > This is a loaded question, since it really depends on what you plan > to > do. What does your migr

Re: [asterisk-users] problem with crashing Asterisk 1.8

- Original Message - > My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. > It just keeps restarting. > Any pointers on log files to watch? I tried to debug it but i > couldn't find a good reason for the crashes. > Maby the box is just overloaded or something like that b

Re: [asterisk-users] doorphone?

- Original Message - > Hi, > > could anybody suggest a usable doorphone and magnetic door opener > "hardphone" system for me, please? Of course should be connectable to > asterisk. I am in the EU, should be available here. > > thank you, > Csaba > > -- http://www.cyberdata.net/products/

[asterisk-users] Multiple SIP endpoint registrations

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by htt

Re: [asterisk-users] CEL and PGSQL

- Original Message - > Pretty sure I saw those on wiki.asterisk.org . > Thanks, > --Warren Selby, dCAP > On Feb 28, 2011, at 10:39 AM, "--[ UxBoD ]--" < ux...@splatnix.net > > wrote: > > Hi, > > > Would someone know where I can downloa

[asterisk-users] CEL and PGSQL

Hi, Would someone know where I can download the CEL schema for (create commands) for PostgreSQL please ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

[asterisk-users] SRTP Error Message

Apologies in advance if this has come up a thousand times before but is there any way to stop this error in 1.8 ? [ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup SRTP session. -- Thanks, Phil --

[asterisk-users] [OT] Yealink IP Phones

Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear

Re: [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number

- Original Message - > I found some great pieces of script on the internet that I've > combined to allow Asterisk to send voicemails as an MP3 file, and > encode the sender name and number as well as message number as tags > into the MP3 file. I even include a "cover art" image which has o

Re: [asterisk-users] app_calendar and SSL

- Original Message - > Try to disable certificate verification on the app. I had never tried > it personally but check for that option. > > Sent from my iPhone > > On Jan 17, 2011, at 5:51 PM, "--[ UxBoD ]--" > wrote: > > > Hi, > > > &

[asterisk-users] app_calendar and SSL

Hi, Over the weekend tried to setup a test using the new app_calendar code but receiving the following error: [Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar: Unable to retrieve iCalendar 'testcal' from 'https://office.test.net/home/teamsh...@test.net/Calendar/':

Re: [asterisk-users] Voicemail Forwarding

- Original Message - > --[ UxBoD ]-- wrote: > > - Original Message - > > > >> > > Yes exactly that indeed. Though Asterisk appears to ignore which > > context the user is in and selects default instead. Beginning to > > think that it is a b

Re: [asterisk-users] Voicemail Forwarding

- Original Message - > Is that user trying to forward to xxx in the same context? > > On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- > wrote: > > Experiencing a problem when users attempt to forward a voicemail > > from within VoiceMailMain(Option 8) I

Re: [asterisk-users] Attack problem

- Original Message - HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any b

Re: [asterisk-users] How to find , internal, external inbound or outbound

- Original Message - > reply please > > On 12/17/2010 10:03 AM, Nikhil wrote: > > Hi > > Does anyone knows how to find out a call in a asterisk is > > external incoming ,external out going or internal > > > > Thanks > > Nikhil > > Perhaps if you were clearer in the question you ar

[asterisk-users] Voicemail Forwarding

Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The user

Re: [asterisk-users] Asterisk 1.8 crashing

- Original Message - > On Fri, Nov 26, 2010 at 8:00 AM, --[ UxBoD ]-- > wrote: > > Hi, > > > > Has anybody had 1.8 crashing for no reason at all ? It has happened > > a couple of times so far and when I check /var/log/asterisk/messages > > nothing is

[asterisk-users] Asterisk 1.8 crashing

Hi, Has anybody had 1.8 crashing for no reason at all ? It has happened a couple of times so far and when I check /var/log/asterisk/messages nothing is in there at all :( -- Thanks, Phil -- _ -- Bandwidth and Colocation Provi

Re: [asterisk-users] astcanary ?

- Original Message - Hello, I notice that the following proces is running : astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 What is this ?? Kind regards, Jonas. You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf for the line

Re: [asterisk-users] Asterisk 1.8 Release Schedule

- Original Message - > On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote: > > I have read the wiki entry but unsure when we would likely see a > > 1.8.0.1 beta or release candidate ? > > It will be Asterisk 1.8.1-rc1 and that is now available (as of a few >

[asterisk-users] Asterisk 1.8 Release Schedule

I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta or release candidate ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] SIP Extensions and loss of Internet connection

- Original Message - > Hi, Phil. > > > > A few days I have problems connecting to the Internet on my house > > > and since then my local SIP extensions are no longer registered > > > against the local Asterisk server. > > > > > > I'm using Asterisk 1.4.24.1. I was researching on the Intern

