[asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)
testers needed -- Forwarded message -- Date: Wed, 11 Nov 2009 17:48:04 -0600 Subject: [Asterisk 0013405]: [patch] T38 gateway A NOTE has been added to this issue. == https://issues.asterisk.org/view.php?id=13405 == Reported By:dafe_von_cetin Assigned To: == Project:Asterisk Issue ID: 13405 Category: Applications/app_fax Reproducibility:N/A Severity: feature Priority: normal Status: confirmed Asterisk Version: SVN Regression: No Reviewboard Link: SVN Branch (only for SVN checkouts, not tarball releases): trunk SVN Revision (number only!): 140548 == Date Submitted: 2008-08-30 16:44 CDT Last Modified: 2009-11-11 17:47 CST == Summary:[patch] T38 gateway Description: Hi all, I'm sending you patch containing new application app_faxgateway.c (FaxGateway) which is able to mediate T30 to T38 and vice versa. Feature is using spands library (I used spandsp-0.0.4pre18 and spandsp-0.0.5pre4). Best regards Daniel. == -- (0113693) dafe_von_cetin (reporter) - 2009-11-11 17:47 https://issues.asterisk.org/view.php?id=13405#c113693 -- Hi, I've just uploaded the patch update for the newest trunk. The patch is still without previously mentioned transparency. Daniel. Issue History Date ModifiedUsername FieldChange == 2009-11-11 17:47 dafe_von_cetin Note Added: 0113693 == ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk keeps sending invite to sip phone No response to critical packet
Thanks Alex I suspected that no ack was being sent/received too. The invites are getting sent to the phone but nothing is coming back from the phone to the firewall. Does anybody know how I can sniff packets being sent and received to/from the phone and/or modem router the phone is connected to? I can't upload software to the modem/router or the phone (ie a packet sniffer). Can packets be sniffed from a linux box on the lan (please excuse my ignorance!)? If so can anybody point me to a resource that may help? Thanks for any help Marcus Asterisk is not receiving replies to the INVITE - probably due to NAT issues. marcus wells wrote: Hi there I am wondering if anybody can help me illuminate a problem I am having with my asterisk installation. I am using: - IP phone (Siemens gigaset S685IP) behind a modem/router that has ports udp 5060 and 1:10100 forwarded to the static ip of the IP phone (192.168.0.3). This has to go to: - modem that operates in half bridge mode (no nat) to a linux firewall (does natting ip is 192.168.0.20) that has the ports above forwarded to the staitc ip of the asterisk box (192.168.0.21 packaged version for ubuntu hardy). This phone works fine with a commercial provider of viop (via asterisk), but I can't get it to work with my install of asterisk in my remote network! ngrep-ing traffic on the firewall shows asterisk continually sending invites to the public ip of the ip phone. I would be very grateful for any pointers of where to start. If you need sip debug or ngrep info let me know and i'll reply with it. I've been beating my head against this for some time now! Thanks in advance Marcus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need Adapter/Gateway with PSTN-interface
I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the FXO-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! I know about the Digium-cards. Are there alternatives ? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows 7: Unclutter your desktop. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 SIP endless ringing...?
Anyone know what would cause an endless ringing situation? I have a snom360 and cisco 7970 (sip 8.5.3). I have an incoming trunk which dials both phones: [gp710] exten = _[*1-9].,1,Dial(SIP/li...@cisco7970SIP/li...@snom360,60) exten = _[*1-9].,n,Hangup If a call comes in, I can answer the call on the cisco no problem. However if I answer on the snom360, the cisco never stops ringing. -Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
On Thursday, November 12, 2009, jonas kellens wrote: Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! FWIW, I've had a few recommendations for the Linksys SPA3000. However, I haven't tried this for myself yet since I'm still in the planning stage of replacing my current Asterisk machine. In my case, I currently have a full-size tower and I'm planning to move to a mini-itx machine that doesn't have a PCI slot for my TDM400 card. HTH, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
I've read (through google) that the Linksys SPA-products do not have good voice quality on the PSTN-line. Grandstream HT486 is also just lifeline and EOL. The only I come up with is Patton-gateways but these are not at all cheap ! Jonas. On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote: On 12 Nov 2009, at 09:33, jonas kellens wrote: I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the FXO-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! I know about the Digium-cards. Are there alternatives ? Google could tell you this Try the Linksys/Sipura type products S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
On 12 Nov 2009, at 09:33, jonas kellens wrote: I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the FXO-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! I know about the Digium-cards. Are there alternatives ? Google could tell you this Try the Linksys/Sipura type products S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP source address error
Quoting Matt Riddell li...@venturevoip.com: [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1: Operation not permitted Are you binding to an address that the box doesn't own? Check the top of sip.conf. It's set to bind to 0.0.0.0, which IIRC is nothing strange. The question remains: how can a remote Asterisk server be receiving SIP packets that still contain the private net IP address of a client? Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scheduling destruction of SIP dialog
Hello, I got situation which is unclear for me, hope somebody could explain this. A calls to B INVITE sent from A to B B responds with 100 Trying B responds with 183 Progress After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in 32000 ms (Method: INVITE) Asterisk sends CANCEL _instantly_ B responds with 200 OK and 487 Request Terminated Asterisk confirms 102 ACK CLI: Really destroying SIP dialog '..' Method: INVITE Call terminates Asterisk version 1.4.18.1 Total call duration: 11s Timeout on call to B is set to 60 seconds: 'SIP/0277027277...@prov7|60|S(7197)' Call log is here: http://pastebin.ca/1667975 Why Asterisk decided to terminate the call? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Call Ring
Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesn't ring. This is because when the Dial command was executed, that extension was busy. Is there any way to make that extension ring as soon as its available if there is still an incoming call? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters
2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: 2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: Hello, 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a B410P board is to be disabled. There's currently no way to do that. It should be trivial to implment. The more difficult part of it would be how to define exactly what spans / channels to disable. But why do you need that? I don't really know why I thought I needed that feature but as some gateways implement this feature (the ability to enable or disable each port), I must have told myself I may have missed here. On a general point of view, as most Dahdi cards have a light showing nearby port status, it should ideally possible, to turn this light off when a port is disabled. But I must also add it doesn't seem very important to me to have this implemented. dahdi_genconf is an optional tool. Ideally it should need no configuration at all and generate configuration that Just Works (though the fact that it can do that indicates that the current defaults are broken). It should not be another configuration layer. If the configuration it has generated is not good enough, you can also manually edit it. Playing with comments (see bellow) doesn't help : file /etc/asterisk/dahdi-channels.conf is filled with 4 ports data. pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 TE # SPAN/4 TE Currently pri_termtype is the only directive in dahdi_genconf that uses this list syntax. I'm not very happy with it. I'm not exactly sure if there should be some sort of generic way of adding per-span (span? channel? how do you define a span?) definitions. Think of ssh_config. What about adding per-span section headers like Asterisk .conf files ? [span1] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [span2] group_lines 2 pri_termtype SPAN/2 TE This implies you will know span numbers in advance. I would like better ways to specify configuration. Really ? I used this [span1] header as an example. Using any other string would be fine for me as what matters, if I'm not mistaken, is the group_lines number : [foo] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [bar] group_lines 2 I don't think we need to define any further what a span is, beside that rules that applied to the whole genconf_parameters (no more than 1 group_lines statement) should apply to each section. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Termination Question
Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF with SPA941?
