Re: [asterisk-users] AMI status events with res_fax_spandsp.so
>>> Is anyone else using the AMI with res_fax_spandsp.so for real-time status? >> >> As far as I can tell there is no way to get worthwhile status/progress >> information from AMI when using spandsp. >> >> Neil Youngman > >Thanks, I figured that was the case but wanted to be sure I hadn’t missed a >configuration option or something else. For anyone else who might be stuck on Asterisk 11 for the moment I have attached the patch I used which back ports the various Fax status actions and events from Asterisk 13. It is based off of this commit https://github.com/asterisk/asterisk/commit/5c988cc4e6c5693f03080f88e3057cb7a5358597 Steven Wheeler res_fax_5c988cc4e6c5693f03080f88e3057cb7a5358597.patch Description: res_fax_5c988cc4e6c5693f03080f88e3057cb7a5358597.patch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI status events with res_fax_spandsp.so
>> Is anyone else using the AMI with res_fax_spandsp.so for real-time status? > > As far as I can tell there is no way to get worthwhile status/progress > information from AMI when using spandsp. > >> I am working on migrating a FAX application from res_fax_digium.so to >> res_fax_spandsp.so. I have noticed that the spandsp module generates far >> fewer AMI status events than the Digium module and the generated events >> contain less information. For example when sending a fax there is no longer >> an event for every page. There are just a few FaxStatus events at the >> beginning and a couple at the end but they don’t contain many details. I can >> pull the required information from the Asterisk console by running fax show >> session but that output isn’t suitable for parsing. > > The only way I have found to follow fax progress is to configure the logs to > include fax debug and "tail" the logs. ECM fax will produce log messages like > "res_fax.c: FLOW T.30 Starting page 2 of transfer". > Non-ECM fax does not produce any page information as far as I can see. > > Neil Youngman Thanks, I figured that was the case but wanted to be sure I hadn’t missed a configuration option or something else. Steven Wheeler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI status events with res_fax_spandsp.so
Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There are just a few FaxStatus events at the beginning and a couple at the end but they don’t contain many details. I can pull the required information from the Asterisk console by running fax show session but that output isn’t suitable for parsing. There doesn’t seem to be a great deal of information about res_fax_spandsp.so via Google. FaxStatus with res_fax_spandsp.so Event: FAXStatus Privilege: call,all Operation: send Status: FAX Transmission In Progress Channel: Local/1952253@from-internal-user-0001;1 Context: send_fax Exten: s CallerID: 1763210 LocalStationID: 1763210 FileName: /tmp/faxes/1526583220391_merged.tiff FaxStatus with res_fax_digium.so Event: FaxStatus Privilege: call,all Channel: Local/1952253@from-internal-user-0001;1 FAX Session: 1 Operating Mode: FAX_TRANSMITTING Result: RSLT_IN_PROGRESS Error: NO_ERROR Call Duration: 12.088 ECM Mode: yes Data Rate: 14400 Image Resolution: 204x196 Image Encoding: ENC_MMR Page Size: LT Document Number: 1 Page Number: 1 File Name: '/tmp/faxes/1526583612555_merged.tiff' Tx Pages: 0 Tx Bytes: 512 Total Tx Lines: 0 Rx Pages: 0 Rx Bytes: 0 Total Rx Lines: 0 Total Bad Lines: 0 DIS/DCS/DTC/CTC Count: 2 CFR Count: 1 FTT Count: 0 MCF Count: 0 PPR Count: 0 RTN Count: 0 DCN Count: 0 Remote StationID: '952253 ' I am using options dfzs with the SendFAX application on Asterisk 11.6-cert18. Steven Wheeler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] App VoiceMail msg_id is not unique
When multiple mailboxes are passed into the VoiceMail application the records
created in the database share the same msg_id.
For example:
[voicemail]
exten => s,1,VoiceMail(2520@my-VOICEMAIL&252@my-VOICEMAIL,s)
Will create these two records in the voicemessages table:
mysql> SELECT msg_id, msgnum, dir, context, macrocontext, callerid,
origtime,duration, mailboxuser, mailboxcontext, flag FROM `voicemessages` WHERE
msg_id='1473347378-0005';
+-++---+--+--+++--+-++--+
| msg_id | msgnum | dir
| context | macrocontext | callerid |
origtime | duration | mailboxuser | mailboxcontext | flag |
+-++---+--+--+++--+-++--+
| 1473347378-0005 | 4 |
/var/spool/asterisk/voicemail/my-VOICEMAIL/2520/INBOX | voicemail_detect_fax |
| "Steven Wheeler" <252> | 1473347378 | 3| 2520|
my-VOICEMAIL | |
| 1473347378-0005 | 2 |
/var/spool/asterisk/voicemail/my-VOICEMAIL/252/INBOX | voicemail_detect_fax |
| "Steven Wheeler" <252> | 1473347378 | 3| 252 |
my-VOICEMAIL | |
+-++---+--+--+++--+-++--+
2 rows in set (0.00 sec)
This test was done with Asterisk 11.6-cert11 running on CentOS release 6.7
(Final) Linux 2.6.32-504.8.1.el6.x86_64
Is this expected behavior?
