Re: [asterisk-users] Still sipping frustration - only getting state ACK
On Sat, 2010-06-05 at 22:16 +0200, Julien Claassen wrote: > But when I make a call; > channel originate sip/iptel-out/e...@iptel.org Application playback > vm/net_ring >The call is onlyleft in state ACK for a while. Then asterisk tells me, > that > it is destroying the sip dialog (long ID) INVITE. This could be caused by a number of reasons, but the most likely is that your syntax isn't correct above. Try either: channel originate sip/iptel-out/echo Application playback vm/net_ring or channel originate sip/e...@iptel-out Application playback vm/net_ring -- Jared smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI volume
On Wed, 2010-06-02 at 15:35 -0600, Greg Woods wrote: > Is there a reasonably easy way to increase the volume on a DAHDI > channel? The VOIP phones in the house work OK, but for the phones > connected to DAHDI channels on a Digium TDM400P card, the volume is very > low and it's hard to hear if there is any background noise at all. If > this is documented, point me to where and I'll gladly do my reading. You can adjust them manually with the txgain= and rxgain= settings in chan_dahdi.conf. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail : mail attachment to multiple mail-addresses
On Mon, 2010-05-31 at 22:08 +0200, Jonas Kellens wrote: > Is there yet a seperator that actually works to define multiple mail > addresses ? Not that I'm aware of. I simply create an alias on the mail server that then forwards to all the recipients. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting 'username' of sip peer
On Wed, 2010-05-26 at 22:48 +0530, Deepesh D wrote:
> When a call is made from any of these peers I want to get the username
> of the peer.
> for eg:- If a call is being made from 'TestSIPUser' then I want to be
> able to get the value 'testuser'
I can think of two ways of doing this. The first is to use the
SIPCHANINFO() dialplan function, like this:
exten=>123,1,Verbose(0,The call came from ${SIPCHANINFO(peername)})
The other option is to use the "setvar=variable=value" setting in the
peer definition in sip.conf. For example, if you add
"setvar=USERID=jsmith" in a user/peer/friend definition, Asterisk would
automagically create a channel variable named USERID with a value of
jsmith every time this device made a call into Asterisk.
--
Jared Smith
Sr. Trainer
Digium, Inc.
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Re: [asterisk-users] How to get ConfBridge user count
On Tue, 2010-05-25 at 12:07 -0600, Steve Johnson wrote: > How can you determine how many are already in the conference bridge? I don't know that there's a way to do it automagically within ConfBridge. I use the GROUP() and GROUP_COUNT() functions to do these sorts of things. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting presence working in 1.6.2
On Fri, 2010-05-07 at 08:25 -0500, Danny Nicholas wrote: > In which future release of Asterisk are we (since it is open-source, we > theoretically have "some" control) going to stop renaming and deprecating > features? It's obviously more complicated that you make it seem with your comment. Let me try to explain the history of this particular change. In earlier versions of Asterisk (1.2, 1.4, 1.6.0 and deprecated but still working in 1.6.1), you had to set the "call-limit" setting to get Asterisk to keep track of SIP device state. The majority of the people using this call-limit setting set it to an arbitrarily high value (such as 99) so that it didn't really limit the number of concurrent calls, but simply turned on SIP device state tracking. (And, to be honest, it was a whole lot easier to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce arbitrary call limits.) To make it more clear and less cryptic, we split out the "callcounter" functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. Clear as mud? -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portech MV-374 does not register behind NAT
On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote: > All goes well when the gateway is connected directly to the > internet... It's when it is behind NAT the 401 is sent from > Asterisk... Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2 No "soft hangup"?
On Tue, 2010-04-20 at 09:49 -0700, Steve Edwards wrote: > I'd like to see a more natural and intuitive interface following a "verb > noun" model like Oracle, MySQL, or even GDB. We're close to that now, and that's one of the reasons that the "soft hangup" command was changed to "channel request hangup". While it's not "verb noun", most (if not all) of the commands in the Asterisk CLI should follow the "module verb noun" model. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail "maxmessage " setting per mail box
On Tue, 2010-04-20 at 14:34 +0100, Bruce McAlister wrote: > Is it at all possible to have the "maxmessage" setting on per > user/mailbox value? Absolutely, as long as you're talking about the "maxmsg" setting! In fact, there's an example in the sample voicemail.conf file that comes with Asterisk: ;4200 => 9855,Mark Spencer,marks...@linux-support.net, mypa...@digium.com,attach=no|serveremail=mya...@digium.com|tz=central| maxmsg=10 See how we set this particular mailbox to only have a maximum of ten messages? -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS variable and qualify=no
On Sat, 2010-04-17 at 17:38 +0400, Rustam Kovhaev wrote: > could anybody tell me if the info below is still correct: > > Note: In order to obtain useful DIALSTATUS information when dialing a > peer you will need to have qualify=yes in that peer's definition (e.g. > in sip.conf or iax.conf). > http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS > That's not correct. DIALSTATUS will be set whether or not you've got qualify=yes in the peer definition. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer_CONTEXT behaviour
- "Steve Davies" wrote: > I'll have to give that a go. Is there something similar available for > all of the other Channel technologies, or at least for DAHDI and IAX? This works for SIP and IAX from at least the 1.4 release, and in DAHDI since 1.6.0 release. > Is TRANSFER_CONTEXT copied to a bridged channel under any > circumstances? I would be concerned that when Dial() bridges a call, > it will incorrectly copy this variable onto another channel. > Basically > if I dial out of an IAX channel to a 3rd party with a very permissive > TRANSFER_CONTEXT on the calling channel, I do not want to > accidentally > grant permissions to a remote stranger! You'll need to play around with variable inheritance to get it set right. If you define a variable with a single underscore (_TRANSFER_CONTEXT in my example), it'll get inherited by the next spawned channel, but go no further. If you define a variable with two underscores (say, __TRANSFER_CONTEXT), then it will get inherited by the next spawned channel, and any channels spawned by that channel, and so forth. Obviously defining it without any underscores at all means it won't get inherited by spawned channels. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Polycom Dialed Party Name
On Thu, 2010-04-15 at 09:09 -0400, Marc Smith wrote: > On the Avaya's, when you dialed another user's internal extension, on > the phone you are dialing from, it would display the user's name that > you're dialing. It's not supported in your version of Asterisk, but Called Party ID will be supported in Asterisk 1.8. If you're adventurous, you can try out trunk now on a development machine and ensure that it's working the way you want it to before Asterisk 1.8 is released. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer_CONTEXT behaviour
On Thu, 2010-04-15 at 13:59 +0100, Steve Davies wrote: > Let us then assume that the contexts are configured in the config files as: > IAX/1234: context=external > SIP/100: context=default > SIP/101: context=superuser > SIP/102: context=local > It's early and my brain hasn't fully engaged this morning, but couldn't you just do something like "setvar=_TRANSFER_CONTEXT=default" in the definition of user 100 in sip.conf, and "setvar=_TRANSFER_CONTEXT=superuser" in the definition of 101, and so forth? That way, it would get set on the incoming call from that particular user, and be inherited by the spawned call. Am I missing something obvious? -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do AMI Events have timestamps?
On Tue, 2010-04-13 at 13:59 -0500, Danny Nicholas wrote: > They actually do have a timestamp, in a manner of speaking. The uniqueid > field is a pseudo-unixtime stamp. While correct, it's a timestamp of when the call *started*, not when the event happened. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem compiling asterisk with cdr_odbc
- "Nathan Pryor" wrote: "make menuconfig" does not show cdr_odbc as a selectable compile option. I have compiled and installed both unixODBC and freetds from source and have verified both successfully connect to my sql server. Both were installed to standard locations (/usr/lib). I had no problem compiling cdr_odbc on my test server(CentOS 4.6), however following the same steps on my production server (CentOS 5.4) gives no joy. Install the 'libtool-ltdl' and 'libtool-ltdl-devel' packages, and then re-run ./configure. -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User on PC?