Appearently SPA941 is less than a SPA942 without two ports, poe and backlight. There is less features too, it doesn't support BLF. Is it possible to hack 942-software into 941, or is there another workaround? Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST_CONFIG, MEETME_INFO and meetme.conf
Hello, To make my dialplan more robust, I thought I wouldn't include any meetme-specific rules and I would exlusively rely on meetme.conf data. For each dialed number, I would check if this number is used as a conference room number in meetme.conf. When I'm trying to implement this, I can see that : 1. AST_CONFIG is not convenient to parse lines like conf=1234, as several lines are present and AST_CONFIG is dedicated to key-value pairs. 2. MEETME_INFO is focused on live conferences (though issue 15450 extends this behaviour). How would you work around this without using Realtime ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call Ring
- Original Message - From: Dan Journo To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 1:24 PM Subject: [asterisk-users] Incoming Call Ring Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesn't ring. This is because when the Dial command was executed, that extension was busy. Is there any way to make that extension ring as soon as its available if there is still an incoming call? You could put all 6 phones in a queue, and call that instead. But there will still be a delay before Asterisk calls the phone again. You could put the phones in a pickup-group, and the user could pick up the call, default is *8 Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] allowguest defaults to yes for SIP
In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call Ring
Depending on your phone, you can use CallWaitingRing to ring the phone anyway. I do this with Polycom 501's. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, November 12, 2009 6:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming Call Ring Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesn't ring. This is because when the Dial command was executed, that extension was busy. Is there any way to make that extension ring as soon as its available if there is still an incoming call? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful to know. Darryl Dunkin ddun...@netos.net wrote: Sorry to reply so late, I am months behind and catching up. I have been inspecting this on my own systems, and the results are inconsistent to say the least. I’ve been dumping these to the verbose logs for some time and monitoring them, but I have not been able to determine why the numbers are so far off. I am more concerned with the packets lost due to priority queuing within our network. Here is an example just today: ssrc=583450581 themssrc=1093951555 lp=0 rxjitter=0.003219 rxcount=1100 txjitter=0.000275 txcount=1108 rlp=57702 rtt=0.036000 If the txcount is only 1108, how can the remote lost packet count be 57702? Unless the call was nearly inaudible? I did verify with this end user, and the call was just fine. Is this an issue with the phone at the remote end misreporting? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, September 22, 2009 01:01 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] RTPAUDIOQOS Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval Alternatives: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec interface
Hi All, I need to interface a codec-type device to Asterisk. The device uses a TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock supplied by the device. I am about to start on a custom hardware design to interface this device to the computer, but thought I'd ask here before I get started on it. Does anyone know of a hardware interface that is already being manufactured that can tie a codec-based device into Asterisk? Thanks in advance, Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dell Poweredge T105
Hello, I someone successfully using Asterisk and Debian on an Opteron-enabled Dell Poweredge T105 ? If positive, which architecture (i386, amd, ...) w ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Poweredge T105
2009/11/12 Olivier oza-4...@myamail.com Hello, I someone successfully using Asterisk and Debian on an Opteron-enabled Dell Poweredge T105 ? If positive, which architecture (i386, amd, ...) w If positive, which architecture (i386, amd, ...) was chosen ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] my kernel is dazed and confused
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] state_interface backport issue
Any takers?
Still trying to get this resolved...
Thanks!
Robert Broyles wrote:
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I
have everything setup right.
I'm running 1.4.26.2 realtime.
queue_members:
`uniqueid` int(10) unsigned NOT NULL auto_increment,
`membername` varchar(40) default NULL,
`queue_name` varchar(128) default NULL,
`interface` varchar(128) default NULL,
`penalty` int(11) default NULL,
`paused` int(1) default NULL,
`state_interface` varchar(128) NOT NULL,
Data:
1, Name, QUEUENAME, Local/1...@agents/n, 1, , SIP/100
Local agents are setup setup in an 'agents' context.
[agents]
exten = 1050,1,Set(agentsip=${DB(agent_sip/1050)}
exten = 1050,2,Dial(SIP/${agentsip})
Queue shows the agent as unavailable when the SIP device (SIP/100) is
down. (as I would hope)... but shows the agent as available all the
other times.
As a result my CLi is on fire with 'busy' notices, because it's trying
to ring an agent even when they are on a call. If I remove the
state_interface, it shows them as 'busy' in the queue, and doesn't
ring them.
Let's see, what else did I forget? Other details:
sip.conf: limitonpeers=yes
and call-limit=5 on each SIP device
queue.conf: ringinuse=no
Anything else I should look for?
Thanks!
-Rob
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Re: [asterisk-users] allowguest defaults to yes for SIP
Lee Howard wrote: Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. I've always considered it to be good practice that something that may leave your system vulnerable, should be disabled by default. So yes, I would agree. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... -- Francesco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Lee Howard a écrit : In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Agree. Another possibility would be to have a guestcontext defined in default. This context would exist but empty. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Just my .02 - the guest context should torture or hangup instead of being empty. That might encourage a masochistic hacker though... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Thursday, November 12, 2009 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Lee Howard a écrit : In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Agree. Another possibility would be to have a guestcontext defined in default. This context would exist but empty. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
On 11/12/2009 09:42 AM, Francesco Peeters wrote: Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... I'm using Fedora 11, kernel 2.6.30.8-64.fc11.x86_64. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Am I correct in saying that the without allowguest=no anyone can connect and make calls through the default context? If allowguest is set to no, how can I ensure that incoming calls can still be received from our DDI supplier? Many Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 12 November 2009 14:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Just my .02 - the guest context should torture or hangup instead of being empty. That might encourage a masochistic hacker though... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Thursday, November 12, 2009 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Lee Howard a écrit : In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Agree. Another possibility would be to have a guestcontext defined in default. This context would exist but empty. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] POTS 4K linear codec
I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues. Today a cell phone or a POTS line phone can send DTMF clearly enough to operate a credit card or other interactive tone based system at the far end. With SIP it is sometimes chancy. Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF with SPA941?
Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Without the allowguest=no, Asterisk doesn't put up any defense against an unauthorized guest. You still have NAT/Firewall/IPTABLE defenses, for what they are worth. The trick is to get what you need without allowing what you don't want. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, November 12, 2009 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Am I correct in saying that the without allowguest=no anyone can connect and make calls through the default context? If allowguest is set to no, how can I ensure that incoming calls can still be received from our DDI supplier? Many Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 12 November 2009 14:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Just my .02 - the guest context should torture or hangup instead of being empty. That might encourage a masochistic hacker though... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Thursday, November 12, 2009 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Lee Howard a écrit : In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Agree. Another possibility would be to have a guestcontext defined in default. This context would exist but empty. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah To: Asterisk Users Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. -- Windows 7: Unclutter your desktop. Learn more. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On Thursday 12 November 2009 08:59:16 Danny Nicholas wrote: Without the allowguest=no, Asterisk doesn't put up any defense against an unauthorized guest. You still have NAT/Firewall/IPTABLE defenses, for what they are worth. The trick is to get what you need without allowing what you don't want. Don't assume that all guests are uninvited. The allowguest setting permits you to publish a SIP address at which new customers may make initial contact. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On Thursday 12 November 2009 09:00:45 Dan Journo wrote: Am I correct in saying that the without allowguest=no anyone can connect and make calls through the default context? If allowguest is set to no, how can I ensure that incoming calls can still be received from our DDI supplier? You're correct in stating that this is the purpose of the allowguest configuration option. If you disable it, only peers with which you have established settings will be able to call into your system. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On Thursday 12 November 2009 07:47:34 Lee Howard wrote: In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Actually, they only have access to your default context. Whether you make available outgoing calls in your default context is your choice. By default, there is no capability of making outgoing calls from your default context. Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. The purpose of the allowguest option is to allow persons to call into your system from a zero-knowledge position. This allows you to publish a general SIP address as a point of contact. The reason why it is set that way in the sample configuration is to make it easy for new users to get to that magic moment when Asterisk first responds to their call (in essence, to get the user hooked). It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Correct, you should be using something like internal instead. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POTS 4K linear codec
On Thu, 2009-11-12 at 08:53 -0600, Cary Fitch wrote: Digital 64K telco sounds very good as a phone conversation. Digital 64k audio coming across a T1 is essentially identical to the ulaw codec in VoIP. Digital 64k audio coming across an E1 is essentially identical to the alaw codec. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
Hello, I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs . Can anyone tell me what I'm doing wrong? Thanks, Andreas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] solution for NAT issues?
Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward port 49923 to port of the IP phone, maybe nat mapping has expired or something. I have tried STUN, still sometimes the phones just cannot be reached. is there any other software to manage binding of ports on specific users so that the routers always keeps the port mapped to port of the ip phone . TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote: On Thursday 12 November 2009 07:47:34 Lee Howard wrote: In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Actually, they only have access to your default context. Whether you make available outgoing calls in your default context is your choice. By default, there is no capability of making outgoing calls from your default context. Well, yes, the default configuration is useless. But, let's say I follow doc/security.txt exactly and have this: [default] exten = 6123,Dial(Zap/1) ... therefore, by default, an unauthenticated user from anywhere can call the extension Zap/1. It's not my point whether or not this poses a financial risk. My point is that this is an insecure default behavior to have allowguest=yes. Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. The purpose of the allowguest option is to allow persons to call into your system from a zero-knowledge position. This allows you to publish a general SIP address as a point of contact. These people should need to deliberately use allowguest=yes. I would venture to guess that these people already know who they are and deliberately have this set. I would venture to guess that there are far, far more people who have it turned on by default who really don't want it that way than there are who expected it to be that way and desire it to so be. The reason why it is set that way in the sample configuration is to make it easy for new users to get to that magic moment when Asterisk first responds to their call (in essence, to get the user hooked). This is a poor excuse for a poor default security setting. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Correct, you should be using something like internal instead. And yet this point is not even made clear in the doc/security.txt file. It says to not use default for anything you don't want to get abused, but it doesn't say *why*. So I can envision, then, someone reading the document and then changing context=internal in the [general] section of sip.conf... and thinking that they responded correctly to what the document said. If this default is to persist then I think that it behooves the developers to at least make this exposure clear to the users. Therefore, the in the [general] section of sip.conf the context should not be set to default, but rather to unauthorized or public or open or free or something that makes it clear that this is where unauthenticated SIP calls go. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
On 12 Nov 2009, at 15:38, Cyprus VoIP wrote: I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs Can anyone tell me what I'm doing wrong? You are not posting on the FreePBX forums? ;) The solution however, is to install the modules using the module admin. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
Karl Fife wrote: Perhaps there's an arcane way to query lipbri the older releases from the CLI? Can anyone speak to that? Quick and dirty: strings /usr/lib/libpri.so That's CLI, tho' not the one you're talking about. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
OK. Thanks On Thu, Nov 12, 2009 at 4:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.comwrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Windows 7: Unclutter your desktop. Learn more.http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs Can anyone tell me what I'm doing wrong? You are not posting on the FreePBX forums? ;) I figured Asterisk-Users would know ;) The solution however, is to install the modules using the module admin. The problem is that the online module update is not working for me (Cannot connect to online repository (mirror.freepbx.org). Online modules are not available.) and I couldn't find online a working solution :-( Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
On 12 Nov 2009, at 16:29, Cyprus VoIP wrote: The problem is that the online module update is not working for me (Cannot connect to online repository (mirror.freepbx.org). Online modules are not available.) and I couldn't find online a working solution :-( DNS/Gateway ok on server? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't connect to voip provider over NAT
On Thu, 2009-11-12 at 14:50 +1100, Michael Wyres wrote: Have you tried nat=yes in the definition in sip.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Thursday, 12 November 2009 13:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't connect to voip provider over NAT Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=theprovider's server username=username secret=password port=5060 canreinvite=YES dtmfmode=rfc2833 I've tried opening all ports to test this but, still doesn't work. Now, I need to know which especific ports to open in order to allow sip flow correctly. Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=1 rtpend=2 Don't know what else to try. Please help. Thanks in advanced for your help. I think this is more a problem that you are not setting your external IP address correctly so the provider can send RTP back to you. Make sure you have either externip, externhost or stunaddr(1.6) set correctly. The do a sip show settings in the CLI to see if the correct address is set. If you are behind nat canreinvite should be set to no. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
Dear Steve, Do you have your DNS settings ok? Otherwise include these settings(DNS1 DNS2) in your network configuration. Regards, Nelson Granados -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP Sent: Thursday, November 12, 2009 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1 I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs Can anyone tell me what I'm doing wrong? You are not posting on the FreePBX forums? ;) I figured Asterisk-Users would know ;) The solution however, is to install the modules using the module admin. The problem is that the online module update is not working for me (Cannot connect to online repository (mirror.freepbx.org). Online modules are not available.) and I couldn't find online a working solution :-( Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
The problem is that the online module update is not working for me (Cannot connect to online repository (mirror.freepbx.org). Online modules are not available.) and I couldn't find online a working solution :-( DNS/Gateway ok on server? Yes. The problem is with the FreePBX modules. I forced the mirror file to include version 2.5, and I get a list, but when I try to install the modules, it says that the modules need FreePBX version 2.5.0alpha or rc1 or higher, but although 2.5.2 is indeed higher, it's rejected. I've given up on this software and will continue to edit my .conf files manually. what a waste of time :-( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
On 12 Nov 2009, at 16:54, Nelson Granados wrote: Dear Steve, Do you have your DNS settings ok? Yes, but its not me with the problem. ;) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?
Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using toneduration I can control duration of tone per digit. But I can't find a way to do both at the same time Application sendDTMF simply ignores the value set in toneduration and sends DTMF at some default value, which I don't know what it is, but it is obviously not 50ms because the hardware can't reliably recognized the digits. Is there a way I can send digits with 50ms tone duration and 50ms gap between them? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On Thursday 12 November 2009 09:53:17 Lee Howard wrote: Tilghman Lesher wrote: On Thursday 12 November 2009 07:47:34 Lee Howard wrote: In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Actually, they only have access to your default context. Whether you make available outgoing calls in your default context is your choice. By default, there is no capability of making outgoing calls from your default context. Well, yes, the default configuration is useless. But, let's say I follow doc/security.txt exactly and have this: [default] exten = 6123,Dial(Zap/1) ... therefore, by default, an unauthenticated user from anywhere can call the extension Zap/1. It's not my point whether or not this poses a financial risk. My point is that this is an insecure default behavior to have allowguest=yes. Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. The purpose of the allowguest option is to allow persons to call into your system from a zero-knowledge position. This allows you to publish a general SIP address as a point of contact. These people should need to deliberately use allowguest=yes. I would venture to guess that these people already know who they are and deliberately have this set. I would venture to guess that there are far, far more people who have it turned on by default who really don't want it that way than there are who expected it to be that way and desire it to so be. And the people who use this probably believe that YOU should be the one who has to deliberately turn this option off. I would venture to guess that 90% of all statistics are made up on the spot, including this one and the two you specified above. The reason why it is set that way in the sample configuration is to make it easy for new users to get to that magic moment when Asterisk first responds to their call (in essence, to get the user hooked). This is a poor excuse for a poor default security setting. It's not a security setting; it's a functionality setting. You see it behind rose-tinted spectacles because in your specific case, you don't have a use for it. That's fine, but please do not extrapolate from your limited use cases what the global settings should be. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Correct, you should be using something like internal instead. And yet this point is not even made clear in the doc/security.txt file. It says to not use default for anything you don't want to get abused, but it doesn't say *why*. So I can envision, then, someone reading the document and then changing context=internal in the [general] section of sip.conf... and thinking that they responded correctly to what the document said. You've just made a case for enhancing the documentation, not for changing the defaults. If you contribute documentation changes to this effect on the issue tracker, I would be more than happy to commit them. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote:
On Thursday 12 November 2009 09:53:17 Lee Howard wrote:
These people should need to deliberately use allowguest=yes. I would
venture to guess that these people already know who they are and
deliberately have this set. I would venture to guess that there are
far, far more people who have it turned on by default who really don't
want it that way than there are who expected it to be that way and
desire it to so be.
And the people who use this probably believe that YOU should be the one
who has to deliberately turn this option off. I would venture to guess that
90% of all statistics are made up on the spot, including this one and the
two you specified above.
I made it clear that they were guesses. But, please *DO* take a vote on
this. I'm not seeing anyone but you stand up to support the default
setting. Unless you take a vote there's really nothing I can do but guess.
The fact that this problem is being exploited leads me to believe that
this is far-more prevalent a problem than just my single case. Take
care of your users when you can do something so easily. Don't
deliberately let them learn things the hard way on the basis that they
should have known better. The mere fact that this issue is addressed in
doc/security.txt should be an indication that there is a common risk
that could be averted.
And yet this point is not even made clear in the doc/security.txt file.
It says to not use default for anything you don't want to get abused,
but it doesn't say *why*. So I can envision, then, someone reading the
document and then changing context=internal in the [general] section of
sip.conf... and thinking that they responded correctly to what the
document said.
You've just made a case for enhancing the documentation, not for changing
the defaults. If you contribute documentation changes to this effect on the
issue tracker, I would be more than happy to commit them.
The patch is attached. Feel free to add it to bug tracker issue ID
16226 which some maintainer was happy enough to close already.
And, for what it's worth let me restate my vote that the default for
allowguest be changed to no on the basis of keeping ignorant people
from making a stupid mistake.
Thanks,
Lee.
--- asterisk-1.4.21.2/doc/security.txt.old 2009-11-12 09:53:03.0 -0800
+++ asterisk-1.4.21.2/doc/security.txt 2009-11-12 09:56:38.0 -0800
@@ -48,12 +48,15 @@
Therefore, you should NOT allow access to outgoing or toll services in
contexts that are accessible (especially without a password) from incoming
-channels, be they IAX channels, FX or other trunks, or even untrusted
-stations within you network. In particular, never ever put outgoing toll
-services in the default context. To make things easier, you can include
-the default context within other private contexts by using:
+channels, be they IAX channels, SIP channels, FX or other trunks, or even
+untrusted stations within you network. Keep in mind that the default setting
+for SIP configuration is allowguest=yes. So unauthenticated SIP users will,
+by default, be able to access the context specified in the [general] section.
+Therefore, never ever put outgoing toll services in the public context.
+To make things easier, you can include the default context within other
+private contexts by using:
- include = default
+ include = public
in the appropriate section. A well designed PBX might look like this:
@@ -63,9 +66,9 @@
[local]
exten = _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
-include = default
+include = public
-[default]
+[public]
exten = 6123,Dial(Zap/1)
--- asterisk-1.4.21.2/configs/sip.conf.sample.old 2009-11-12 09:57:19.0 -0800
+++ asterisk-1.4.21.2/configs/sip.conf.sample 2009-11-12 09:58:41.0 -0800
@@ -24,7 +24,7 @@
;
[general]
-context=default ; Default context for incoming calls
+context=public ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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Re: [asterisk-users] allowguest defaults to yes for SIP
On Thursday 12 November 2009 12:08:39 Lee Howard wrote: Tilghman Lesher wrote: On Thursday 12 November 2009 09:53:17 Lee Howard wrote: And yet this point is not even made clear in the doc/security.txt file. It says to not use default for anything you don't want to get abused, but it doesn't say *why*. So I can envision, then, someone reading the document and then changing context=internal in the [general] section of sip.conf... and thinking that they responded correctly to what the document said. You've just made a case for enhancing the documentation, not for changing the defaults. If you contribute documentation changes to this effect on the issue tracker, I would be more than happy to commit them. The patch is attached. Feel free to add it to bug tracker issue ID 16226 which some maintainer was happy enough to close already. The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. If you want to create a new issue and attach your patch there, I'll look at it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote: The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. Sigh... of course. It's a gentlemen's club and only members have a say. If you want to create a new issue and attach your patch there, I'll look at it. I sent a patch. I pointed you at a case. That should have been FAR more than enough for my attempt at contribution to be acceptable. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
On 12 Nov 2009, at 17:09, Cyprus VoIP wrote DNS/Gateway ok on server? Yes. The problem is with the FreePBX modules. I forced the mirror file to include version 2.5, and I get a list, but when I try to install the modules, it says that the modules need FreePBX version 2.5.0alpha or rc1 or higher, but although 2.5.2 is indeed higher, it's rejected. I've given up on this software and will continue to edit my .conf files manually. what a waste of time :-( Well, its clearly not an Asterisk issue, so yes it is a waste of time :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP source address error