Steven Wheeler
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk server stress test
On 20 Aug 2015, at 11:12, Sevana Oy sa...@sevana.fi wrote: Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score the quality? Using voice files for tests has more representation to my opinion. Spot the salesman? ;) Steve-- _ -- 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] packages.digium.com
On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote: On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes steve-li...@geekinter.net wrote: Anyone know where it’s gone?.. Appears to have been down all day. The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food from me. Steve -- _ -- 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] packages.digium.com
Anyone know where it’s gone?.. Appears to have been down all day. Steve -- _ -- 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] (no subject)
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote: Submission. Thanks, Uh, no problem?.. Steve -- _ -- 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] Investigating international calls fraud
Hello, I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface *Questions I have at this moment:* 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? 2) how does this typically get sorted out with the phone company? they are charging $6.25 per minute for the Texas to Cambodia calls. The phone system owners are at fault, but how have these situations worked out in the past? I'll be tightening things up, but any feedback is appreciated. Thanks, Steve -- _ -- 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] Investigating international calls fraud
The UI (or anything really) is not open to the internet. The only things open are SSH and RDP (on alternate ports). The freepbx web interface has a strong username/password. The only weakness I see is a weak secret SIP password, and default mitel admin password used. There is no provisioning server for the Mitel phones right now. The phone system is on the same subnet/VLAN as the internal network. My guess is some internal computer has a trojan which allowed attackers to do some internal configuration changes. I don't yet know how they launched an outbound call from the internal extension. On Wed, Jan 28, 2015 at 4:38 PM, Terry Brummell te...@brummell.net wrote: You don't mention if the phone is remote, or local. Although you do mention it had a default user/pass. If the UI of the phone was/is accessible from the I'net, the GUI does have the ability to place a call from it, that is one way the calls could have been placed. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steven McCann *Sent:* Wednesday, January 28, 2015 4:03 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Investigating international calls fraud Hello, I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface *Questions I have at this moment:* 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? 2) how does this typically get sorted out with the phone company? they are charging $6.25 per minute for the Texas to Cambodia calls. The phone system owners are at fault, but how have these situations worked out in the past? I'll be tightening things up, but any feedback is appreciated. Thanks, Steve -- _ -- 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] Investigating international calls fraud
Hmm the calls are made during the day (and sometimes very early in the morning). Right now it looks like someone actually made these calls. If that is the case it's somewhat comforting to know the system wasn't compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 per minute to Cambodia seems quite steep to me. On Wed, Jan 28, 2015 at 6:07 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: Le 28/01/2015 22:03, Steven McCann a écrit : Hello, Hi I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Are you sure the calls weren't actually made internally? Can you see anything to suggest the ip or mac address of the phone changed? Because for someone to take advantage of the calls (assuming they don't get cash out of ringing Cambodia) they needed to proxy through to that phone line, which maybe required them leaving some sort of device on the network. Otherwise I am guessing they got onto your PBX somehow. As suggested logs are important, including DHCP, syslog to see if anything unusual happened. Did the calls run all day or just at night when no one was around? Was there more than one call up at a time? (how many calls does the Mitel phone support?) How long were the calls? Were they varying lengths (more human like) and did they just redial as soon as they were dropped? Or were they automated to trigger as much cost as possible e.g. if the 1st minute is the most expensive then you get a lot of short calls. Good luck Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface *Questions I have at this moment:* 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? Check your logs. In the full log with verbosity 3 you can follow how calls were treated. Also the CDR should give you informations like the extension(s) who placed those calls [...] -- 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 -- _ -- 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] Investigating international calls fraud
Hi Michelle, DISA is not in use. I'll check out the SecAst product you mentioned for rebuilding the server. I'm digging into the logs to get some more information. Thanks, Steve On Wed, Jan 28, 2015 at 5:30 PM, Michelle Dupuis mdup...@ocg.ca wrote: Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxx number etc). Have a look at SecAst (www.generationd.com) - it detects callers sending too many digits, monitors digit dialing speeds, etc. to help identify and block these types of attacks. The free version is better than nothing (but if you've already suffered one $25k attack then you probably don't mind spending a bit of money). Or have a look at http://www.voip-info.org/wiki/view/Asterisk+security for other ideas. There were some (at least one) critical FreePBX weaknesses discovered this summer (you'll find them if you google). Even if you don't expose the management interface to the internet, don't trust FreePBX security alone. -MD- My opinions expressed are my own and do not necessarily reflect those of my employer. However, as an employee of Generation D Systems my opinions are probably biased. From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Administrator TOOTAI ad...@tootai.net Sent: Wednesday, January 28, 2015 5:07 PM To: Asterisk Users List Subject: Re: [asterisk-users] Investigating international calls fraud Le 28/01/2015 22:03, Steven McCann a écrit : Hello, Hi I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface *Questions I have at this moment:* 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? Check your logs. In the full log with verbosity 3 you can follow how calls were treated. Also the CDR should give you informations like the extension(s) who placed those calls [...] -- 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 -- _ -- 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] Subscribe event ua-profile
On 10 Nov 2014, at 13:01, Norman Laidla norman.lai...@telegrupp.ee wrote: Well, pants. It actually is causing a problem, because the phone doesn't use any other methods to register to Asterisk. This is a bit of a big issue. What softphone is it? It sounds like rather odd behaviour. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 stable?