On Mon, 2010-03-01 at 23:46 +0100, Leif Neland wrote: > I'm looking for a way for linux to query a pc if user X is on, and has > used the pc recently or the screensaver is not active. > > If so, I'll route a call for user X to the phone near that PC. If you're using a relatively modern version of Asterisk, you could use the res_jabber and the JABBER_STATUS function to see if they're marked as available in their XMPP IM client. (Most IM clients will set the status to away when the screensaver kicks in.) -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
On Mon, 2010-02-22 at 16:13 -0500, JT wrote: > Is this something that is fixed in an update? (Currently running 1.2) Yes... modern versions of Asterisk support SIP session timers. (If I remember correctly, Asterisk 1.2 could tear down a call based on lack of RTP data, but I never found it worked well enough in my tests to warrant its use.) -- Jared Smith 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
Re: [asterisk-users] Realtime extensions
On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote: > Does a context need completely be written or in extensions.conf or in > the mysql-table 'extensions_table' ? Or can I combine the two with the > 'switch'-statement ?? You can certainly combine the two with a switch statement. Asterisk will then only look in the switch if it doesn't find a match in extensions.conf. -- Jared Smith 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
Re: [asterisk-users] Per user voicemail greeting
On Wed, 2009-12-30 at 14:56 +, listu...@spamomania.co.uk wrote: > It all works fine, playing the system VM greating, but I would like to > use the custom .gsm for this user only. Can anyone help? The greetings and voicemail messages are typically stored in the /var/spool/asterisk/voicemail directory. There will be a directory for each voicemail context, and a subdirectory for each mailbox within the context directory. Please be aware that the voicemail system records the greetings in several different formats, so you want to convert your custom recordings to all of those formats, or otherwise ensure that Asterisk doesn't play one greeting for callers with one codec and another greeting for callers using another codec. -- Jared Smith 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
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
On Tue, 2009-12-08 at 14:47 -0300, Andrew Latham wrote: > As most of us already know an E1 has 32 channels of which 30(1-15 > 17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is > not presented in Asterisk Zaptel/DAHDI. There are other > configurations but this is the most common. As an aside, I've seen different documentation in various places that shows this "sync" channel as being channel zero (coming before the first bearer channel), not the 32nd channel. I'm not familiar enough with E1s myself to be able to say definitively that this is the case, but thought I'd throw this out there for discussion (and hopefully more enlightenment). -- Jared Smith 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
Re: [asterisk-users] automon => *1 "one touch recording"
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: > After pressing "*1" console is not showing anything indicating that the call > is being recorded: I find that I often have to adjust the "featuredigittimeout" setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith 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
Re: [asterisk-users] Call Limits
On Sun, 2009-12-06 at 08:49 -0500, Dan Journo wrote: > I’m trying to figure out how to limit the number of concurrent calls a > client can make. I prefer to use the GROUP() and GROUP_COUNT() dialplan functions to enforce arbitrary call limits in Asterisk -- Jared Smith 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
Re: [asterisk-users] b option in Directory
On Wed, 2009-12-02 at 09:40 -0600, Danny Nicholas wrote: > There are indeed lots of improvements, but if the OP NEEDS some features > that automagically worked in Zaptel that still don't in DAHDI (POTS Line > supervision...) *Please* don't continue to bring up this example in the mailing lists. >From everything I've been able to find in my research, there is *no difference* in answer supervision (or far-end disconnect supervision, for that matter) between Zaptel and DAHDI. I've explained this once before, but I'll explain it yet again so that hopefully Google will index it for the next person that comes along and asks it: Let's assume I'm making an outbound call on an analog phone line connected to an FXO port in my Asterisk system. In the United States, the telephone companies don't send any kind of signaling to let me know that the far end has answered my call. Hence the reason Asterisk treats *all* outbound calls on FXO ports as having been answered, unless you go changing settings in zapata.conf/chan_dahdi.conf (like setting "answeronpolarityswitch=yes"). Obviously telephone signaling can and does vary from country to country, which is why have settings like "answeronpolarityswitch" and "hanguponpolariyswitch". In short, there's no difference in Zaptel and DAHDI in this regard, so please don't keep using it as an excuse for people to stick with Zaptel. If in fact there were any regressions of this nature in the transition from Zaptel to DAHDI, rest assured that we would have corrected them by now. -- Jared Smith 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
Re: [asterisk-users] Crosstalk - Is there a debug option for logging this?
On Tue, 2009-11-24 at 14:05 -0500, JT wrote: > I'm struggling with an intermittent crosstalk issue resulting in a > caller's audio being broadcasted to other calls (only one way as they > are unable to hear the others listening in). Crosstalk like this isn't a common occurrence, especially on digital audio paths. > So my thoughts are leaning towards this NOT being an Asterisk issue, > but instead being related to the telco (PRI) config...however without > proper logging this is a guess. I would lean that same direction, but it's not a common problem to see on a PRI, so don't be surprised if your telco's first reaction is "It can't be us... go talk to your PBX vendor". > Is there a debug option in which I can see how Asterisk is routing the > audio for callers? This would at least allow me to capture logging of > the call routing to determine if Asterisk is doing this or if it's > occurring outside of my control. Type "core show channels" at the Asterisk CLI to see each channel, and what it's being bridged to. -- 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
Re: [asterisk-users] make sounds - doesn't pull all audio tarballs.
On Thu, 2009-11-19 at 10:50 -0600, Karl Fife wrote: > The 'Make sounds' routine into Makefile doesn't seem to "pre-fetch" all of > the audio tarballs. > Is this an oversight or is there a strategic reason for it? As I understand it, it only pulls the tarballs you have selected in "make menuselect". Is there a particular reason you want to pull *all* of them? -- Jared Smith 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
Re: [asterisk-users] can't call through voip provider
On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: > Please help me with this, I can find any solution on this pls help. Your help > will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- 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
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
Re: [asterisk-users] Queue device state problem
On Wed, 2009-11-04 at 15:16 +, Alexandre Rodrigues wrote: > I changed my call-limit to one, and the same problem continues when I > restart asterisk. Have you any moore ideias to solve this problem? There's a note in the sample queues.conf configuration file (at least in the 1.6.2 branch) which states the following: ; It is important to ensure that channel drivers used for members are loaded ; before app_queue.so itself or they may be marked invalid until reload. This ; can be accomplished by explicitly listing them in modules.conf before ; app_queue.so. Additionally, if you use Local channels as queue members, you ; must also preload pbx_config.so and chan_local.so (or pbx_ael.so, pbx_lua.so, ; or pbx_realtime.so, depending on how your dialplan is configured). I think if you load chan_sip (and optionally the other modules listed) in modules.conf, that should take care of your problem. -- 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
Re: [asterisk-users] ring groups with different caller id
On Tue, 2009-11-03 at 14:02 -0500, Derek Belrose (OMEGABYTE) wrote: > Is there a way to ring multiple phones simultaneously but use > different caller id settings depending on the type of phone that is > being called? This can be accomplished via chan_local, a channel driver that treats an extension in the dialplan as if it were an external device. As an example, let's say we want to call Alice and Bob (both on SIP devices), and Charlie (on his cell phone, which we'll assume is 555444). In addition, we want to modify the CallerID name and number before calling Charlie's phone. Here's one way to do it: [testing] exten => s,1,Dial(SIP/Alice&SIP/Bob&Local/123...@testing) exten => 123456,1,Set(CALLERID(name)=Someone Else) exten => 123456,n,Set(CALLERID(num)=5551212) exten => 123456,n,Dial(DAHDI/g1/555444) In this example, Asterisk will dial the two SIP devices and extension 123456 at the same time. Extension 123456 modifies the CallerID and then calls Charlie's cell phone number. I realize that chan_local takes a bit of work to understand, but trust me -- once you get used to it, you'll wonder how you got along without it. -- 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
Re: [asterisk-users] Problem with ChanIsAvail
On Tue, 2009-11-03 at 12:14 +, Dan Journo wrote: > I am having a problem with ChanIsAvail. It always returns the same > result, regardless of whether an extension is available or not. > > It always returns 0 Unknown Status. Do you have chan_sip keeping track of device state? By default, it doesn't keep track of device state, as that takes extra CPU cycles. You can turn it on for a particular SIP user/peer/friend by setting "call-limit=99" (or some other reasonable level) in Asterisk 1.4 or "callcounter=yes" on Asterisk 1.6.0 or later. You may also need to investigate "limitonpeer=yes" in Asterisk 1.4 and/or "counteronpeer=yes" in Asterisk 1.6.0 and later. -- 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
Re: [asterisk-users] pattern matching DID
On Sun, 2009-11-01 at 18:50 -0500, Thomas Perron wrote: > Where is everyone located? I am in Virginia, USA There are literally thousands of people on this mailing list, so I doubt it's worth having everyone tell you where they're from. That being said, I'm also in Virginia (near Fredericksburg), and there's enough interest in the area that we might start up a local Asterisk users group in the area. What part of Virginia are you from? -- 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
Re: [asterisk-users] voicmail: no entry in voicemail config
On Fri, 2009-10-30 at 13:23 -0600, Joseph wrote: > In asterisk 1.6 the voicemail prefix u b don't work, I have: > exten => 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail > config file for "u11" > > exten => 1,3,Voicemail(11) works, > > Isn't prefix "u" suppose to play: "The person at extension ... 11 ... is > unavailable," ? Instead of prefixing the mailbox with a 'u' or 'b', use it as the second parameter to the Voicemail() application, like this: exten => 123,n,Voicemail(1...@default,u) You can always type "core show application voicemail" at the Asterisk CLI to see the complete syntax for the Voicemail() dialplan application. -- 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
Re: [asterisk-users] OT - How to organize TFTP root directory ?