It's set to bind to 0.0.0.0, which IIRC is nothing strange. The question remains: how can a remote Asterisk server be receiving SIP packets that still contain the private net IP address of a client? It sounds to me as if the client hasn't been told to use its gateway's public IP address in the SIP conversation, and as if the client isn't sending its outbound packets through a gateway/NAT which is SIP-aware and can rewrite the SIP data accordingly. There are several approaches which can work: - The gateway is properly configured to forward its external ports to the client, and the client is manually configured to use the gateway's external IP address in its SIP protocol exchanges. - The gateway does port forwarding and NAT properly, and is also SIP-aware - it intercepts and rewrites the contents of the outbound SIP packets, changing the IP address and port given by the client to its own IP address and whatever external port it has NAT'ed / redirected to the client. - The gateway does port forwarding and NAT properly, and the client is configured to use STUN to figure out what public IP address/port its packets are being NAT'ed to. - The client doesn't talk directly to the outside peers, but goes through a SIP proxy running on the gateway. In your case, it sounds as if the client and gateway aren't doing one of these things. As a result, the client's SIP protocol packets still contain its private-net IP and port, at the time they reach the remote server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Thursday, November 12, 2009 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Tilghman Lesher wrote: The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. Sigh... of course. It's a gentlemen's club and only members have a say. If you want to create a new issue and attach your patch there, I'll look at it. I sent a patch. I pointed you at a case. That should have been FAR more than enough for my attempt at contribution to be acceptable. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POTS 4K linear codec
On Thu, 12 Nov 2009, Cary Fitch wrote: I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues. Today a cell phone or a POTS line phone can send DTMF clearly enough to operate a credit card or other interactive tone based system at the far end. With SIP it is sometimes chancy. Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. Cary Fitch Isn't that ulaw/alaw? j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. You're suggesting that if I pay someone they'll be able to get the default setting for allowguest changed to no ? I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. So either I am stupid or lazy or there is a risk here that can catch even others off-guard. I've been down this contribution road-path a half-dozen times before with Asterisk. So forgive me if I don't play it out to the final futile note. In ESR's CatB there's the idea where the maintainer encourages (and wants) bug reporting, feedback, and other non-code forms of contribution (as well as code contributions). He refers to it as grooming co-developers. That's not how Asterisk development works... here you can contribute if you're already in the meritocracy, but if you're not, then you have more than a difficult time in trying to even contribute in small non-monetary ways. So anyway, I've been down this road a half-dozen times already, and it ends up being futile, frustrating, and time-consuming. I'm too busy today to be interested in playing. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters
On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: 2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: What about adding per-span section headers like Asterisk .conf files ? [span1] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [span2] group_lines 2 pri_termtype SPAN/2 TE This implies you will know span numbers in advance. I would like better ways to specify configuration. Really ? I used this [span1] header as an example. Using any other string would be fine for me as what matters, if I'm not mistaken, is the group_lines number : [foo] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [bar] group_lines 2 How can you tell which spans / channels will use each section? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Eh... if VoIP fraud weren't so rampant, and I didn't constantly see mailings to the Asterisk list about How do I secure my system from the people who've been costing me tons of money lately, I would say that having a lax stance on security in exchange for additional usability might be a good thing. But as is, that's simply not the case. The 'usability' you get from this is really only questionably essential in its ability to save time, but the security one would get from a change could save some people actual money -- not just time. As someone who used to design systems and networks, I would vote for security over nebulous desire to keep the status quo. True, you can't keep stupid people from doing stupid things, but given a choice between protecting the ignorant from a bad situation or catering to those who want to avoid an extra step or two on installation, I'd side with protecting the ignorant every time. There's always a trade-off between usability and security, and I'm of the opinion that security is the more important of the two when dealing with systems connected to the Internet. Call me a cynic. :) N. Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Thursday, November 12, 2009 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Tilghman Lesher wrote: The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. Sigh... of course. It's a gentlemen's club and only members have a say. If you want to create a new issue and attach your patch there, I'll look at it. I sent a patch. I pointed you at a case. That should have been FAR more than enough for my attempt at contribution to be acceptable. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
I did not mean to state or imply that you are lazy or stupid; It's just that some folks expect to spend 10 minutes reading a PDF, set up Asterisk and all is well - That's not what Open Source is about. If you want limited or no risk, you have to pay the piper. I'll bet there are thousands of pieces of code that are great that don't get through the contribution process. You can't have any type of *cracy without crazy :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Thursday, November 12, 2009 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. You're suggesting that if I pay someone they'll be able to get the default setting for allowguest changed to no ? I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. So either I am stupid or lazy or there is a risk here that can catch even others off-guard. I've been down this contribution road-path a half-dozen times before with Asterisk. So forgive me if I don't play it out to the final futile note. In ESR's CatB there's the idea where the maintainer encourages (and wants) bug reporting, feedback, and other non-code forms of contribution (as well as code contributions). He refers to it as grooming co-developers. That's not how Asterisk development works... here you can contribute if you're already in the meritocracy, but if you're not, then you have more than a difficult time in trying to even contribute in small non-monetary ways. So anyway, I've been down this road a half-dozen times already, and it ends up being futile, frustrating, and time-consuming. I'm too busy today to be interested in playing. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
On Thu, Nov 12, 2009 at 09:31:11AM -0500, Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue NMI - Non Maskable Interrupt. This is a rather generic error message. Search a bit to see how to make some more sense of the messages following it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
At 02:25 AM 11/12/2009, you wrote: FWIW, I've had a few recommendations for the Linksys SPA3000. However, I haven't tried this for myself yet since I'm still in the planning stage of replacing my current Asterisk machine. In my case, I currently have a full-size tower and I'm planning to move to a mini-itx machine that doesn't have a PCI slot for my TDM400 card. I was able to assemble a MiniITX box with a laptop HD and Atom 330 that had room for my TDM400, so it's possible if you want. I've not seen one assembled that would work, but I got all the parts I needed from NewEgg. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec interface
On Thu, Nov 12, 2009 at 09:22:41AM -0500, Bill Shaw wrote: Hi All, I need to interface a codec-type device to Asterisk. The device uses a TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock supplied by the device. I am about to start on a custom hardware design to interface this device to the computer, but thought I'd ask here before I get started on it. Does anyone know of a hardware interface that is already being manufactured that can tie a codec-based device into Asterisk? There's codec_dahdi , that implements g729 and g723 through a specific Digium card. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 11:16, Thu 12 Nov 09, Lee Howard wrote: Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. You're suggesting that if I pay someone they'll be able to get the default setting for allowguest changed to no ? No, he was saying that if you dont know the system you are going to setup, and dont have the time/resources to read up on how it works, you can always hire someone who knows how stuff works. I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. No-one told you you are stupid or lazy. It's just that this option only allows unwanted stuff if the configuration is made to do that. So either I am stupid or lazy or there is a risk here that can catch even others off-guard. I've been down this contribution road-path a half-dozen times before with Asterisk. So forgive me if I don't play it out to the final futile note. In ESR's CatB there's the idea where the maintainer encourages (and wants) bug reporting, feedback, and other non-code forms of contribution (as well as code contributions). He refers to it as grooming co-developers. That's not how Asterisk development works... here you can contribute if you're already in the meritocracy, but if you're not, then you have more than a difficult time in trying to even contribute in small non-monetary ways. This is so untrue. When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. Sometimes it takes some time before the first review of your patch is happening. This is mainly because the developers are really busy, and only part of the developers is being paid to do this stuff for asterisk, all the others are doing it in their free time. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Michiel van Baak wrote: When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. I'm sincerely pleased to know that you've had a different experience than have I. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. I truly wish it were. I've seen more than a few that didn't. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wcse...@selbytech.com wrote: The 7960 and 79x2 use different sip firmwares and as far a I have seen the 7960 does not have the same port issue the 7941/2 seems to have (which technically is not a problem, just an implementation of the sip protocol that you don't typically see). As to your issue, are you still seeing the same error messages in the ssh logs? I haven't ever had to use the register with proxy settings in my configs, but I've only worked with the 79x1 series phones, not the x2. I've actually got a post up on my blog addressing setting up a 7941 in a situation similar to yours: http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/ In that post is a sanitized version of my conf file that I use on my own deskphone, if you'd like to download it and try it out with your setup. My config is very similar though my only question is you have registerWithProxy set to true though nothing defined. Was this a sanitation mistake? sipProxies backupProxy/ backupProxyPort/ emergencyProxy/ emergencyProxyPort/ outboundProxy/ outboundProxyPort/ registerWithProxytrue/registerWithProxy /sipProxies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Request for Review: Building Queues with Asterisk