The Asterisk 13 is already stable for production environment? It’s only been out a couple of days - hard to make judgements just yet. But it is out of Beta so should be of reasonable quality - give it a try :) Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Phone ( Telecom feature )
On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote: Kindly note that I asked about the capability of the phones and now I am asking about the way I can do it to my client's phones, because he is asking for a demonstration. Yet you’ve not even told us the phones in use. You can’t just expect a mailing list to do your work for you. You need to look at the handsets and see what they support, and how they support it. You don’t even say if the handsets are SIP or not, the PSTN connectivity is pretty irrelevant. Sales Executive Engineer That explains that one. Steve -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be wrote: So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve-- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote: Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ? As far as I know that’s going to require a source change. May not be the case with PJSIP though - not used that yet. Steve-- _ -- 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] AMI Elastix
On 18 Aug 2014, at 09:27, Усин Айбек prince...@gmail.com wrote: I have trouble with connection to AMI 1.1 wich enabled on Elastix Asterisk Call Manager/1.1 Action: Login Username: admin Secret: qweasd123 Response: Error Message: Missing action in request You are missing the newline characters.. Action/Username/Secret should be on their own lines. Read the AMI spec. Steve-- _ -- 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] The plain old PBX functionality
On 8 Aug 2014, at 06:05, Gergo Csibra csi...@gmail.com wrote: back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search this functionality in Asterisk. What versions, and what extension functions (or other settings), and what VoIP phones can do this? It’s called presence, it’s in every version of Asterisk you’re likely to find. Most SIP handsets support it. All of it depends on how you configure it. Steve -- _ -- 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] Unregistered ports on SPAxxxx
On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote: All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10. If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly… Freezing and requiring power-cycle, clocks stopping (and showing minus figures!) and major struggles downgrading again. Had about a dozen of them doing the same, eventual downgrade to 6.1.3 and it’s all happy. Steve-- _ -- 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] Call Identifier Logging
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Which version was that added? I don’t see it on my 11.10.0
[daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link
[daffy-01 ~]#
According to funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
Steven Wheeler
--
_
-- 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] Call Identifier Logging
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Which version was that added? I don’t see it on my 11.10.0
[daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link
[daffy-01 ~]#
According to funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
Steven Wheeler
Where is this documented?
It does not appear to be documented. However, there is a reference in the
Asterisk: The Definitive
Guidehttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Monitoring_id246945.html.
Steven Wheeler
--
_
-- 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] Call Identifier Logging
Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} | grep
Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n
-f1):0:-1};
Att,
Rafael dos Santos Saraiva
This isn't a suitable long term solution as it requires launching several
external processes just to gain access to an internal variable. It is also
likely to create bugs in the future if someone changes the output of that
command. For instance if they fix the typo in Call Identifer.
Steven Wheeler
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Identifier Logging
Hello,
I am working on upgrading from Asterisk 1.8 to Asterisk 11.6. One of the
features we are excited for is Call Identifier
Logginghttps://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging.
However, it doesn't appear that this new Call ID is accessible from the dial
plan. Ideally we would like to store this Call ID in the CDR. Does anyone know
if this is possible?
I could do something like this, but it seems like a terrible hack:
same = n,Set(CALLID=${SHELL(asterisk -rx core show channel ${CHANNEL} | grep
' Call Identifer' | egrep -o 'C-[0-9a-f]+')})
Also as a side note, in the core show channel output ' Identifier' is misspelt
as ' Identifer'
Steven Wheeler
--
_
-- 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] Attack on Sip server.
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com wrote: There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. If you can capture the packet, surely you have the IP? If they intend to get the response then the IP header can’t be forged. Steve-- _ -- 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] autoservice.c MAX_AUTOMONS
Hello, I am currently load testing some new hardware and have been receiving the following warning. Does anyone happen to know if there are any risks or performance implications for increasing the MAX_AUTOMONS value? The current value is 1500. asterisk[30322]: WARNING[30423]: autoservice.c:110 in autoservice_run: Exceeded maximum number of automatic monitoring events. Fix autoservice.c Steven Wheeler -- _ -- 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] Elastix Architecture
On 2 May 2014, at 10:07, upendra uppi...@gmail.com wrote: Am new to Elastix and wanted to try build new modules in the Elastix , so i want to know how the PHP is running ?? as i see no Apache server inside ?? so wanted to know how its running ? which server and architecture? This is not an Elastix mailing list, and even if it was I doubt there is sufficient information there for anyone to help you. There are multiple appliances and software versions, without you saying what you’ve actually got, it’s going to be hard to help. Steve-- _ -- 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] Putting a notice in the logs from the dialplan
On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.ukmailto:i...@pack-net.co.uk wrote: Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance What about the Log application? It is available on our Asterisk 1.8.26 box. Connected to Asterisk 1.8.26.0 Verbosity is at least 3 CLI core show application Log -= Info about application 'Log' =- [Synopsis] Send arbitrary text to a selected log level. [Description] Sends an arbitrary text message to a selected log level. [Syntax] Log(level,message) [Arguments] level Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' or 'DTMF'. message Output text message. [See Also] Not available -- _ -- 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] Numbers hackers call
On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... It’s an international call to +972595XX, tried with the 00, 001 and no prefix What is confusing? Steve-- _ -- 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] Numbers hackers call
On 26 Mar 2014, at 16:20, Michelle Dupuis mdup...@ocg.ca wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) I never mentioned the 972 area code. It’s a country code - and as others have said it’s been mapped to a Palestinian mobile network. I’ve added this to my bar list - I’ve seen quite a lot of toll fraud to Palestine (and the middle east in general in recent months). If you’re referring to country code, then the 0 of the local number is dropped when dialled internationally, see: https://en.wikipedia.org/wiki/Telephone_numbers_in_Israel I'm guessing you have nothing to add to the discussion? Think what you will. Steve -- _ -- 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] Spammer direct replying to those posting on the users list
On 25 Mar 2014, at 14:16, Digium's Asterisk Development Team asteriskt...@digium.com wrote: We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. There’s plenty of people harvesting the list archives too, I get loads of spam about gateways etc :( S -- _ -- 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] IAXModem or T38Modem?
