RE: [Asterisk-Users] Pager Notification
Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Yes, asterisk supports both email and pager notification out of the box. This is actually pretty flexible, since you can customize the content of the emails to be whatever you want (e.g. some phones support two-way text messaging). Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Yes, this is possible. You would use the outgoing call API to make the call and then use a combination of Sleep(), SendDTMF() and Background() to do the actual work. Personally, I'd use an AGI script to do this so that I could fine-tune the operation. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 IP Address memory
Can anyone tell me how (and for how long) asterisk remembers the IP address for an IAX2 peer? Voicepulse has been going up and down for me, and it seems to have something to do with the IP address changing. Is there a way to force asterisk to run gethostbyname() again for the peer? Or do I just need to restart the daemon when this happens? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 10:10am CST - VoicePulse appears to be down
Ditto here. I can ping but not log in. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 8:46 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 10:10am CST - VoicePulse appears to be down On 30 Jun 2004 at 10:17, Michael Graves wrote: ...my VoicePulse Connect account is timing out on its login requests. Was working fine a hour ago. Michael I can confirm that this is the case from here in New Zealand too. Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Service Agent Auto Login
Title: Message Does anyone know of a solution where I would be able to setup some sort of permanent connection to the asterisk server via IP? I can't have a dial tone in their ears constantly and I need to find a phone or solution which is $150 or less (preferably under $100) per workstation. Check out the XTen soft phone (http://www.xten.com/). This should have just about everything you want except adding the agent to the Queue, which can be handled by using an autodial sequence. If you REALLY don't want the agents to have access to the phone, I think there are a few open-source softphones that you could strip down to bare minimum. --Ernest
RE: [Asterisk-Users] CVS login
If you don't have CVS, then you probably also don't have the kernel source, the development tools, etc. What Linux (hopefully) distro are you using? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Thursday, May 27, 2004 9:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS login On Thu, 27 May 2004, Fabio Donaggio wrote: [EMAIL PROTECTED] src]# cvs login -bash: cvs: command not found Anyone can help me?? Yes. First of all, you need to install CVS. http://www.fluidthoughts.com/howto/cvs/install/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 200 and hold
PS Someone mentioned about some other problems with 2.05e. What kind of problems are they ? For me it would be important to know. The biggest one I know of relates to the speakerphone. When you have the phone set to ring on speakerphone but use headset to talk, an incoming call will bump the current call to speakerphone (i.e. no longer on headset). Annoying when you have a call center that relies on both the headset (for when they are at their desk) and the audible ring (for when they are dealing directly with customers). --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 200 and hold
First, try moving back to 2.05c or earlier. 2.05e has a few problems (remember, it's beta quality) that could be causing this. Second, are you sure that the disconnect on hook or transfer on hook settings are the way you expect them to be. That caught us for a while since we were putting people on hold and then putting the phone on hook, which had the result of disconnecting them. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Thursday, May 20, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom 200 and hold Hi, I've looked through the archives and seen references to placing calls on hold on a snom 200 (any version of the firmware but we have the latest: 2.05e.) Basically, we can't place calls on hold on the snom 200! The manual talks about the Flash button (which is really the R button, as far as I can tell.) Pressing the R button will immediately disconnect the incoming call. Another poster to this list indicated one could just choose another line and the current line will be put on hold. This is not true on our phone: again, the original call is immediately disconnected. We've been all over the settings in the snom 200 and have tweaked a bunch of parameters. So: how does one place an incoming call on hold on a snom 200 so that we can do attended transfer? Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.05a firmware
Christian, That's the wonderful thing about VoIP phones... Just upload new firmware and we can have the best of both worlds! (Thanks for making the change in 2.05c.) Great phones, by the way :) --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Monday, May 17, 2004 12:20 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Please take a look at http://www.snom.com/faq/FAQ-04-04-28-ut.pdf. It describes the hand/headset policy! It was supposed to be an improvement... CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ** Asterisk Sunday Morning News: Contribute to the community
You know what would be cool? A Show Variables command in the cli. It could
return something like this...