I have been working on some documentation for how to build queues for Asterisk. This is an introduction for getting device state working for queues, and building queues. It contains the documentation file (text format) and also has the .tar.gz file of the /etc/asterisk/ directory I was using for testing. The modules.conf file has autoload=no enabled, and just loads the modules that were required for the example (along with probably a couple extra modules, but the list of modules has been toned down). Please review and let me know how it goes for you! Thanks! Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for Review: Building Queues with Asterisk
Leif Madsen wrote: Please review and let me know how it goes for you! Where is it? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
Just checked with my actual config file, and it's not a sanitation mistake, that's how I've actually got mine setup. Like I said earlier, I've never even messed with that section of my config before...I set mine up based on a combination of configs I've found around the net (I think you've already linked to them in another post to the list). Thanks, --Warren Selby On Thu, Nov 12, 2009 at 4:11 PM, Stephen Reese rsre...@gmail.com wrote: On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wcse...@selbytech.com wrote: The 7960 and 79x2 use different sip firmwares and as far a I have seen the 7960 does not have the same port issue the 7941/2 seems to have (which technically is not a problem, just an implementation of the sip protocol that you don't typically see). As to your issue, are you still seeing the same error messages in the ssh logs? I haven't ever had to use the register with proxy settings in my configs, but I've only worked with the 79x1 series phones, not the x2. I've actually got a post up on my blog addressing setting up a 7941 in a situation similar to yours: http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/ In that post is a sanitized version of my conf file that I use on my own deskphone, if you'd like to download it and try it out with your setup. My config is very similar though my only question is you have registerWithProxy set to true though nothing defined. Was this a sanitation mistake? sipProxies backupProxy/ backupProxyPort/ emergencyProxy/ emergencyProxyPort/ outboundProxy/ outboundProxyPort/ registerWithProxytrue/registerWithProxy /sipProxies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 3:59 AM, Danny Nicholas wrote: Without the allowguest=no, Asterisk doesn't put up any defense against an unauthorized guest. You still have NAT/Firewall/IPTABLE defenses, for what they are worth. The trick is to get what you need without allowing what you don't want. A slight clarification - I wouldn't say it's defences. By default these calls are sent to the default context (which should not have the capability to make calls other than test the system). So, yes you are allowing unauthenticated calls, but to the echo test etc. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for Review: Building Queues with Asterisk
On 17:19, Thu 12 Nov 09, Leif Madsen wrote: I have been working on some documentation for how to build queues for Asterisk. This is an introduction for getting device state working for queues, and building queues. It contains the documentation file (text format) and also has the .tar.gz file of the /etc/asterisk/ directory I was using for testing. The modules.conf file has autoload=no enabled, and just loads the modules that were required for the example (along with probably a couple extra modules, but the list of modules has been toned down). Please review and let me know how it goes for you! Where can we find all of this ? -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 8:30 AM, SIP wrote: Eh... if VoIP fraud weren't so rampant, and I didn't constantly see mailings to the Asterisk list about How do I secure my system from the people who've been costing me tons of money lately, I would say that having a lax stance on security in exchange for additional usability might be a good thing. But as is, that's simply not the case. The 'usability' you get from this is really only questionably essential in its ability to save time, but the security one would get from a change could save some people actual money -- not just time. The problem there is normally lax usernames and passwords. Not that there is default access to the echo test. As someone who used to design systems and networks, I would vote for security over nebulous desire to keep the status quo. Because you're already using Asterisk. If it had been too hard at the start maybe you wouldn't. True, you can't keep stupid people from doing stupid things, but given a choice between protecting the ignorant from a bad situation or catering to those who want to avoid an extra step or two on installation, I'd side with protecting the ignorant every time. There's always a trade-off between usability and security, and I'm of the opinion that security is the more important of the two when dealing with systems connected to the Internet. Call me a cynic. :) The ignorant won't have changed the default context - they likely won't even know how to edit a config file - so they're safe. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 9:37 AM, Lee Howard wrote: Michiel van Baak wrote: When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. I'm sincerely pleased to know that you've had a different experience than have I. I've had an experience which is a little of both. I've had some patches accepted, and other not accepted (MySQL userfield2-5). I think it's really important that not every patch gets accepted, and I really like the discussion which has taken place on this one. Basically the two sides of the argument are: For: I put stuff in my default context, now people can use it without authentication - I didn't expect this. Against: I'm a new user, I tried to get Asterisk working but had authentication problems, now I'm moving to Microsoft OCS (or 3cx or whatever). I kinda think that you want to make it as easy as possible for new users to at least run an echo test (and maybe make a call through to Digium). Once they've done that they're going to need to edit config files. If there is strong wording in the config files explaining that they shouldn't be adding anything here without first reading the security document I think it would suffice. Maybe the best way would be to make it that the default context only provides the info from the examples unless you provide an option: read_security_document=yes or whatever. I know that it seems really easy for most of us to chuck a couple of sip devices into the config and set up some extensions, but for a new user, any step at all they need to make before getting a call working is bad. The average new user won't know much about VoIP, nor much (if anything) about Linux, and seeing some text interface provide some random error when they try it for the first time will just turn them away. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. I truly wish it were. I've seen more than a few that didn't. :) just consider yourself lucky it's not glibc or something you're trying to commit to :) The people with commit access tend to just say no. Even if the change stops something from breaking on multiple platforms (see eglibc discussion). Basically to get a change into Asterisk, you need a reasonably good percentage of people agreeing that the change is worthwhile (and the best way to implement it). Don't get me wrong, I understand the change you're proposing, just that it may not be the 100% best way to do it, and it needs to be carefully thought out before proceeding with something which may have a large impact on new users. Think what it's like for the 3G video people who have a huge patchset that they wrote before bringing it up for discussion only to hear it was the wrong way to do it. At least the patch is small :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Can't connect to voip provider over NAT
Have you tried nat=yes in the definition in sip.conf? Yes, I have that definition in sip.conf. Now, I'm getting the following error -- SIP/voipprovider-094132d8 is making progress passing it to SIP/102-09423d58 -- Got SIP response 603 Declined back from 208.xx.xx.xx -- SIP/voipprovider-094132d8 is busy == Everyone is busy/congested at this time (1:1/0/0) and I get a This account number is not valid on the headset. I've called my provider and they've said that everything is fine at their end. I don't know why I'm getting the message saying the account is not valid. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Home line noise problem
I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.