On 25 Mar 2014, at 15:00, Jeff LaCoursiere j...@jeff.net wrote: On 03/24/2014 05:50 PM, Thorolf Godawa wrote: But your carrier has to support T38, when we began to evaluate this some years ago, this was not true for all. Would you share the provider you are using? I have had almost zero luck so far. What country are you in? Every carrier we’ve tried in the UK supported T38 without problem (so far.. I expect there is a few that don’t). Not sure I can name specific carriers as this is the non-commercial list though. Steve -- _ -- 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] Enterprise VoIP Trunk
Probably should post this to the asterisk-biz list. This is the non-commercial discussion list. Post to the -biz list and you’ll probably have loads of sales droids happy to help :) Steve -- _ -- 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] Realtime Call Queues : call members in certain order
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, February 27, 2014 7:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain order On 13-02-14 17:33, Steven Wheeler wrote: From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, February 13, 2014 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain order On 12-02-14 16:58, Steven Wheeler wrote: From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, February 12, 2014 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table queues and table queue_members). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy lineair and I put the members into the table queue_members in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No| yes | 5 | 10 | 0 | NULL | linear | strict | strict | NULL | NULL| NULL | NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member queuemem4 is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang
Re: [asterisk-users] SIP OPTIONS storm?
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). S -- _ -- 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] Realtime Call Queues : call members in certain order
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, February 13, 2014 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain order On 12-02-14 16:58, Steven Wheeler wrote: From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, February 12, 2014 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table queues and table queue_members). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy lineair and I put the members into the table queue_members in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No| yes | 5 | 10 | 0 | NULL | linear | strict | strict | NULL | NULL| NULL | NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member queuemem4 is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is queuemem1. How come ?? Kind regards, Jonas. Jonas, We encountered the same problem. It is a bug in the Queue application. The Queue application actually orders members by their interface value. Here is the bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 which was closed as Not A Bug by Digium. We worked around this by prepending an integer (001__, 002__
Re: [asterisk-users] Realtime Call Queues : call members in certain order
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, February 12, 2014 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table queues and table queue_members). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy lineair and I put the members into the table queue_members in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No| yes | 5 | 10 | 0 | NULL | linear | strict | strict | NULL | NULL| NULL | NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member queuemem4 is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is queuemem1. How come ?? Kind regards, Jonas. Jonas, We encountered the same problem. It is a bug in the Queue application. The Queue application actually orders members by their interface value. Here is the bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 which was closed as Not A Bug by Digium. We worked around this by prepending an integer (001__, 002__, ...) to the interface in the database table and then removing it later in the dial plan. Hope this helps. Steven Wheeler -- _ -- 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
Re: [asterisk-users] how can I get authenticate from my own server?
On 17 Jan 2014, at 02:18, Sean Darcy seandar...@gmail.com wrote: I'm used to seeing fraudulent attempts to authenticate, But now I'm getting them from the server itself. I have an asterisk server behind a firewalled router. The local subnet is 10.10.10.0/24, the server is 10.10.10.100. Now I'm seeing in the log lots of: Failed to authenticate device *00sip:*00@10.10.10.100:5060;tag=9c565e6e How can this happen? I’d get an actual SIP trace rather than relying on the logs. If you get it at IP level, it’s a little harder to spoof (i.e. sometimes the SIP headers contain nonsense) Steve -- _ -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. Why? it’s what contexts were invented for. Steve -- _ -- 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] IAX2 bridge failing
Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis mdup...@ocg.ca wrote: meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [ mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [ jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 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-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 -- *Steven Davis* VoIP Engineer Multi Service +1-913-663-9748 o +1-913-871-5155 m stda...@multiservice.com http://www.multiservice.com/ -- -- This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email.. -- -- _ -- 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] Multiple IAX2 Trunks Load balancing
On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Sounds wonderful. S -- _ -- 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] Queue linear unordered feature when using realtime
From: Leandro Dardini [mailto:ldard...@gmail.com] Sent: Thursday, November 14, 2013 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue linear unordered feature when using realtime Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify orders in the list of results, so the members for the queue are returned in random order. Anyone experiencing the same problem? How do you solve it? Leandro I opened the ticket you linked to. We ended up prefixing the interface value with an integer which indicated the agent's position in the queue. In our dialplan this ended up looking like 'Local/001-agent@queue/n' our 'queue' context then strips off the prefix and continues as normal. Steven Wheeler -- _ -- 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] Hack
On 18 Oct 2013, at 04:06, John T. Bittner j...@xaccel.net wrote: Today I was hacked but caught it very quickly. This is the weird part, they hacked an IP Auth based account by simply knowing the account name. How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I used a dictionary based account name but how did they bypass the set ip I had under the account for this host. Did the IP show under sip show peer xxx? If it's realtime it's possible to set it and need to prune it / sip reload. Steve-- _ -- 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] RTP port ranges
On 13 Sep 2013, at 11:44, A J Stiles wrote: In the Windows world, where you usually don't get the Source Code, you never know what is running on your computer; in which case, you are never sure that there isn't a daemon listening on a particular port number, so it is wise in that case not to leave ports open unnecessarily. (Though not half as wise as just not running un-audited software in the first place .) Netstat will tell you what's running on Windows, just like on other platforms. Steve -- _ -- 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] Introducing Sippy Cup: SIPp Load Testing Made Easy
On 27 Aug 2013, at 15:34, Ben Klang wrote: But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP audio. If you've ever needed to drive an IVR from SIPp you're probably familiar with the pains - it usually requires capturing an actual call, isolating the RTP, and then giving it to SIPp to play back. Sippy Cup makes that easier by actually generating uLaw silence interspersed with appropriately timed RFC4733 DTMF. That alone has saved us tremendous time when tweaking our load test scenarios. That's awesome. Steve -- _ -- 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] Cut off last character of EXTEN
On 20 Aug 2013, at 12:25, Pat Collins wrote:
Here ya go:
112233# use ${EXTEN:0:6})
123# use ${EXTEN:0:3})
123456789# use ${EXTEN:0:9})
I think 'variable length' implied 'unknown length'...