Re: [asterisk-users] SIP Extensions and loss of Internet connection

- Original Message - > Hi all! > > A few days I have problems connecting to the Internet on my house and > since then my local SIP extensions are no longer registered against > the > local Asterisk server. > > I'm using Asterisk 1.4.24.1. I was researching on the Internet and I > found th

Re: [asterisk-users] URGENT Help needed

- Original Message - > Hello, > > > We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after > trying > to install iksemel (jabber support) and spandsp, but now Asterisk > doesn't work anymore and we can't get it to run, althorugh we tried to > remove it completely and reinstall

[asterisk-users] Asterisk 1.8 and Zimbra

Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber service ? I have opened http://issues.asterisk.org/view.php?id=18198 as it keeps failing for me. Am wondering whether it is due to using a self signed cert. -- Thanks, Phil -- _

Re: [asterisk-users] SIP Load Balancing

- Original Message - I have a very simple setup with two SIP routes to my carrier. I need to have every other phone call placed to that carrier go to a different address. This is what I need the call flow to look like. I have spent many hours searching and have not found a working ex

Re: [asterisk-users] fraud advice (Also advice on using ipbanning)

- Original Message - When we designed our systems on asterisk we designed it to me multi-tenant. Se we use customer prefixes on all extensions. This allows us to have multiple customers using the same extension pools. It also reduces the hack foot print as hackers must know the prefix

[asterisk-users] Phantom phone ringing

Hi, Running 1.6.2.11 and getting the odd occation that all phones will start ringing with nobody on the other end. From the information we have received from the client we can see that a call comes in, it is either answered or not answered, but at the same time a second call comes in and it ha

Re: [asterisk-users] Help me Out!!!!

- Original Message - > Rough area. Consider yourself lucky you haven't been ripped apart :P > > Pete wrote: > > I hope someone has helped poor Rob, I would as I am just over the > > bridge > > in Bristol, UK but some evil internet scammer has stolen all my > > money! ;) > > > > Cheers! > >

Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)

- Original Message - > Roger Burton West wrote: > > > I want to hook one of them to the PSTN. Given that I am in > > the UK, what is a reasonably easily-available device to > > provide an FXO interface from a Linux box, with a minimum of > > faffing around with drivers? Just one line is ne

Re: [asterisk-users] All phones ringing when temporary loss of Internet

- Original Message - > On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]-- > wrote: > > This is a real strange one and trying to phantom it out. One of our > > clients is connected to our Asterisk installation, from two sites, > > via VPN which works great. Every so of

Re: [asterisk-users] DAHDI not detecting caller hangup

- Original Message - > On Tue, 24 Aug 2010, --[ UxBoD ]-- wrote: > > > - Original Message - > >> On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- > >> > >> wrote: > >>> Odd problem have just noticed in that when I call into the

Re: [asterisk-users] DAHDI not detecting caller hangup

- Original Message - > On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- > wrote: > > Odd problem have just noticed in that when I call into the PBX DAHDI > > detects the call and hands it off to the extension, if I then hang > > up it still continues to proc

[asterisk-users] All phones ringing when temporary loss of Internet

Hi, This is a real strange one and trying to phantom it out. One of our clients is connected to our Asterisk installation, from two sites, via VPN which works great. Every so often one of the sites VPN tunnel goes does for say a couple of seconds. When that happens all the extensions, includin

[asterisk-users] DAHDI not detecting caller hangup

Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsi

[asterisk-users] EMail on Missed Call

Hi, Running Asterisk 1.6.2.11 and wondering what would be the best way to send an email when a missed call has occurred ? I believe you can modify [stdexten] is this still the case in V1.6 ? -- Thanks, Phil -- _ -- Bandwidth

Re: [asterisk-users] OT: Grandstream GXV3140

- Original Message - > On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote: > > > Hi, > > > > Do any of you have these phones ? How have you found it ? Are you > > using them over WiFi or hard wired ? Does it play nicely with > > Asterisk ? > > > > Ne

[asterisk-users] OT: Grandstream GXV3140

Hi, Do any of you have these phones ? How have you found it ? Are you using them over WiFi or hard wired ? Does it play nicely with Asterisk ? Need to replace my Snom M3s and this phone maybe a contender. -- Thanks, Phil -- _

Re: [asterisk-users] SIP response 482 "Loop Detected"

- Original Message - > - Original Message - > > On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- > > wrote: > > > > > > - Original Message - > > >> Hi, > > >> > > >> We have tried upgrading from 1.6.1.14 t

Re: [asterisk-users] SIP response 482 "Loop Detected"

- Original Message - > On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- > wrote: > > > > - Original Message - > >> Hi, > >> > >> We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found > >> that > >> we are unable

Re: [asterisk-users] SIP response 482 "Loop Detected"

- Original Message - > Hi, > > We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that > we are unable to URI dial our clients. We run a multi-tenant server > and have set sip.conf to forward calls to a public context based on > incoming domain name. This was all working befo

[asterisk-users] SIP response 482 "Loop Detected"