VariableScope Channel
=
CallerIDC ZAP/1-1
EPOCH G
EXTEN C ZAP/1-1
...
--Ernest
* Dial plan tips of the week: Discover the variables!
-
When creating a dial plan, there's a lot of logic to help
you. One thing that
takes time to discover is the use of variables.
Asterisk has a range of globally defined variables that you can use to
configure extensions the way you want. Here's a list:
${CALLERID} Caller ID
${CALLERIDNAME} Caller ID Name only
${CALLERIDNUM} Caller ID Number only
${EXTEN}Current extension
${CONTEXT} Current context
${PRIORITY} Current priority
${CHANNEL} Current channel name
${ENV(VAR)} Environmental variable VAR
${LEN(VAR)} String length of VAR (integer)
${EPOCH}Current unix style epoch
${DATETIME} Current date time in the format: -MM-DD_HH:MM:SS
${TIMESTAMP}Current date time in the format: MMDD-HHMMSS
${UNIQUEID} Current call unique identifier
${DNID} Dialed Number Identifier
${RDNIS}Redirected Dial Number ID Service
${HANGUPCAUSE} Asterisk hangup cause
${ACCOUNTCODE} Account code (if specified)
${LANGUAGE} Current language
${SIPDOMAIN}SIP destination domain of an inbound call (if
appropriate)
${SIPUSERAGENT} SIP user agent
${SIPCALLID}SIP Call-ID: header verbatim (for logging or
CDR matching)
Applications that works with variables
* set your own variables with the setvar() and the
setglobalvar() application.
* the gotoif() app lets you can make conditional tests on
variables and
jump to various extensions or priorities of your dial plan.
* the cut() app lets you divide a variable in two or more parts
To learn more, read README.variables in your docs/ directory.
Or visit the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+variables
___
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RE: [Asterisk-Users] snom200 call wait indication
This is pretty obvious, but have you logged into the phone to make sure that the CWI is turned on? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nicolas Sent: Friday, May 14, 2004 1:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom200 call wait indication on my snom200 the call wait indication do not work. if a call comming in my phone say busy here, can anyone help ? nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 2.05b firmware
This happened to me as well. I resolved it by logging into the web interface, going to the Advanced Networking screen and turning off automatic updates. Then, I manually entered the firmware URL and updated through the website (on the Updates screen). It took a few tries, but I think the problem was with the then-current firmware so upgrading solved the problem once and for all. --Ernest From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve TotaroSent: Friday, May 14, 2004 8:29 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] snom 2.05b firmware Is anyone else having a problem updating their snom phones. I updated mine and it is stuck in a loop of checking configuration and then rebooting over and over.
RE: [Asterisk-Users] 2.05a firmware
Does anyone know what kind of file needs to be uploaded for the custom ring tone? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.05a firmware
They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear
The newest snom firmware (2.05a) resolves this issue. It's not yet freely available, but it is in the pipeline. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Wednesday, May 12, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear we have the same problem could you please send me the chan_sip2 info. Thanks! On Sat, 2004-04-24 at 14:23, Geert Nijpels wrote: Ian White wrote: On Apr 22, 2004, at 23:48, Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied? I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: application/simple-message-summary, but with Content-Type: text/plain, so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) Any idea what sort of time frame before chan_sip2 becomes usable in a production environment, or at least becomes part of the CVS tree? I see your note saying that you are using it in production. I'm using it with some changes with -stable. It's developed by oej for -devel. Works great with my SNOM's and Cisco 9760. You can get chan_sip2 through the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=759 I can also send you my -stable version, but you can backport it with some minor trouble yourself. Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 200
My overall impression with the SNOM 200 phones is quite good. Snom (or the people at ABP) have helped me resolve most of the issues that I had with them. Good: Five lines Headset support for both 1/8 and RJ11 cables Attended transfer and conference calling Address book Multiple rings based on caller and callee Speakerphone Works just fine in all core functionality Bad: The handset and phone are too light It's not possible to do a blind transfer while someone is on hold The call parking feature doesn't work with asterisk unless you use the #-transfer feature (this is *'s fault, not SNOMs) In a recent update they took away my ability to switch between headset and handset during a call :( I was using that. I'm also having some volume issues with these phones, but I think the problem is with the telco and our bad phone lines, not the phones. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Wednesday, May 12, 2004 10:33 AM To: Asterisk Mailling List Subject: [Asterisk-Users] SNOM 200 Sorry to ask this here but I believe that it is the best place to receive a feedback... I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *, and the overall impression about these phones... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?