S--
_
-- 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 6 Aug 2013, at 19:28, Mike Diehl wrote: 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. PMTU should take care of that. Are you blocking ICMP somewhere? S -- _ -- 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] Microsoft CRM Integration
On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote: I’m hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Hi, You've not said what you're trying to integrate... Creating tasks for calls, contact lookups, automatic case creation. Either way, all possible with ODBC and FreeTDS. Steve-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP. Call-limit dialstatus
On 3 Jul 2013, at 12:28, I.Pavlov wrote: [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: Call to peer '0014' rejected due to usage limit of 1 -- Couldn't call 0014 == Everyone is busy/congested at this time (0:0/0/0) -- Executing [0014@sub_pbxdialco:50] NoOp(SIP/1295-01f8, CHANUNAVAIL) in new stack I think that isn’t correct. Is it possible to change dialstatus and CDR(disposition) to BUSY-value when call-limit reached? You could look for CHANUNAVAIL in dialplan and run Busy(), bit of a workaround but may work I've not actually used the Busy() app recently, but I assume the CDRs would work with that? S -- _ -- 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] Jabber
On 23 May 2013, at 10:49, bilal ghayyad wrote: Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to communicate with them. But, how much jabber channel in asterisk is stable and updated? You can find out the support status from menuselect https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve -- _ -- 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] POKE from command line
On 26 Feb 2013, at 16:52, Gary Carr wrote: Is it possible to issue the POKE to a end point from the CLI? Our asterisk servers is not seeing some end points drop off and I would like to create a script to manually check end points. http://www.geekinter.net/iaxping.txt May be of use to you. Just dug it out of my subversion repo of useful bits so make need some poking (excuse the pun) to get it running. No warranty etc etc. Steve -- _ -- 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 / check / tool
On 18 Feb 2013, at 17:03, Christopher Harrington wrote: On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.com wrote: ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, etc.) I'd like to see an AGI written using Fortran or Cobol. Don't tempt me ;) S-- _ -- 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] RPM updates
On 28 Jan 2013, at 13:55, Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers for the replies regarding alternative repos. I'm looking to keep using the Digium ones, but they're still broken. Guess I'll just have to wait until someone at Digium notices :S Steve -- _ -- 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] RPM updates
On 6 Feb 2013, at 20:06, Rusty Newton wrote: - Original Message - From: Steven Howes steve-li...@geekinter.net On 28 Jan 2013, at 13:55, Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers for the replies regarding alternative repos. I'm looking to keep using the Digium ones, but they're still broken. Guess I'll just have to wait until someone at Digium notices :S I'm not involved in the build process for RPMs, but it sounds like they are waiting on the dahdi-linux 2.6.2 release to finish the new set of RPMS. I'd throw out an estimate of 1-3 weeks. Hi Rusty, thanks for the update. Sounds like it's being done to save wasting time building both the old and new dahdi-linux against the new kernel. I can see why that might be done. Makes it a bit awkward if we cant build a PBX by just yum installing for a few weeks. I'll have to try rolling back to an out of date kernel on the box involved - my project is on around the 4 week mark so it'd be cutting it a little fine otherwise. Cheers for the update. Steve -- _ -- 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] CallerID external call after Attended Transfer
On 4 Feb 2013, at 12:53, Jonas Kellens wrote: I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the colleague sees on his IP-phone. In step A the colleague sees the CallerID of the receptionist, which I normal. In step B, after I am connected to my colleague, the colleague still sees the CallerID of the receptionist (and not my cellphone number). How come my colleague does not see my cellphone number ? What is the correct setting ( IP-phone ? Asterisk ? ) to obtain this functionality. It's called connected line ID (it sends clid updates when things change). Asterisk supports it in recent versions (i believe 1.8 is sufficient) - your handsets may or may not (their method of transfer, and their ability to process the updates can affect it's workability). Given you've not mentioned your handsets, we cant make that judgement for you. Steve-- _ -- 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] CallerID external call after Attended Transfer
On 4 Feb 2013, at 13:45, Jonas Kellens wrote: The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? Quick google doesn't turn up any results. Handsets probably dont support it. Steve-- _ -- 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] RPM updates
Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On 10 Jan 2013, at 02:10, Jai Rangi wrote: I have removed yours right away. Yes, I agree, But just like any company we have purchased/collected email from different source. Also just like any company we are not perfect, we make mistakes. Then buy your addresses from different sources, remove EVERYONE from that source - it's clearly unclean data. And fck off, that too. Steve -- _ -- 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] Verizon SIP trunking Field Trial
On 3 Jan 2013, at 15:13, Michael L. Young wrote: So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 2 Jan 2013, at 15:54, Eric Wieling wrote: On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk wrote: There is a sanction. People like me will score top posters lower and soon not see their posts at all. I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
We should all top *AND* bottom post! On 31 Dec 2012, at 06:03, isr...@gmail.com wrote: Just my pitch in to post From a blackberry you can only top post there is no way of bottom posting So if I would have to wait to get to a computer to bottom post I would just never answer We should all top *AND* bottom post! (tongue firmly in cheek here..) S -- _ -- 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] ODBC Connection Problem