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop

[asterisk-users] OT: Physical SIP phone with inbuilt VPN support

Hi, all Would any of you be able to suggest physical SIP phones that support inbuilt VPN capabilities; akin to the Snom 370/870 ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t

Re: [asterisk-users] Small PC to build and run Asterisk

> * Skype for Asterisk needs to run on this <- so this means x86, right? or x86_64 is fine -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduc

Re: [asterisk-users] Dual Atom mobo - call capacity

- Original Message - > On Thu, 10 Jun 2010, Michelle Dupuis wrote: > > > I'm looking for a small formfactor mobo for an install that needs to > > handle 25 phone sets (no transcoding). I found a new dual atom > > 1.66GHz > > mobo - anyone know what kinds of call volume that will handle? >

Re: [asterisk-users] Asterisk 1.6.2.8 Now Available

- Original Message - > The Asterisk Development Team has announced the release of Asterisk > 1.6.2.8. This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ Which release will http://issues.asterisk.org/view.php?id=17135 make it into; w

Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message - > On Friday 28 May 2010 12:50:19 --[ UxBoD ]-- wrote: > > - Original Message - > > > > > You're missing this in your chan_dahdi.conf: > > > > > > #include dahdi-channels.conf > > > > Hmm, I changed t

Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message - > - "--[ UxBoD ]--" wrote: > > /etc/asterisk/chan_dahdi.conf > > - [channels] > > language=en usecallerid=yes > > cidsignalling=v23 sendcalleridafter = 2 > > rxgain=2.0 txgain=3.0 >

Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message - > On Fri, 28 May 2010, --[ UxBoD ]-- wrote: > > > [NON-Text Body part not included] > > > Er, my mailer's obviously struggled to interpret this, however did you > do > the include and I also have this in /etc/modprobe.d in a file

Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message - > --[ UxBoD ]-- wrote: > > Looking for some help from the UK please. I backed up all my > > Asterisk configuration before re-installing the server from 32 -> 64 > > bit. Unfortunately I did not transfer the backup to another > > mac

[asterisk-users] DAHDI Help (made a cardinal sin :()

Looking for some help from the UK please. I backed up all my Asterisk configuration before re-installing the server from 32 -> 64 bit. Unfortunately I did not transfer the backup to another machine! I now have a TDM400P that is not picking up the line. Can you see what I have done wrong

Re: [asterisk-users] is my PHPAGI Soap code right?

- Original Message - > 2010/5/14 --[ UxBoD ]-- : > > > > - Original Message - > >> Hello, > >> > >> i try to use soap in the phpagi. > >> i want to call a function from a web service > >> when a user call a telephne f

Re: [asterisk-users] is my PHPAGI Soap code right?

- Original Message - > Hello, > > i try to use soap in the phpagi. > i want to call a function from a web service > when a user call a telephne failed. > > this is my phpagi script, Could you show me what's wrong ? becasue i > can't excute it successfully. > > please open the following

Re: [asterisk-users] Interesting email project.

- Original Message - > mike mosier wrote: > > > > Hey all. > > > > My boss asked me to implement the following > > > > When DID 713xxx is dialed send an email to mmos...@xxx.com > > . with the time date and CID included in the > > email. I know how to code some b

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > Randy- > > > On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy > > wrote: > >> Assuming that every such spamming/hacking/attack site is funded on > >> a stolen identity/CC number, it will soon sink into Amazon that > >> they are > >> getting a bad rep, and losing money

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

; > > On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- < ux...@splatnix.net > > wrote: > > > > - Original Message - > > Speaking of all these attacks, are there any good web managed > > security monitor tools for CentOS out there that can be installe

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > Speaking of all these attacks, are there any good web managed security > monitor tools for CentOS out there that can be installed on the system > so that it can give us a visual of let's multiple failed attempts > against SSH or HTTPd? > > > Something nice that is s

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > On Tue, 13 Apr 2010, Alyed wrote: > > > Think we need some solution WITHIN the Asterisk core. Roderick A. > > suggested something that looks nice using iptables, some others have > > pointed out using > > RBL or fail2ban, but the best would be to have some generic so

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > Think we need some solution WITHIN the Asterisk core. Roderick A. > suggested something that looks nice using iptables, some others have > pointed out using RBL or fail2ban, but the best would be to have some > generic solution not dependant on third party programs. >

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > On 04/12/2010 12:05 PM, Randy R wrote: > > On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman > > wrote: > >> I don't think anyone else brought up the Spamhaus DROP project. > >> It's a > >> blacklist of IP addresses and address ranges which are known to > >> ONLY be >

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message - > Am 11.04.2010 17:05, schrieb Mark Smith: > > Same this end from 184.73.17.150. > > Use this little piece of iptables magic to block the whole of > > Amazon's EC2 ip- > > range. > > > > iptables -F > > iptables -A INPUT -m iprange --src-range > > 216.182.224.0-216.182.

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