On Thu, 2009-10-22 at 11:15 -0400, Dave Fullerton wrote: > #2 might be possible, but there's a lot of "depends on" factors. > > The ISC dhcpd often packaged in linux distributions has the ability to > specify different dhcp options to different "pools" of addresses. You > can then assign clients to pools based on a substring match of their mac > address. This then requires that the client (phone) will use the URL > specified in dhcp option 66. With all this put together you can assign > each brand of phone to its own pool/options where the options point it > to a URL containing the firmware for that brand of phone. > > I do this with my polycom phones and it works well. Don't know if it > works with other brands of phones. I've done this on a number of different phones, using both the ISC dhcpd server as well as dnsmasq. I've never encountered any problems with it. -- 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
Re: [asterisk-users] AGI STREAM FILE and not blocking execution
On Thu, 2009-10-22 at 08:43 +0200, Patrick wrote: > I'm wondering if I can take benefits of long prompts to compute in the > background the next step to be performed by Asterisk. > > Do you know what will be the behavior of asterisk if I send a STREAM > FILE command immediately followed by another command ? Will asterisk > stack commands or will it stop the first one to execute the second one > ? If you want non-blocking (asynchronous) commands, check out the ExternalIVR interface instead of using AGI. -- 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
Re: [asterisk-users] MWI for multiple voice mail boxes
On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: > Hello, all. I have a user who needs to monitor their voice mail box > and > the general delivery voice mail box. I defined them in sip.conf as > follows: > > [tkeeley](a10f) > mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a10&6...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Mon, 2009-10-05 at 12:33 -0500, Danny Nicholas wrote: > What are the limitations of ActionID? In all of the examples I see, it is > usually 1 or some integer. Can it be a timestamp like uniqueid? It is simply a unique string. You can make it a timestamp if you'd like, but I doubt that means you can guarantee that it's going to be unique across concurrent calls. Otherwise, it's not likely to be very useful to you in the long run. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote: > I'm executing some parallel Originate async, is there a way to know > the result of each originate?... > I was looking at the OriginateResponse event, but I don't know how to > get it from my web service. Also, if I have 3 originate in parallel, > how can I identify the OriginateResponse of each one? Whenever you send an action through AMI, you should also provide an ActionID string, which is something you create and should be unique for each action you send. The response from that action should contain that same ActionID, so that you can identify the responses with the corresponding action based on the ActionID. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UpdateConfig
- "Danny Nicholas" wrote: > Two questions: 1. do you need an ActionID line? Danny, It's *always* considered best practice to have an ActionID line in AMI commands, so that you can easily differentiate the responses, especially to asynchronous commands. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
On Wed, 2009-09-23 at 10:17 -0500, Martin wrote: > BTW there should be an Originate app executable from dialplan ... > But since there's none you can do There is an Originate application, but it's only available in newer versions of Asterisk. (I know I have it on the 1.6.2 branch, but I don't remember if it's available on the 1.6.1 branch. I know it's not available on the 1.6.0 branch.) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
On Thu, 2009-09-17 at 15:12 -0400, jon pounder wrote: > Not that I would ever consider taking an exam like that, but I have been > using/configuring asterisk since nearly the beginning of this mailing > list, and I have never touched dahdi or polycom. Someone should still be > able to pass an exam without knowing about specific hardware where there > is more than one alternative to use in real configurations. Let me try to clarify things a bit here... The dCAP test is primary a test of Asterisk skills, not your familiarity with the configuration of a particular brand of phone or with the Digium line of hardware cards. Part of the test does require you to get an IP phone registered and talking to Asterisk, but the instructor should be more than happy to walk you through the web interface of the phone and say "Put the SIP username here" and "Put the SIP password here" and "Put the IP address of your Asterisk server here". If you'd rather use a softphone on Linux, you're free to use that instead of or in addition to the IP phone. Another part of the test asks you to get an analog phone connected to Asterisk. If you don't get this part working, it doesn't have a huge effect on your score. (Less than 5% of the total score comes from configuring the analog phone correctly.) In addition, you are also asked to connect Asterisk to a (emulated) telco. We do give you the choice, however, of using *either* PSTN or VoIP connectivity to do so. In a nutshell, you can pass the test without having any experience on Polycom IP phones and Digium cards, as long as you know how to use Asterisk itself. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not getting the ability to leave a voicemail-message
On Thu, 2009-09-17 at 17:31 +0200, jonas kellens wrote: > vm-intro is an empty file. I deleted the original and replaced it with > a "touch vm-intro.gsm". I'm curious as to why you did this. Why didn't you simply pass the 's' option to the VoiceMail() application to have it skip the introductory message? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
On Wed, 2009-09-16 at 13:14 +1200, Neeraj Chand wrote: > Hmm...so by open book, that means access to the internet? Possible to > get own notes ? You get access to voip-info.org and searching Google to use as a reference. We don't allow copying/pasting of config files, or copying files via the internet or USB sticks. > The most helpful thing would be a past scenario, something that has come > up in previous dCAP exams. > > Can anyone send in a short descriptor of the final prac [real scenario > that has happened before?] Without going into too much detail on the exact details of the dCAP exam, the general idea is this: A small company has hired you to build a typical small-business PBX using Asterisk, and you have 90 minutes to get it up and running. Given the time constraint, we really stick to the basics, so there shouldn't be anything unexpected during the test. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
On Wed, 2009-09-16 at 13:28 -0400, Steve Totaro wrote: > Just tunnel your HTTP traffic over an SSH link and go to some dCAP > brain dump sites. Yes, there are all kinds of technical ways of trying to cover your tracks... I've certainly seen a number of them. That being said, it's pretty easy for me to tell whether someone understands Asterisk or are just copying/pasting configurations from a website. Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. If you're an Asterisk novice, you probably won't pass (even if you do copy/paste configs from a website). If you have further questions about the dCAP exam, I'd be happy to do what I can to answer them. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SPAM] RE: dCAP Exam
On Thu, 2009-09-17 at 14:00 +0430, C. Savinovich wrote: > What about if I use the browser from my cellular phone? Sorry, cell phone use is not permitted during the testing. We've had students try to snap pictures of the exam with their cell phone cameras, so we had to institute a policy against cell phone use. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reproducible crash - known bug?