On 10 Dec 2012, at 16:13, Christopher Harrington wrote: On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Password= c3podb@2012 In case you didn't realize you were sending this out publicly to a publicly archived and searchable list, you might want to change that password now. Hostname address is RFC1918, he'll probably be ok ;) S-- _ -- 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] Why all the 401 Unauthorized
Hi, SIP registrations typically try to register, are them prompted for a password (via a 401 message) it then sends a new request with authentication . This is normal. Steve On 23 Oct 2012, at 13:26, Jerry Geis wrote: I have a connection between two asterisk boxes, both running 1.4.43 The connection is alive and good and working. however, I see a bunch of 401 Unauthorized messages using wireshark - then it eventually registers again just fine. Why would it not successfully register the first time - every time? Jerry -- _ -- 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] realtime sip peers status
On 22 Oct 2012, at 15:21, Ishfaq Malik wrote: On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote: I have successfully setup Asterisk realtime. Now I can create extensions dynamically. But when I put this command on cli mode sip show peers it returns no result. can any one guide me to fix this problem. The extensions you have created will not show up in the cli command of sip show peers until the sip extensions have tried to connect to the asterisk server. You may also need to cache realtime peers for some of the stats you're probably after. There are plenty of guides online for this. Google is your friend. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as IVR using 3g usb modem
On 18 Oct 2012, at 16:50, Mitul Limbani wrote: U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Look up chan_datacard (i think that's what it's called from memory). Steve -- _ -- 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] using analog phones
On 20 Aug 2012, at 01:37, Noam Birnbaum wrote: A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? You'd need an ATA or an analogue card. And how well they work, depends on if they are 'true' analogue, or have extra digital pairs etc. 2. Would they be losing any features that they currently have with their analog PBX? You'd need to find out what features they have on their PBX, and compare this to FreePBX. Steve -- _ -- 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] Problem provisioning Cisco SPA303
On 1 Aug 2012, at 07:05, Support wrote: I've looked on Cisco's website and Googled around, but I can not find a true example of a provisioning file for this device. Anything I could find would be enough for me to make a template. Download the SPC tool from the Cisco site. It includes the capability to generate examples. You can also do http://phoneip/admin/spacfg.xml to get the full config. Reading up on examples for the spa941/942/504 might help - they're a bit more common. S -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On 1 Aug 2012, at 12:39, D Tucny wrote: HP refuse to let you put your data at risk, refusing to activate write caching without a charged battery attached or NV cache. I personally would like to have the option to override things like this at my own risk, but, HP don't give you that option. You can enable it in the ACU (if you install it). There's an override in there. S -- _ -- 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] Digium IP Phones D40
On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote: Actually we get that complaint a lot too (Polycom ring volume). We typically install in hotel environments, and in their back office the environment can be noisy, as well as in their restaurants. I imagine in a typical office environment this wouldn't be an issue... The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The newer SPA5xx series have been seriously toned down. I'm guessing it's all 'safety' related :S Steve -- _ -- 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] Fwd: Flashphoner
Thought this deserved a name and shame! ;) Steve Begin forwarded message: From: Pavel Ismailov pavel.ismai...@gmail.com Date: 27 April 2012 06:58:07 GMT+01:00 To: steve-li...@geekinter.net Subject: Flashphoner Hello! My name is Pavel Ismailov and I`m CEO of www.flashphoner.com project. We noticed that you quite active in Asterisk-user mail list, and would like to offer you buy signature in your messages for some monthly price. Is it interested for you? -- Thanks, Pavel Ismailov skype: pavel.ismailov www.flashphoner.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] Company info
Can't tell if this is a transparent attempt at advertising, or...? S On 19 Apr 2012, at 22:09, Josué Conti wrote: This is your website: http://www.convergia.com/ Thanks in advanced for any informations. Best Regards Josue Em 19 de abril de 2012 17:11, Josué Conti josueco...@gmail.com escreveu: Dear all, Please let me know if anybody have informations about a company called Convergia, like your products, ASR/ACD or more details. With Best Regards Josue -- _ -- 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] Voicemail crashs asterisk
On 3 Apr 2012, at 16:42, Vik Killa wrote: #disasterisk fail #freeswitch win #unhelpful comment S -- _ -- 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] concurrent channels limit
On 30 Mar 2012, at 10:14, Syco wrote: Finally the problem is: I cannot manage more than 80 concurrent calls. What happens on the 81st call?.. S -- _ -- 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] unsubscribe
On 30 Mar 2012, at 10:04, Sean McMaster wrote: asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It's not tricky.. Really.. It's on the bottom of every email.-- _ -- 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] use of Read cmd with AGI
On 2 Mar 2012, at 11:35, Kamlesh Kumar wrote:
$agi = new AGI();
$agi- exec('Background','press_one0press_two0press_zero0');
$agi- exec('Read','NUMBER,,1,3');
$agi- verbose (You have entered.$NUMBER);
You need to use AGI to read the Asterisk variable.. Asterisk variables don't
magically become PHP ones.. Or you get Asterisk to process it instead using
${NUMBER}.