Yeah, but would you automate an install with additional contents in the
default context?
--
Cheers,
Matt Riddell
Director
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[asterisk-users] TDM400p , asteriskNow and may other woes.....
Hello all, I am new to asterisk and have spent a good 4 or 5 days trying to get things sorted out. I initially installed it in Fedora Core 11 and compiled mods + asterisk. After much problems, I went with asteriskNow. The biggest problem I am have is getting some kind of base configuration going. I have been all over Google, but what I oftwen find is conflicting or outdated information. Commands to use that no longer work because things have changed from zaptel to dahdi. I have a TDM400P with 1 FXO module and 1 FXS module installed. The card is readily seen. I cannot get a dial tone (and I did plug in the power). And am unsure what to do. 1. /etc/init.d/asterisk does not exist, so I have no idea how the system is even starting. 2. The /etc/asterisk folder has zapata.conf.template AND chan_dahdi.conf.template. Which one am I supposed to use? Anyone want to take some pity? Just looking to get to the point of a dial tone. At least will know things are working. I can go on from there. But at this point I am stuck. Not trying to take the lazy way out, just trying to get a handle on this. BELIEVE ME I have put forth a GREAT deal of effort. Went to the irc channel and though there were some 200 users, most were prob just idleing. Been all over the forums. And every google setup tdm400p asterisk page that exists. Just want to plug in regular telephone, and dial out through my telephone company. Thanks dmesg Nov 8 00:01:37 localhost kernel: Module 0: Installed -- AUTO FXS/DPO Nov 8 00:01:37 localhost kernel: Module 1: Not installed Nov 8 00:01:37 localhost kernel: Module 2: Not installed Nov 8 00:01:37 localhost kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 8 00:01:37 localhost kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules) Nov 8 00:01:37 localhost kernel: dahdi_transcode: Loaded. -- dahdi_cfg -vv -- DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 2 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 4 to mg2 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question ...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah mailto:tareksa...@hotmail.com To: Asterisk Users mailto:asterisk-users@lists.digium.com Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. Learn more. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-U S:WWL_WIN_evergreen:112009 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its own sip account. Martin - Original Message - From: jonas kellens To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 5:38 AM Subject: Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface I've read (through google) that the Linksys SPA-products do not have good voice quality on the PSTN-line. Grandstream HT486 is also just lifeline and EOL. The only I come up with is Patton-gateways but these are not at all cheap ! Jonas. On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote: On 12 Nov 2009, at 09:33, jonas kellens wrote: I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the FXO-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! I know about the Digium-cards. Are there alternatives ? Google could tell you this Try the Linksys/Sipura type products S -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Friday, 13 November 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. This has nothing to do with stupidity or laziness. The way I see it, the reason you have encountered some resistance to your opinion in regards to whether guest access should be allowed by default or should not be, is not because your opinion is right or wrong - everyone is entitled to an opinion - and your stance has merit, certainly - I don't think anyone is actually disputing that. It is more that a lot of the people on this list have been using Asterisk for a LNG time, and have explained why it might be advantageous to have guest access enabled by default. There are definitely uses for this functionality, as has been demonstrated by a number of examples contained in this thread. Isn't this why you joined the list? To learn more about the product, and get ideas and assistance from the more experienced users of the product? You raised your concern, and Tilghman (a senior developer at Digium) explained the reasoning behind the default setting. He suggested that you take your concern to the tracker and post a patch. You resisted. The open source community (despite what some think) is a highly organised community, with structures in place to get things like that done. If you consistently did end runs around established corporate procedures in your workplace, you'd expect a foot up the ass from management. Tilghman was as politely as possible asking you to follow the established procedures. You chose to resist. Now, the default extensions.conf contains the following snippet: snip [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include = demo /snip Now, a lot of people never RTFM for anything. Moreover, how many people actually read the EULA for any piece of software they use? It's not Asterisk/Digium's fault if people don't read the available documentation that they provide. The quite plainly clear statement above is in a production system, you probably don't want to have the demo there. Did you read that bit? Did you wonder why that bit is there? When I first started working with Asterisk, I clearly remember that line (or something very similar) piquing my curiousity to dig a little deeper as to why that statement was made. Lo, I discovered that this was because by default, guest access is allowed. Digium has made that available in the distribution for EVERYONE to read, and extensions.conf is probably the most accessed file in an Asterisk system not using RealTime, so people who choose to ignore reading the excellent notes and annotations in all of the default configuration files is doing themselves a disservice. I too found the default access odd at first, but I chose to understand the reasoning from people who knew better, instead of chucking a hissy fit. IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
I have no first-hand experience with the fussy idiosyncrasies, but the BIG PICTURE is to have server A set up the call, and then reinvite the media directly from B to C. The call control messages flow to server A, the media goes directly. If you don't have NAT traversal Kung-Fu, I suggest using IAX2 over SIP. -K - Original Message - From: B.Masoud @ SH To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 12, 2009 6:10 PM Subject: Re: [asterisk-users] Termination Question So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question ...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah To: Asterisk Users Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. Windows 7: Unclutter your desktop. Learn more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
That could work, but I have no control over server B, not server C ! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Friday, November 13, 2009 3:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question I have no first-hand experience with the fussy idiosyncrasies, but the BIG PICTURE is to have server A set up the call, and then reinvite the media directly from B to C. The call control messages flow to server A, the media goes directly. If you don't have NAT traversal Kung-Fu, I suggest using IAX2 over SIP. -K - Original Message - From: B.Masoud @ SH mailto:i...@saudihome.com To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Thursday, November 12, 2009 6:10 PM Subject: Re: [asterisk-users] Termination Question So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question ...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah mailto:tareksa...@hotmail.com To: Asterisk Users mailto:asterisk-users@lists.digium.com Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. Learn more. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-U S:WWL_WIN_evergreen:112009 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
I add this line in our in/out contexts:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging on).
I'm sure you could output it anwhere else as well with a system call/echo.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful
to know.
Darryl Dunkin ddun...@netos.net wrote:
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are
inconsistent to say the least. I’ve been dumping these to the verbose logs
for some time and monitoring them, but I have not been able to determine why
the numbers are so far off. I am more concerned with the packets lost due to
priority queuing within our network.
Here is an example just today:
ssrc=583450581
themssrc=1093951555
lp=0
rxjitter=0.003219
rxcount=1100
txjitter=0.000275
txcount=1108
rlp=57702
rtt=0.036000
If the txcount is only 1108, how can the remote lost packet count be 57702?
Unless the call was nearly inaudible?