On Tue, 2009-09-15 at 22:41 -0500, Ian Pilcher wrote: > Running asterisk-1.6.1-0.23.rc1.fc11.i586 on Fedora 11. I can > reproducibly crash Asterisk by associating a single voicemail mailbox > with two SIP extensions. For example: Please open a report on our issue tracker at http://issues.asterisk.org/ -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Time of Day Branching problem
On Tue, 2009-09-15 at 12:39 -0500, Don Kelly wrote: > So, either the book is wrong or Asterisk has been coded to use something > that looks like "and" to mean "or." The book is indeed wrong. (And yes, I wrote the section that is wrong!) Until Asterisk 1.6.3 comes out, you'll need a separate GotoIfTime stanza for each day you want to match on (Tuesday, Thursday), etc. unless they're in a range (tue-thu, for example). -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI hangup detection
On Tue, 2009-09-15 at 11:23 -0500, Danny Nicholas wrote: > The issue is that POTS as a technology does not have Answer/Hangup > Supervision control (This is per the good folks at Digium). This is incorrect. Asterisk *does* support far-end disconnect supervision, if you're using Kewlstart signaling. (Check to make sure you're using signalling=fxo_ks or signalling=fxs_ks in your channel driver configuration. And yes, you must spell signaling with two ls if you're using Asterisk 1.4 or earlier.) If you're simply using loop-start signaling, Asterisk won't be looking for the far-end disconnect signal. (For future reference, and to clarify Danny's point, Asterisk has no way of doing *answer* detection on FXO ports, as the telcos here in the US don't signal when the far end has answered. In other countries, the telcos will often do a polarity reversal to indicate far-end answer. That being said, we absolutely support *hangup* supervision.) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a way to show caller id information on the desktop
On Thu, 2009-09-10 at 12:21 -0500, Jonathan Moore wrote: > I would like to have either a web page or an application that I can > view that whenever a call arrives on the Asterisk server > the application will display the callerid information. A good friend of mine has Asterisk send a Jabber message with the CallerID information of each incoming call. That way, he can be at work and see who is calling his home line. That might be a quick and easy way to do what you're looking for. Otherwise, you're either using the dialplan to push the information to a relational database (using something like func_odbc), or using the Asterisk Manager Interface to poll for the data. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
On Fri, 2009-08-07 at 17:18 +0200, harry R wrote: > Anyone know how to use regcontext et regexten parameter from sip.conf > and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 When this phone registers, Asterisk will automatically create an extension that looks like: exten => 6123,1,NoOp() in the [some-context] context. I use this in combination with DUNDi by setting the regcontext setting to point at my DUNDi advertising context, so that when my phone registers to a particular Asterisk server in my DUNDi cloud, calls get routed to the proper server. I'm sure there are other uses for it as well. For example, you might have something like this: exten => _6XXX,1,Playback(this-phone-is-not-registered) exten => 6123,2,Dial(SIP/myphone,20) exten => 6123,3,Voicemail(6...@default,u) Notice my priority numbering on extension 6123? If the phone is registered, then Asterisk creates priority number one for me. Otherwise, the pattern match plays a message saying that the phone is not registered. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf parameter and sip msg between server <-> client
On Wed, 2009-08-05 at 14:32 +0200, harry R wrote: > - what's the difference between a subscribe request et a register > request ? A subscription in the SIP protocol is saying "Hey, I'd like to be notified when something happens." This is most often used when a phone wants to subscribe to the state of another extension, or to the status of a voicemail box. A registration is where one SIP device tells another "Hey, I'm over here. If you get any calls for me, send them to me at this IP address and port." -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Several mailboxes on SIP peer
On Wed, 2009-08-05 at 13:12 -0500, Jon Moore wrote: > I have in my sip.conf the following > > [jon.moore] > type=friend > mailbox=8100,8150 > > In voicemail.conf, both mailboxes are defined. Have you tried 8100&8150 (using an ampersand instead of a comma)? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone actively using RLT for mobile phone forwarding?
On Tue, 2009-08-04 at 10:36 -0400, Brian Thompson wrote: > My question is, is anyone actively using the Asterisk "RLT" (Release > Link Trunking) feature to bounce these sorts of calls back to the > telco? I know several people that are using Two B-Channel Transfer, which is very similar to RLT. > If so, any caveats pertaining to the combination of RLT and Asterisk > that I should be aware of before attempting to build such a system? The only caveat that stands out in my mind is that your CDR records are only going to reflect the portion of the call up to the point that the transfer happens... Asterisk doesn't currently do anything with the facility message coming back from the telco when the call ends. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone actively using RLT for mobile phoneforwarding?
On Tue, 2009-08-04 at 09:45 -0500, Danny Nicholas wrote: > This is a "hack" solution; There's nothing hackish about it. It's a very useful tool for shortening the call path and freeing up bearer channels that would otherwise be tied up in bridging the calls. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: > Hello, all. After reading the README, UPGRADE.txt, and a quick tour > through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, > one simply compiles and installs over the old installation being careful > to NOT install the sample files? Yes, that's a safe assumption to make, given the fact that you're just bumping minor releases on the same development branch. If you were moving from the 1.6.1 branch to the 1.6.2 branch, for example, you'd definitely want to check UPGRADE.txt for more details of configuration options that might have changed, etc. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote: > Why PC modems were not used as FXO devices? Why chan_modem was > deprecated? it seemed a nicer option instead of buying expensive > gateways. This question has been answered many times, but just for the fun of it I'll answer it again: If PC modems had been ideal telephony cards, we'd still be using them. My own experience with using modems as FXO devices (long before I became a Digium employee) was that they were awful. I encountered problems with echo, half-duplex audio, and lack of far-end disconnect supervision. All of those problems are solved with most modern telelphony cards (except for the ultra-cheap cards, which are still just modems). To put it frankly, I wouldn't wish one of those modems on my worst enemies. > Anyway, for people living really far from USA the price gets > incremented twice or more and this is without considering the > conversion between currencies. > > 1 $ = 5100 Gs., not cheap at all. I understand that the cards are disproportionately expensive in many parts of the world as compared to the United States, because of the difference in economies. I spent a couple of years in Paraguay in the mid 90s, and know what it's like to pay outrageous prices for specialized electronics just because they have to be imported from other countries. (I'm guessing that you're from Paraguay, based on on the monetary conversion you gave. Does Antelco still dominate the telco market in Paraguay, I wonder?) That being said, the cost per port of the Digium cards (or any of our competitors who design their own cards) is still much lower than what you'd pay for traditional telephony cards, such as those manufactured by Dialogic or Aculab. I know that probably doesn't help you afford to be able to buy a more expensive card, but hopefully you have a better understanding of why we don't use modems as FXO devices. If your time and sanity are worth anything at all, it's a worthwhile investment to buy a good solid telephony card. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: > I have a carrier who tells me he will be sending me traffic from a wide > range of IP addresses. > > so I set up a realtime peer as follows: > > [peer] > defaultip=xxx.xxx.xxx.xxx > host=xxx.xxx.xxx.xxx > deny=0.0.0.0/0.0.0.0 > allow=xxx.xxx.xxx.0/255.255.255.0 > insecure=port,invite > > > Yes, he's really claiming to originate from any of the IP in the block > > When I leave the host blank, we reject calls with a 404. > > shouldn't I be able to put in a kind of "wildcard" for his IP block or > am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... "permit" and "deny" statements are used to create Access Control Lists and to limit the IP address ranges. The "allow" and "disallow" statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_dahdi.conf parser question