S--
_
-- 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] India Pune Pri call problem
On 13 Feb 2012, at 12:06, virendra bhati wrote: You can't set callerid for outgoing calls in case of PRI. Why not? Every PRI I have used supported it. Is this a carrier-specific thing? S -- _ -- 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] Stumped about adding a semi-colon to a variable
On 11 Feb 2012, at 13:41, Kevin P. Fleming wrote:
At this time, there's no way to do it directly in the dialplan
In extensions.conf
[globals]
SEMICOLON=;
Then use ${SEMICOLON}
S--
_
-- 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] Major changes between 1.4/1.6.1.8/10?
On 9 Feb 2012, at 11:08, Gilles wrote: Does someone of a good site/blog that keeps track of new releases of Asterisk, and explains what the major changes/features when they do occur? Why not just use the latest version?.. S -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On 7 Feb 2012, at 14:27, Gilles wrote: On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett rmudg...@digium.com wrote: The UPGRADE.txt and CHANGES files do just that. They have been a part of the Asterisk source files for a long time. Thanks for the info. The problem is that the ChangeLog files http://downloads.asterisk.org/pub/telephony/asterisk/releases/ are very long to read, and make no distinction between tiny features/bug fixes and major changes, so non-experts are unable to tell them apart. The upgrade files may be more to your tastes than changes files. There is no comparison chart that I know of. Just use the latest version that has a support-lifetime suitable to your needs. S -- _ -- 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] Securing Asterisk
On 27 Jul 2011, at 17:11, CDR wrote: This is turning into a political issue such as the one in Washington and the impending default on US debt. No, YOU are turning this into a political discussion. The point is that a minor change in the code would have a dramatic effect on security, and carry a lower impact on CPU that using Iptables. The simplicity of the change cannot understated. The hackers do not continue sending packets with new REGISTER attempts unless they see a response. The would move on. Much as they do after you firewall them out. Have you ever tried? No? Too busy blaming others is suspect. Digium is being monarchical about this. Why do you keep blaming Digium? Asterisk is made by a community. It looks like a loss of contact with reality. Couldn't agree more. The vast ecosystem of Digium is made of hundreds of people like me. I am being forced now to place Opensips in front of Asterisk, in port 5060, set Asterisk to listen at Port 5061, and block access to 5061 from outside. Instead of a minor change, I have to bring a second application to the picture. There, problem solved. The reason why I find useless using iptables and a rule that bans an IP address if it communicates more than a threshold of times, is simple. I have customers that hit me 10+ times per seconds from the same IP. It would look like a hacker, and it is not. Which is why you don't use packet count, you look in the logs for auth failures. I use a cluster of Asterisk in the same box, a big server, and each asterisks listens in its own network interface, and responds from it. It does work. But iptables or fail2ban would not work in a wholesale scenario. Any way, thanks for your attention. Sure it would. If they're hacking one, you can block them from the lot.. I see no problem. Just make it look at all of the logs. S -- _ -- 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] Monitoring connection to VoIP provider?
A quick to implement open source network monitoring tool is smokeping: http://oss.oetiker.ch/smokeping/index.en.html Tobi Oetiker and Niko Tyni's awesome tool can monitor latency on a number of layers, and maintains charted records of connection quality. It has a probe specific to SIP: http://oss.oetiker.ch/smokeping/probe/SipSak.en.html While there are many great implementation guides out there, I've drafted a basic install (that is somewhat OS X centric) at: http://www.stevenstromer.com/grok/smokeping-installation-for-os-x-10-5-10-6 And a basic configuration guide: http://www.stevenstromer.com/grok/smokeping-configuration-for-a-home-or-small-business-network (it doesn't describe implementing the SIP probe just yet) Hope you find smokeping as helpful as I have, and not only for VoIP services. I am sure there are a number of other, more dedicated and equally useful apps. Also, I believe there have been numerous previous discussions in this thread about monitoring options. Look back. Steven On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com wrote: Community can help you better if you provide some details about you scenario and requirement. It's a very simple scenario: The Asterisk server is connected to a VoIP provider for calls to the PSTN, and I'd like to have Asterisk (or some other app) monitor the connection so that I can tell how good it is at any time, especially before calling out or receiving a call. The VoIP provides doesn't support any tool, eg. iperf. Is tracert/ping the only tools available in that scenario? Thank you. -- _ -- 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] Asterisk 1.6.2.19 RPM
Hi All, Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a timescale for this reaching the RPM repository? We're badly affected by a bug in previous versions that has only recently become apparent to us. It's in a situation where rebuilding from source isn't too practical so we need to rely on RPMs. Failing that, is there a way I can build an RPM for myself on another box and have the right compile-time options (i.e. matching what would be used for a real RPM)? Thanks in advance.. Steve -- _ -- 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] Re : Re : Re : Re : Direct RTP with Asterisk
On 20 Jun 2011, at 16:33, Sagbo Romaric wrote: Any client behind his NAT can talk with another behind his NAT. Still not possible.. The internet doesn't really work like that. SIP even more so. S-- _ -- 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 CONFIGURATION
On 15 Jun 2011, at 11:20, mahesh katta wrote: i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG A lot of filesystems are case sensitive. Maybe you wrote your configuration in caps? This would also explain why you couldn't provide anything from the logs ;) S -- _ -- 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] Voxbone numbers
On 3 Jun 2011, at 11:57, devr devr wrote: My query now is willl all voxbone numbers show up as the operator as Voxbone SA as above. I wanted to find out who the service provider is on some numbers, I suspected the service to be voxbone but the operator shows as other companies. My idea on how voxbone works is that voxobone is a intermediatary enabler with the actual hardware with third parties in which case the operator will show as the hardware owner. Is this acuratrate? If you're trying to find the carrier for a number in the UK, start here: http://www.ofcom.org.uk/static/numbering/index.htm Won't cover ported numbers though. S-- _ -- 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 about null routing calls to DIDs we don't handle
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing loop.