I did verify with this end user, and the call was just fine. Is this an issue
with the phone at the remote end misreporting?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys
Sent: Tuesday, September 22, 2009 01:01
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: 2009 m. rugsėjo 22 d. 09:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTPAUDIOQOS
hey all,
can any body know what this parameter stands for
i got RTPAUDIOQOS while i have SIP channels
but could not understand then what this parameter tell
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
if any one know plese help me to or give any documentation link
regards
Dhaval
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Re: [asterisk-users] Request for Review: Building Queues with Asterisk
Barry L. Kline wrote: Leif Madsen wrote: Please review and let me know how it goes for you! Where is it? Ah yes, in my eagerness to get ready for dinner with the g/fs parents, I have forgotten to post where this exists :) I posted it to the issue tracker here: https://issues.asterisk.org/view.php?id=16237 Enjoy! Leif. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need opinion about GSM codec for Internet
If you doesn't need transcoding, you doesn't need any licenses... Martin - Original Message - From: Vinícius Fontes vinic...@canall.com.br To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 06, 2009 11:43 AM Subject: Re: [asterisk-users] Need opinion about GSM codec for Internet In my opinion, GSM sounds great but not as good as G.729. So if you can't afford getting G.729, GSM is the way to go. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - Alejandro Cabrera Obed aco1...@gmail.com escreveu: Dear all, I have implemented an Asterisk SIP server for a WAN VPN over Internet. We have users distributed along all my country (Argentina) that register to my Asterisk in order to talk among them. I'll plan to have voice and voicemail with GSM codec, because we can't afford the payment for the G.729 licenses (it's an administrative problem of our company, not an echonomical problem). So in this way Asterisk won't care about codec traslations, this sounds good. What do you think about the use of GSM codec for Internet calls ??? Do you think GSM is the best narrow-band codec if I can't use G.729 ??? Thank you !!! Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
OK, thanks -- will have to try and see what I get.
Darryl Dunkin ddun...@netos.net wrote:
I add this line in our in/out contexts:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging
on). I'm sure you could output it anwhere else as well with a system
call/echo.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful
to know.
Darryl Dunkin ddun...@netos.net wrote:
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are
inconsistent to say the least. I’ve been dumping these to the verbose logs
for some time and monitoring them, but I have not been able to determine
why the numbers are so far off. I am more concerned with the packets lost
due to priority queuing within our network.
Here is an example just today:
ssrc=583450581
themssrc=1093951555
lp=0
rxjitter=0.003219
rxcount=1100
txjitter=0.000275
txcount=1108
rlp=57702
rtt=0.036000
If the txcount is only 1108, how can the remote lost packet count be 57702?
Unless the call was nearly inaudible?
I did verify with this end user, and the call was just fine. Is this an
issue with the phone at the remote end misreporting?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas
Kezys
Sent: Tuesday, September 22, 2009 01:01
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: 2009 m. rugsėjo 22 d. 09:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTPAUDIOQOS
hey all,
can any body know what this parameter stands for
i got RTPAUDIOQOS while i have SIP channels
but could not understand then what this parameter tell
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
if any one know plese help me to or give any documentation link
regards
Dhaval
Alternatives:
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--
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How do
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cov...@ccs.covici.com
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How do
you spend it?
John Covici
cov...@ccs.covici.com
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[asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?
Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a door stopper? -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] solution for NAT issues?
i have also tried setting qualify='yes' but cpu usage spiked to 100%. Ron wrote: Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward port 49923 to port of the IP phone, maybe nat mapping has expired or something. I have tried STUN, still sometimes the phones just cannot be reached. is there any other software to manage binding of ports on specific users so that the routers always keeps the port mapped to port of the ip phone . TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?
On Thursday 12 November 2009 21:18:18 Joseph wrote: Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a door stopper? There is no reason why it should not continue to work. However, there are certain features that the IAXy will never have. One particular item is that you'll need to turn off calltoken support for peers/users which specify an IAXy, as the firmware will never be modified to support that. In addition, the IAXy will never support any codec in the extended space (it probably doesn't have enough CPU to master other codecs anyway). -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters
2009/11/12 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: 2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: What about adding per-span section headers like Asterisk .conf files ? [span1] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [span2] group_lines 2 pri_termtype SPAN/2 TE This implies you will know span numbers in advance. I would like better ways to specify configuration. Really ? I used this [span1] header as an example. Using any other string would be fine for me as what matters, if I'm not mistaken, is the group_lines number : [foo] group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE [bar] group_lines 2 How can you tell which spans / channels will use each section? My understanding of Dahdi is that I mostly need a group number to use with Dial application : Dial(DAHDI/g1/0123456789). To get that dahdi-channels.conf file generated with dahdi_genconf, the only missing feature (if my understanding is correct) is to be able to group together a couple of ports so that I could either include in my diaplans, lines such as Dial(DAHDI/g1/0123456789) or Dial(DAHDI/g2/9876543210). So with a /etc/dahdi/genconf_parameters like this ... [foo] group_lines 1 pri_termtype SPAN/1TE [bar] group_lines 2 pri_termtype SPAN/2TE ... I think we've got everything needed to generate a /etc/asterisk/dahdi-channels.conf file this : ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=1,11 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS group=2,12 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 context = default group = 63 So I don't understand where I would have to tell which spans / channels will use each section. The only purpose of sections within genconf_parameters is to set the scope of parameters like group_lines. Am I correct to think I can't today generate /etc/asterisk/dahdi-channels.conf files in which 2 groups of BRI ports are defined ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 12:38, Fri 13 Nov 09, Matt Riddell wrote:
On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.
Yeah, but would you automate an install with additional contents in the
default context?
We do. It's the only way to get ENUM running on new boxen ;)
and yes I know, I'm not the beginning user anymore.
--
Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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[asterisk-users] Multimedia PBX Solution
We are planning to develop a Multimedia PABX to connect about 500 or more personnel for Voice, Video and Text Communication. www.*gvsc*net.net is a similar solution but we wish to have our own independent solution. Please advise if anyone can offer a ready to go end to end Asterisk based solution. -- السلام عليكم ورحمة الله وبركاته Nazir Ahmed Vaid Cell:+92300-828 eHealth Services (Pvt) Ltd. http://www.ehealth-services.com NexSource Pakistan (Pvt) Ltd. ASK Development http://www.askdevelopment.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multimedia PBX Solution
Nazir Ahmed Vaid wrote: We are planning to develop a Multimedia PABX to connect about 500 or more personnel for Voice, Video and Text Communication. www.*gvsc*net.net http://net.net is a similar solution but we wish to have our own independent solution. Please advise if anyone can offer a ready to go end to end Asterisk based solution. 1) If someone else is offering a ready-to-go, end-to-end product and you purchase it, would that not conflict with your goals of having your own independent solution? 2) A bit of marketing and communication advice, which you can take or leave: Reduce the frequency with which the word solution appears in your sentences by about ... 5000%. -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Health IVR Recordings
Nazir Ahmed Vaid wrote: We are looking for Pre-Recorded IVRs for Health Services in English and other languages. If anyone is aware of a source kindly advise. We are launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose. What makes you think that generic recordings of medical terminology (or whatever is meant by Health Services) are going to work? Triage and emergency room intake is a rather specific sub-domain of the medical lexicon in both its clinical and administrative dimensions. These are unlikely to exist unless someone has already built your intended product, and rather similarly at that. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Health IVR Recordings
We are looking for Pre-Recorded IVRs for Health Services in English and other languages. If anyone is aware of a source kindly advise. We are launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose. -- السلام عليكم ورحمة الله وبركاته Nazir Ahmed Vaid Cell:+92300-828 eHealth Services (Pvt) Ltd. http://www.ehealth-services.com NexSource Pakistan (Pvt) Ltd. ASK Development http://www.askdevelopment.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