On Tue, 2009-07-28 at 15:32 -0500, Karl Fife wrote: > My config works fine but I must be missing a concept because a small change > gives an unexpected result. Can someone help me understand the > chan_dahdi.conf parser that would explain this? I'll do my best. > Based on the config below, Channels 1-23 are assigned to the context > inbound-pri, and Channels 25-47 are assigned to the context outbound-pri. As > expected. So far, so good. > HOWEVER when I simply reverse the order of the channels on the last few > lines like so: > > [trunkgroups] > ; We do not do NFAS at this time > > [channels] > echocancel = yes > switchtype = national > > ;This A part (4 lines) was swapped with the B part > context = inbound-pri > signalling = pri_cpe > group = 1 > channel => 1-23 > > ;This B part (4 lines) was swapped with the A part > context = outbound-pri > signalling = pri_net > group = 2 > channel => 25-47 > > > UNEXPECTED: > Channels 1 is unexpectedly assigned to the context outbound-pri > Channels 2-23 are 'properly' assigned to the context inbound-pri > Channels 25-47 are 'properly' assigned to the context outbound-pri That *is* unexpected. If this can be reproduced on your system, please open an issue report at http://issues.asterisk.org/ as this is not the intended behavior. > For what it's worth I notice that in the sample chan_dahdi.conf file > signalling is usually spelled with two L's but at least once spelled with > just one L. Both are correct in canonical human English, but does the > parser understand them both? That was one of my pet peeves with Asterisk 1.4 and earlier. I asked the developers to address it, and it's my understanding that in 1.6.x and later that Asterisk will accept the word "signaling" with either one, two, or even three 'l's. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and G.729 codec: short questions
On Tue, 2009-07-21 at 08:06 -0300, Alejandro Cabrera Obed wrote: > 1) Does Asterisk have installed the G.729 codec by default ??? No, you have to install it separately, as it's not open source. > 2) If I don't want to pay for a codec license, using Asterisk in > "pass-through" mode for G.729 voice communications, do I just have to > download the open source version of the G.729 codec or can I use the > one coming in Asterisk ??? You can use G.729 pass-through in Asterisk without adding anything extra. You only really need the codec module if you're having Asterisk play prompts or record calls or transcode to/from G.729. > 3) If I use G.729 for voice communications and GSM for voice mail > sounds, does Asterisk execute trascoding ??? It will, if you have added the G.729 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
Re: [asterisk-users] 2 Problems with 1.6.2
On Fri, 2009-07-17 at 11:26 -0700, Ira wrote: > I've searched voip-info for MWI information, but either I'm just really > being stupid or something changed. In 1.2 just adding the line > "mailbox=102,104" was all it took to make it work on the Aastra > 480i-CTs we use. I really tried to figure this out without asking > here, but it's been 2 weeks and I'm still failing. Have you tried "mailbox=...@default"? It appears as though you need to specify a voicemail context. -- 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
Re: [asterisk-users] 2 Problems with 1.6.2
On Fri, 2009-07-17 at 13:30 -0500, Danny Nicholas wrote: > In some cases MWI is referred to (perhaps incorrectly) as BLF. Try > searching on that. MWI and BLF are two separate and distinct items. The only thing they have in common is that they both deal with lighting up little lights on a handset. MWI is Message Waiting Indication, where Asterisk sends a SIP NOTIFY message to a to a phone to let the phone know that there is new voicemail in the mailbox corresponding to that SIP device. (You set the corresponding mailbox by setting "mailbox=1...@default" in the peer or friend definition in sip.conf, where 1234 is the mailbox, and default is the voicemail context or section name in voicemail.conf.) BLF stands for "Busy Lamp Field". BLFs are used for *all kinds* of different things, but most often they're used for monitoring extension state of another extension. To make this work, you create a dialplan hint for the device in question to map an extension state to a device state and then make sure that call limits are enforced in the SIP channel driver (so that it keeps track of device state. The phone with the BLF will then SUBSCRIBE to the status of the hint, and then when the extension state changes, Asterisk will send a SIP NOTIFY to the phone to let it know that the subscribed hint has changed states. I know you're only trying to help, but please don't muddy the water by telling people that MWI and BLFs are the same thing. -- 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
Re: [asterisk-users] How to Change size of CDR(accountcode) variable?
On Tue, 2009-07-14 at 00:01 +0200, Benny Amorsen wrote: > Last concern: Does setvar work even for transfers, like accountcode > does? At least in theory, the setvar= setting in sip.conf or iax.conf (or in Asterisk 1.6.0 and later, chan_dahdi.conf) should work just like the Set() dialplan application, in that you can prepend an underscore or two to the variable name to make it inheritable by spawned channels. So, in theory, "setvar=_FANCYLONGACCOUNTCODE=foo" should make that channel variable inheritable by the *next* spawned channel (but not any channels beyond that), and "setvar=__FANCYLONGACCOUNTCODE=foo" should make it inheritable by the spawned channel *and* any channels it spawns, and so forth. That being said, it's just theory. I have not tested this in my lab, but I offer it as a simple suggestion for you to try. Please let us know if this helped. (My gut feeling is that it should work for DTMF and flash-based transfers. I'm a little less sure about SIP-initiated transfers.) -- 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
Re: [asterisk-users] AGI to announce temperature from weather.com XML file
On Thu, 2009-07-16 at 04:49 -0700, Trevor Hammonds wrote: > I would like to have the ability to have Asterisk announce the temperature > -- not using TTS -- within the dialplan. Chapter 9 of "Asterisk: The Future of Telephony" shows you how to build an AGI script to do just that. For a free download, check out www.asteriskdocs.org. There are obviously many other ways to do it. -- 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
Re: [asterisk-users] q: install asterisk + asteris-gui
On Wed, 2009-07-08 at 14:49 -0400, tom wrote: > - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html The Asterisk GUI uses the web server built into Asterisk, so what you're attempting to do here isn't going to work. I suggest you follow the instructions at http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect111_tt1363.html. They may be a bit out of date (as the Asterisk GUI has changed quite a bit since we wrote the book), but it should help you get started. -- 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
Re: [asterisk-users] Resetting Day/Night setting
On Tue, 2009-07-07 at 10:47 -0400, Jeremy Winder wrote: > It seemed to me cron was going to be the best solution. Sounds like overkill to me... why not just use a GotoIfTime clause in your dialplan? -- 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
Re: [asterisk-users] documentation of DAHDI dial options
On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote: > I am searching for the description of the available dialstrin options > for the DAHDI channel (and also other channel types). > > I am not looking for outdated voip-info links, but for the authoritative > source, e.g. something like "core show application Dial" > > Does such thing exists? I don't think that such a thing exists. The only ones I'm aware of are: 1) Channel Groups. DAHDI/g1/5551212 dials 5551212 on the first available channel in group one, searching from lowest to highest DAHDI/G1/5551212 dials 5551212 on the first available channel in group one, searching from highest to lowest DAHDI/r1/5551212 dials 5551212 on the first available channel in group one, going in round-robin fashion (and remembering where it last left off), searching from lowest to highest DAHDI/R1/5551212 dials 5551212 on the first available channel in group one, searching in round-robin fashion from highest to lowest. 2) Distinctive ring DAHDI/4r1 dials channel 4 (presumably an FXS channel), and uses distinctive ring style one. If I recall, there are four different distinctive ring styles... so you could replace r1 with r2, r3, or r4. 3) Answer confirmation DAHDI/1c/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and not consider the call answered until the called party presses #. This is useful because of the way analog signaling works. Without this setting, Asterisk considers any outbound analog call on an FXO port answered just as soon as it has been dialed. 4) Digital calls DAHDI/1d/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and that it's a digital call. If I remember correctly, this is used for ISDN calls to set the bearer capability. I've taken a quick look in channels/chan_dahdi.c in TRUNK, and it seems to match up with my understanding, as I didn't see any other options stand out. While poking around in there, I found the following comment: /* * data is ---v * Dial(DAHDI/pseudo[/extension]) * Dial(DAHDI/[c|r|d][/extension]) * Dial(DAHDI/(g|G|r|R)[c|r|d][/extension]) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * r - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ That's probably as definitive an answer as you're going to get. -- 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
Re: [asterisk-users] Testing the manager.conf: sending and receiving commands
On Wed, 2009-07-01 at 10:25 -0700, bilal ghayyad wrote: > Can I telnet to the asterisk machine at the port 5038 and send and receive > commands to test if the manager is working fine? Absolutely! > How? 1) Make sure manager is enabled in manager.conf (enabled=yes in [general] section) 2) Create a manager user, and give that user permissions (see the sample section in manager.conf named [mark]) 3) Type "manager reload" from the Asterisk CLI 4) Telnet to port 5038, as shown below: [jsm...@mybox ~]$ telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.1 Action: Login Username: jsmith Secret: doughnuts Events: on ActionID: 12345 Response: Success ActionID: 12345 Message: Authentication accepted Action: ExtensionState Exten: 555 Context: lab ActionID: 987654321 Response: Success ActionID: 987654321 Message: Extension Status Exten: 555 Context: lab Hint: SIP/linksys Status: 0 -- 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
Re: [asterisk-users] * as VM for legacy PBX?