Put this line:
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here get
routed upstream
in the 'local' context instead of the other one? Letting a carrier use you
as a carrier seems like quite a bad idea generally..
S
--
_
-- 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] receive faxes
On 4 May 2011, at 15:01, vip killa wrote: screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? There is so much wrong with that sentence, I don't know where to start. On 4 May 2011, at 16:02, vip killa wrote: Honestly Digium's Asterisk is not a quality project. Though it has lead the way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I refuse to pay Digium. Digium seems to make a bazillion dollars off of these flaws by selling commercial support/addons anyway... so that should be worth some bad karma points. Don't use Asterisk then. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Export Fax from Wave file
On 21 Apr 2011, at 13:46, A J Stiles wrote: You *might* be able to recover the document, *if and only if* the recording quality is high enough. Easiest way to try it is to call up a fax machine (either an actual real one, or a copy of Hylafax) from Asterisk and play the wav file down the line to it. But fax requires two-way negotiation right?.. S -- _ -- 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] Call Center Reporting
On 18 Apr 2011, at 11:06, bilal ghayyad wrote: I am using Asterisk for Call Center (so agents login, logout, ready, not ready, ... etc). To be able to have a good call center reporting, on what I have to depend? On the CDR of Asterisk or there is another way? Is there a good open source tool to be used for Asterisk call center reporting? http://www.google.com/search?q=asterisk+call+cener+reporting There are certainly some nice commercial ones. Can't comment on OSS stuff. But given it's for a call centre, I'd be tempted by the you don't get something for nothing approach. S -- _ -- 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] changing port 5060 to 5061
On 11 Apr 2011, at 10:03, darin iv wrote: please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.. Are you trying to be a pain in the arse? You've posted this far too many times, and ignored responses. If you ask a question and someone is kind enough to reply, it's polite to at least acknowledge what they suggested. S -- _ -- 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 odbc Length is .....
On 11 Apr 2011, at 15:28, vip killa wrote: I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see Length is 186545 or something similar but a different number in Asterisk CLI everytime someone leaves a message? Because not all messages are the same length. I'd guess it's length in bytes?.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP MESSAGE method support
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote: Is the following is the link for getting the source, http://svn.asterisk.org/svn/asterisk/trunk/ Please try not to reply to the entire digest.. S -- _ -- 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] Call recording - methodology
On 6 Apr 2011, at 11:54, Silver Thorne wrote: Does anyone know of any opensource or otherwise solutions out there that I can try out? Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy for that: http://www.voip-info.org/wiki/view/MixMonitor S -- _ -- 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 call back loop
On 6 Apr 2011, at 17:46, vip killa wrote: I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the externnotify again causing an infinite loop. Has anybody encountered this problem or is there an option to not have it run externnotify after checking messages? Look at the docs. Externnotify sends mailbox + mailbox count. Make your script exit if it's 0 messages. S-- _ -- 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 downloads.asterisk.org down?
On 31 Mar 2011, at 13:52, Sebastian wrote: http://www.downforeveryoneorjustme.com/downloads.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-users
Re: [asterisk-users] asterisk and fail2ban
On 28 Mar 2011, at 14:19, vip killa wrote: Yes I followed directions on that page Running Asterisk 1.6.1.22, anybody else experiencing this? How often does fail2ban check the logs? It can only block that often, so if more attempts happen in that time period it can't do anything until it knows. S -- _ -- 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 most stable version of asterisk?
On 24 Mar 2011, at 16:38, Gordon Henderson wrote: 1.2 has been the most stable version for me. Same setups with 1.4 +DAHDI has never been as stable with random crashes and re-starts - however they're not predictable and sometimes months apart. I had one instance of 1.2 run for over a year without a hiccup. I've got a 1.4 process that's been running 2 years, 6 weeks.. S -- _ -- 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] Fwd: asking for some help
On 24 Mar 2011, at 16:46, tahar .H wrote: so plz is there any one who can give me a puch to learn this extraordinary Asterisk plz(video things will be better :)) Learn to ask questions. Learn to read books. Learn to use google. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk using as a SIP client
On 23 Mar 2011, at 10:40, Nikhil wrote: I am planning to use asterisk as a IP phone(Porting asterisk into a hardware). Interesting.. Is there any limitations if I use asterisk as a SIP client?,and asterisk has any advantages if use like this? It's not really designed as a SIP client. It's probably possible, but it's very sledgehammer. S -- _ -- 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