On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: > Sounds like good stuff, but my most substantial concerns involved things > like MWI: is asterisk able to "push" that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- 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
Re: [asterisk-users] G.729 licence in devices connected to Asterisk
On Fri, 2009-06-26 at 16:17 -0300, Alejandro Cabrera Obed wrote: > Do IP phones and GSM gateway include valid G.729 licenses or do I have > to pay for them ??? You shouldn't have to worry about them -- the G.729 licensing for those devices is typically included in the cost of the DSP chips inside the phones and gateways. All you'd need to worry about would be licenses for the G.729 transcoding that Asterisk is doing. -- 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
Re: [asterisk-users] video call doesn work
On Thu, 2009-06-25 at 10:56 -0700, gmail wrote: > i am trying to make a video call on asterisk 1.6 Video support in Asterisk 1.6.0 and later appears to be broken. I have a hackish patch that makes *some* calls work, but it's not an elegant fix. See https://issues.asterisk.org/view.php?id=15121 for more details. -- 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
Re: [asterisk-users] Learn Asterisk
On Mon, 2009-06-22 at 23:22 +0530, David @ULC wrote: > I am from the eastern part of India and there is No institute to have > a formal training for Asterisk. Digium does have an authorized training partner in India, but since this is a non-commercial list, I kindly ask that you contact me directly for more information. -- 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
Re: [asterisk-users] Realtime extensions
On Mon, 2009-06-22 at 12:30 -0500, Danny Nicholas wrote: > Hey Jared, Do you have a FAQ reference for some commonly asked questions > like this? I've got an email archive that goes back almost a year, but > finding a reference in 4K+ emails is sometimes difficult, even if I can > remember some keyword to help out. I wish I did... the only thing I really have is my brain. When searching mailing list archives, I tend to use www.markmail.org, but I did a quick search and didn't find the post I was looking for, so I'll re-iterate my understanding of the pattern matching here for posterity's sake: Pattern Matching: Asterisk searches the extensions and patterns digit by digit, from left to right, and applies the following three rules: Rule 1) For the current digit, order the possible matching extensions based on the most constrained match. For example, let's say Asterisk was looking at the first second digit in this context, and the caller dialed extension 123: [pattern-test] exten => _1XX,1,NoOp(Option 1) exten => _1[2-4]X,1,NoOp(Option 2) exten => _1NX,1,NoOp(Option 3) In this example, the second option would be given priority over options one or three, as there are only three possible matches for this digit (2, 3, or 4), while the other options have more possibilities (8 in the case of the N, or 10 in the case of the X). Rule 2) In the case of a tie (when the number of possibilities for this digit) is the same between two extensions, the extensions are then sorted into ASCII sort order. Consider this context for a moment: [pattern-test-two] exten => _1[1-8]X,1,NoOp(Option 4) exten => _1NX,1,NoOp(Option 5) In this example, the second digit has eight possibilities in both extensions... so the extensions will be sorted in ASCII sort order. The 1 comes before the N in the ASCII table, so option 4 will be selected before option 5. Rule 3) If the dialed digit can't match a particular pattern, exclude the pattern from the list of matches. This means that if a pattern was more constrained in earlier matches and therefore at the top of the list of matching extensions, later digits can disqualify it. To illustrate this point, let's look at the following example: [pattern-test-three] exten => 1[2-4]N,1,NoOp(Option 6) exten => 1NX,1,NoOp(Option 7) In this case, let's assume the caller dialed extension 130. After the first digit, the two patterns are tied. After the second digit, option 6 gets sorted above option 7 because it is more constrained. After the third digit, however, option 6 is eliminated because the last digit can't be a zero. That means that Option 7 will match. Clear as mud? -- 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
Re: [asterisk-users] Learn Asterisk
On Mon, 2009-06-22 at 22:50 +0530, David @ULC wrote: > What the best website and book to start learning asterisk ? I'm obviously biased (as I'm co-author of the book), but I recommend O'Reilly Media's "Asterisk: The Future of Telephony", Second Edition. You can download a free PDF of the book at http://www.asteriskdocs.org/ or you can obviously buy a dead-tree version of the book from you favorite bookseller. -- 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
Re: [asterisk-users] Realtime extensions
On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas wrote: > You can prove this by switching the order of the two filters in > extensions.conf. The order that the extensions in extensions.conf appear has no bearing on the sort order. I've explained the sorting mechanism at length in this list before, but I'd be happy to go over it again if anyone wants me to. -- 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
Re: [asterisk-users] Realtime extensions
On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: > For example when I have these two extensions: > > -- _0699[134]X > -- _06[069]XXX > > that are in the database and number 0699123123 comes in asterisk will > always choose exten _06[069]XXX > and when they are in the extensions.conf file asterisk always choose > exten _0699[134]X. > > My question is why? Is it my misconfiguration or that's how it works. I'm no expert on Asterisk realtime, but this definitely sounds like a bug to me. Mind opening a bug on the issue tracker (issues.asterisk.org) so that the developers can investigate further? -- 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
Re: [asterisk-users] Update Caller-ID after Dial()
On Tue, 2009-06-16 at 13:02 +0200, Philipp Kempgen wrote:
> Can you confirm that currently there is no way to update the caller
> ID via the manager interface once the B leg is ringing or connected?
Correct. Well, at least not with 1.6.0 or 1.6.1 or 1.6.2 branches.
> Looks like this would be feasible with the functions introduced in
> https://issues.asterisk.org/view.php?id=8824 ("[patch] Remote (called)
> Party Identification - chan_sip & chan_skinny implementation").
Yes... that bug number spawned a *lot* of additional work for connected
party information (transmission, reception, and updates) that recently
went into the trunk of Asterisk. Those features will be available in
the 1.6.3 branch of Asterisk, once it has been branched from trunk.
I think few people realize just how much work went into getting that
feature working in the core of Asterisk, so I'm going to tip my hat to
everyone that worked on it and say a big "thank you".
> Such functionality could be desirable in situations when a custom
> callerid number to name lookup takes more time than I am willing to
> spend before Dial()ing.
It would be desirable in *many* situations, which is why I'm really
looking forward to doing more with it in the next few months.
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Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
On Wed, 2009-06-10 at 23:00 +0100, Wayne wrote: > I was wondering what the current development plans / patches etc are to > allow Asterisk to talk to Exchange 2007 Unified Messaging with respect > to adding SIP over TCP support? There is experimental support for SIP over TCP in Asterisk 1.6.0 and later. It's probably not perfect yet, but we'd be happy to hear how it works for you, and that will help the Asterisk developers make it better. As far as other things related to the vague notion of "unified communications", there's the code that Terry Wilson just added on being able to read Exchange calendars (iCal/CalDAV are supported as well) from the Asterisk dialplan, there's plenty of Jabber work being done on the IM side, and Asterisk can already store voicemail in an IMAP mail server. (I've long since let go of my Windows skills, but I'm assuming that modern versions of Exchange still let you communicate via IMAP, right?) In short, there are a lot of exciting things happening in the world of Asterisk with regards to unified communications. -- 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
Re: [asterisk-users] IAX2 issue?
On Tue, 2009-06-09 at 19:58 +0100, Steve Kennedy wrote: > Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to > the US. > > The IP address of the remote end changed (though in the config file it's > registered as a name i.e. asterisk.remote.end), my system didn't > recognised the IP change, it must be cached once and then the cached > value used for ever. It's my (limited) understanding that the IAX2 channel driver in Asterisk 1.2 caches any DNS names it resolves, and doesn't bother looking them back up later. Fortunately, that problem has been addressed in later versions of Asterisk. If I remember correctly, Asterisk 1.6.0 and later use the DNS Manager (see dnsmgr.conf) to periodically re-resolve DNS names. -- 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
Re: [asterisk-users] voicemail
On Tue, 2009-06-09 at 12:11 -0500, Tilghman Lesher wrote: > It does, but it also makes listening to messages rather difficult, as the > fallback for languages only works in one direction. That's a very valid point... My intention was to be able to set a language for just the sound prompts, not for the messages themselves, so obviously my use of the CHANNEL function to set the language was short-sighted. Thanks for keeping me honest and on my toes! -- 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
Re: [asterisk-users] voicemail
On Tue, 2009-06-09 at 14:04 +, Jeff LaCoursiere wrote: > "Has anyone patched the voicemail app such that inside/outside messages > are CLEARLY supported", i.e. they have menu options for recording inside > and outside greetings, and the app can accept some form of argument > specifying an inside or outside call? > > This is something that is pretty standard on PBX systems, and I have been > beaten up about it again this morning. Just wondering if anyone had taken > the time to make such a patch. I'm not aware of any such patch, but I was wondering if another approach would be to support multiple languages in busy/unavailable prompts. You could then set CHANNEL(language)=outside for outside callers before calling the VoiceMail application to get the proper prompts. (Of course, the approach makes it much more difficult to record and manage said prompts, but it does have the added benefit of supporting messages in a variety of languages.) Anyway, just my 2 cents (before taxes)... -- 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
Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
> > exten => s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} > 140] ?
> > ${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
>^ ^
> remove the trailing spaces
You'll also want to remove any spaces from around the question mark
(after your expression).
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Re: [asterisk-users] 400 calls at g711 how much cpu power
On Wed, 2009-04-01 at 22:46 -0500, Erick Perez wrote: > So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or > 1.6Ghz. It's been my experience that CPU load of Asterisk don't scale linearly with call volume. I don't pretend to understand all the reasons why, but it probably has a lot to do with call structures inside of Asterisk. For example, searching a linked list is simple when there are only a few items in the list, but the more items that get added to the list, the more CPU time it takes to finish the task, on average. I know the Asterisk developers spent a lot of time and effort improving the performance of the internal structures between the 1.4 branch and the 1.6.0 branch... if I were you, I'd at least give the 1.6.0 branch a shot. -- 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
Re: [asterisk-users] Call quality - how to debug
On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote: > However - my question would still stand, how exactly would I be able to > debug whats going on in the RTP stream? And why its stuttering > (sometimes halfway through a call). > > Any tips or tricks for actually debugging within Asterisk ? Wireshark has a lot of RTP tools for looking at the latency and jitter and dropped packets on the line, which are the most common problems I find when helping people diagnose poor audio connections. It won't tell you what is *causing* the problem, but it will help you know what the problem actually is. >From there, you can start to track down the source of the problem one network segment at a time. For example... is the poor audio being caused by network problems between the phone and Asterisk, or between Asterisk and your upstream provider. -- 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
Re: [asterisk-users] MeetMe and setting conference timeout
On Mon, 2009-06-01 at 14:42 -0500, Danny Nicholas wrote: > That sounds good Jared; This would kill the channel 1 hour after caller 1 > joined conference? Close... it will kill each channel one hour after each channel joined the conference. If Bob joins at 5:00 and John joins at 5:01, then Bob's channel dies at 6:00 and John's dies at 6:01. (You could obviously add dialplan logic to calculate a smaller timeout value for John's call, but I'll leave that as an exercise for the reader.) -- 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
Re: [asterisk-users] Best Current Release for Long Term Use
On Fri, 2009-05-29 at 09:24 -0500, Danny Nicholas wrote: > I beg to differ with you Jared. Since I don't have your email, I'll post > this here. Create this call file. Funny... as you copied me on the email... > On my "test" machine using a TDM400P, the playback would occur in a manner > such that the user would only get 50-70 percent of the message. Correct, as DADHI assumes that the call has been answered just as soon as the last digit has been sent, so the length of time the phone rang before being answered would determine how much of the message was missing. > Changing 5551212 to a cell phone number would create another 5 to 10 percent > loss of message. Correct, because cell phones often have a longer call setup time than pots lines. > Changing DAHDI/g1/5551212 to a local SIP extension would deliver the message > in its entirety. Correct, because the SIP channel driver knows when the call has actually been answered. Please go back and re-read my previous message... it explains why you're seeing the behavior that you've explained here. > On my "live" machine (TDM410P), the DAHDI call actually waits for a > connection, then plays the message after a 1/2 second pause, so I STAND > CORRECTED, AT LEAST SORT OF. The bug is reproducible on the TDM400P but not > TDM410P. The only reason I can think of that this would be happening is if you happen to have "callprogress=yes" in chan_dahdi.conf on your live system. Please note that the callprogress setting is *highly* experimental, and in my experience causes more problems than it's worth. -- 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
Re: [asterisk-users] SVN vs "Regular" Asterisk
On Mon, 2009-06-01 at 11:02 -0500, Danny Nicholas wrote:
> What branch does the SVN release roughly equate to?
Let me see if I can help clear things up here...
Imagine, if you will, a tree growing in a forest. It has a nice sturdy
trunk, and a few branches. As time goes on, the tree gets taller (from
the top of the tree), but the branches stay at the same height relative
to the ground.
Now that you have this image in your head, let me explain a bit about
Subversion and Asterisk development. In Asterisk, developers add new
features to the trunk of SVN. This changes on an almost daily basis,
and always contains the very latest changes to Asterisk. You can check
out the trunk of Asterisk by typing "svn co
http://svn.digium.com/svn/asterisk/trunk";.
In addition to the trunk, our SVN repository has branches as well.
There's a branch for 1.4, one for 1.6.0, one for 1.6.1, etc. Along
these branches, we try to only apply bug fixes and not new features.
Tarball releases of Asterisk are made from these branches, not from the
trunk. So, for example, the only differences between 1.6.0.8 and 1.6.0.9
would be bug fixes. (From time to time a new feature will be backported
if it helps to solve an existing bug, but this is a rare exception to
the rule.) So, let's look at the 1.6.0 branch for a minute. If you
were to check out the 1.6.0 branch using Subversion ("svn co
http://svn.digium.com/svn/asterisk/branches/1.6.0";), you'd essentially
end up with 1.6.0.9 plus any bug fixes that have been applied since the
1.6.0.9 release.
Does that make sense?
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Jared Smith
Training Manager
Digium, Inc.
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Re: [asterisk-users] MeetMe and setting conference timeout
On Mon, 2009-06-01 at 10:22 -0500, Danny Nicholas wrote: > Write an AGI to hangup the users using Asterisk Manager. If you’re > the ambitious type, you could do it with grep and awk from the > dialplan; just hangup the appropriate channels. Wow... that sure sounds like complicated overkill to me. Why not just set an absolute timeout on the channels? Something like: exten => 123,1,Set(TIMEOUT(absolute)=3600) exten => 123,n,MeetMe(blah,d) -- 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
Re: [asterisk-users] Call telco transfer q931
On Thu, 2009-05-28 at 19:38 -0500, Andres Gomez wrote: > Please help me, i need transfer a call in asterisk to other telco > number and free the channel. Can i do with any q931 function?. Asterisk will automatically attempt a "Two B-channel Transfer" if the following conditions are met: 1) You must have facilityenable=yes and transfer=yes set in chan_dahdi.conf for the channels on your PRI 2) The telco switch must have 2BCT turned on for your trunk group. Some telcos require that you pay extra for this feature, and some refuse to turn it on at all. 3) For at least one switch type (I don't remember which off the top of my head), at least *one* of the calls must be inbound from the telco to your Asterisk box. -- 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
Re: [asterisk-users] Best Current Release for Long Term Use
On Thu, 2009-05-28 at 14:47 -0500, Danny Nicholas wrote: > The bug number for #1 is 14935. I developed a similar app that was working > great in the 1.4.21/Zaptel environment, but is now iffy at best in the > 1.4.25/1.6 environment. If this is an analog line connected to an FXO port, then Asterisk has no way of telling whether or not the remote party has answered the call or not. This is entirely due to the way analog signaling works, and works exactly the same under both Zaptel and DAHDI. -- 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
Re: [asterisk-users] Best Current Release for Long Term Use
On Thu, 2009-05-28 at 12:58 -0500, Danny Nicholas wrote: > This being said, I’d probably go with 1.4.21.X since anything above > that replaces zaptel with DAHDI. There are still a lot of things “To > be worked out” in DAHDI – Zaptel is a pretty solid standard. It continues to amaze me when I hear this, as there really isn't much difference between Zaptel and DAHDI. In fact, the only two differences I know about are: 1) The name change 2) Making software echo can modules able to be loaded on a per-channel basis 3) DAHDI will continue to be developed, Zaptel will not I've been using DAHDI in both my personal systems and in the Asterisk training classes I teach for more than six months now, and I have yet to find any reason not to use it. If you're having problems with DAHDI, mind sharing the specifics of what those are? I've also been using the 1.6.0 branch of Asterisk, and it's been *much* more solid than the 1.4 branch for me and the things I use. It's not to say there haven't been some quirky little bugs, but overall I've been very happy with it. As far as Linux distributions go, I'd say go with whatever you're most comfortable with, and will have the best chance of supporting over the long term. I personally like RHEL and CentOS, but Debian or Ubuntu LTS would be great choices as well, as they all have *years* worth of updates rather than months. (Please read these comments as own opinion, and not necessarily being officially endorsed by my employer.) -- 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
Re: [asterisk-users] Playtones Volume
On Wed, 2009-05-27 at 13:51 -0400, Lee Spenadel wrote: > I’ve researched my brains out on this, and can’t find any answer. Is > there a way to adjust the level of the tones generated through the > Playtones command? The only thing I can think of is to use the VOLUME dialplan function before calling PlayTones() to decrease the volume on the Tx side, and then possibly restore it after calling StopPlayTones(). I haven't tested it to see if it works. -- 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
Re: [asterisk-users] 1.6.0.9: Now "Unable to create ... 'DAHDI'"
On Wed, 2009-05-27 at 10:46 -0400, sean darcy wrote:
> -- Executing [646xxx...@longdistance:6] Dial("SIP/172-08276a60",
> ""DAHDI/g2"/1646xxx") in new stack
It appears you're attempting to dial "DAHDI/g2"/1646xxxyyy instead of
DAHDI/g2/1646xxx... Did you mean to put those extra quotes in there?
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Jared Smith
Training Manager
Digium, Inc.
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Re: [asterisk-users] Error ON SIP Incoming TOS
On Fri, 2009-05-22 at 13:57 +0530, DHAVAL INDRODIYA wrote: > i got TOS and retranssmission error on receiving SIP call > chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission > 10caed68-0f1d-df82-da1e-a76c1cb3d...@172.18.100.72 for seqno 43156 > (Critical Response) -- See doc/sip-retransmit.txt. Did you read doc/sip-retransmit.txt? As it explains there, the remote device didn't respond to our critical SIP packet, so Asterisk had no other choice but to terminate the call. You need to figure out why the SIP responses aren't getting back to Asterisk. -- 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
Re: [asterisk-users] Zaptel Not Releasing Channel (PRI)
On Thu, 2009-04-23 at 09:18 -0400, Steve Totaro wrote: > Not sure if this is new or not but Zaptel or Libpri is not releasing > channels properly. I have had issues with calls that stay up for > eight hours, long distance on the telco side, so it is more than just > a nuisance. Have you examined the output of "core show channels" to see what application the hung channels are in? I'd start there. -- 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
Re: [asterisk-users] Digium G.729 licenses
On Fri, 2009-04-17 at 14:14 +0200, Arturo Díaz Almagro wrote: > We have already bought 30 Digium G.729 licenses to install in several > machines, but Digium only has provided on key to use in registration. > We suspect that the registration has assigned the whole 30 licenses to > the same server. Do anyone know how to distribute the licenses among > several servers? Please open a support ticket with Digium's support department... they'll take care of your problem for you. -- 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
Re: [asterisk-users] inbound filed
On Wed, 2009-04-15 at 09:59 -0600, Bayardo Sanchez wrote: > i create inbound confi my confi is: > > [incoming] > exten=> 1246463,,1,Dial(SIP/8003,60,rT) > exten=> 6463,1,Dial(SIP/8003,60,rT) > exten=> 1246463,,n,Wait(5) > exten=> 1246463,,n,Hangup > > but y calling and send this error in my CLI: > > [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 > handle_request_invite: Call from '101396_procall' to extension > '246463' rejected because extension not found. It appears that you have some extra commas in your configuration. Try: [incoming] exten=> 1246463,1,Dial(SIP/8003,60,rT) exten=> 1246463,n,Wait(5) exten=> 1246463,n,Hangup exten=> 6463,1,Dial(SIP/8003,60,rT) -- 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
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
On Wed, 2009-04-15 at 09:58 -0500, Kevin P. Fleming wrote: > It's not enabled by default because when it is used the Asterisk server > loses control of the call and the CDR becomes incomplete. Not everyone > wants that behavior. But since many people *would* like that behavior, wouldn't it make more sense to enable this via an option in chan_dahdi.conf? Maybe "enable2bct=yes"? (It's not like you don't already have to set facilityenable=yes and transfer=yes to get it anyway, and I doubt there are many people who want facilityenable=yes and transfer=yes but not 2bct... But for those few, I guess we can add yet another option.) It seems silly to have to recompile just to get this functionality. -- 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
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
On Tue, 2009-04-14 at 17:52 -0400, Max Metral wrote: > I’m trying to get “blind transfer” from an incoming DAHDI line to an > external number to work on an * 1.6 install using a T1 from XO. The > documentation is very “distributed” and incomplete, so while it’s not > working, it’s definitely more likely my error somehow. Couple > questions if anybody is out there who even knows what TBCT is… > 1) Is this even supported? s Yes, it's supported in Asterisk and DAHDI, but your success in getting it to work will depend on many factors. As I understand it, it only works with certain switch types (I've had the best luck on 5ESS), and only when the telco enables that feature on your trunk group. In my experience, the telcos usually don't enable this feature by default, and it can be a pain to talk them into enabling it. > 2) Does it require some settings in dahdi_channels, or features, > or whatever? It requires the following features be enabled in chan_dahdi.conf (or zapata.conf, for later version of Zaptel): facilityenable=yes transfer=yes > > 3) Would I “trigger” it via a Dial command or commands, or via > Transfer? Neither... it happens automagically! Some time after the second leg of the call has answered, Asterisk will send a facility message to the CO switch saying "Hey, mind bridging these two calls on your end, so I can free up the channels on my end?" If the switch says "OK", you'll see the calls disappear from Asterisk (and the people on the calls won't know the difference). Otherwise, the calls will continue to be bridged by Asterisk. Obviously there are options to the Dial() application that would preclude Asterisk from allowing the transfer to happen, such as the t, T, w, and W options (and I'm sure there are probably more). > 4) Do either or both of the legs need to be answered? It's my understanding that both legs need to be answered and bridged before this will happen, but I'm not 100% sure. One other minor thing I'll point out... assuming that your 2-B-channel transfer is successful, the telco will send a message to Asterisk at the time the call is eventually hung up. Unfortunately, Asterisk has long since forgotten about the call by that point, so it simply writes a harmless warning message to the console and goes on its merry way. (If a developer happens to read this and needs a pet project -- it would be nice if this would update the CDR records for the original call!) I hope that's enough documentation to get you started! Please let us know how it works out for you! -- 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
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
- "Scott Gifford" wrote: > The CDR information contains the entire > duration of the call as billable seconds, including time spent > waiting > in the queue. I would like the billable seconds to only include the > time spent actually talking to an agent. You're absolutely right -- the CDR information is for the entire call. Instead, look at the queue log (typically written to /var/log/asterisk/queue_log). It will tell you most (if not all) of the information you need for creating call queue reports. --- 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
Re: [asterisk-users] IVR Survey
On Fri, 2009-04-10 at 11:04 -0500, James A. Shigley wrote: > But I’m clueless as to how to combined the recordings into one file. I > don’t want the questions in the recordings, Only the caller’s side of > the conversation without the dead space while they listen to the > Qs/Think on their response. I'd use the Monitor() application to record the conversation (it records both inbound and outbound audio, but you'll just throw the one away). You can then use the PauseMonitor() application to pause the recording before playing a prompt, and UnPauseMonitor() to start the recording again after the prompt. > And since this isn’t a vmail account and trying to avoid an AGI script > if possible I’m not sure how to email the recording(s). I also want > to be able to structure the body of the email so that it reads > something like If you don't want to use AGI to do this, you're pretty much limited to using the System() application and finding a way to send your email from the system command line. Not impossible by any stretch of the imagination... it just takes a bit more work. -- 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
Re: [asterisk-users] asterisk command line problem
On Thu, 2009-04-09 at 16:44 +0300, Yavuzhan Canli wrote: > [r...@asterisk1 ~]# asterisk -rx "show channels" > Unable to connect to remote asterisk > (does /var/run/asterisk/asterisk.ctl exist?) That typically means that Asterisk isn't running, or that the asterisk.ctl file is in a different location. Is Asterisk running? If so, can you find the asterisk.ctl file that was created when it was started? -- 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