Roughly $1000 - $1500 I believe (I can't get the exact number from this office). We got it from ABP Intl. (http://www.abptech.com/) who were very helpful. I put a review of our complete setup at http://www.voip-info.org/wiki-Asterisk+setup+success+5. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Machado Sent: Monday, May 10, 2004 5:10 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204? On Mon, 2004-05-10 at 12:37, Ernest W. Lessenger wrote: We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make sure you get the most recent software build, the one that came installed on ours was REALLY old. Might if we ask roughly what you paid for it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: X100P keeping PSTN line Offhook
I see that your line signalling is set to kewlstart... Are you sure that your telco provides this? Also, I found that I was having similar problems when there were other devices on the line (like fax machines). The problem usually occurred when someone tried to make an outgoing call on the same line (but not from the * box), but it seemed to happen occasionally when nobody was on the line. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahid Sent: Monday, May 10, 2004 8:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: X100P keeping PSTN line Offhook Tom, Rich and Atif, Regarding your responses, 1. I have previously tried the callprogrees=no. Doesnt solve the problem. 2. If busydetect=yes, calls to PSTN get droped in the middle of the conversations. 3. Havent looked into the MOH thingy. This feature has caused me other problems. Thinking of turning it off altogether. Anyone has any ideas about alternatives ? Thanks for all your help guys. Regards -shahid Shahid [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] basic implimentation
Assuming that you have 1 - analog lines - 4 and that you want your phones to be 100% VoIP (i.e. no Analog handsets): You should just need the new Digium TDM04B bundle and the granstream phone(s). If you have 1 - analog lines = 2 and 1 = analog phones = 2 then you can use the TDM22B bundle. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Wall Sent: Monday, May 10, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] basic implimentation I have confused my self a little bit I want to have a couple of analog lines pointed at my asterisk box, and in turn piped out to a couple of phones. What cards do I need to get? Here is what I understand to be what I need in addition to my computer: Digium X100P + TDM400P Grandstream BT102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?
Can anyone recommend a FXO gateway product that does behave in this more correct manner? We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make sure you get the most recent software build, the one that came installed on ours was REALLY old. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Lawrence Sent: Monday, May 10, 2004 12:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204? I bought a couple of Mediatrix 1204's a few of months back. (Perceived advantages were relatively low overall cost and size per port, and it isn't nearly as vibration sensitive as a PC would be.) Rich Adamson's review from Feb 1 is comprehensive, and the only thing I'd like to add is this: One feature of these units that absolutely infuriates me is its behavior for an incoming PSTN call: 1) upon sensing an incoming PSTN call, it goes off-hook (after approximately two rings) 2) then and only then, it sends an INVITE to the pre-programmed SIP URL 3) if the user agent on the other end doesn't respond with an OK, it eventually plays a fast busy to the PSTN caller This seems so wrong. Shouldn't a more correct implementation be to send an INVITE as soon as the PSTN ringing voltage is detected, and if and only if the user agent on the end OK's should it go off-hook? There's no sense in going off-hook unless there's a SIP phone ready and able on the other end to answer it. Can anyone recommend a FXO gateway product that does behave in this more correct manner? Does anyone know if the new Sipura SPA-3000 can be configured to do this? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Questions about alarm reporting in Asterisk
Any good ideas would be appreciated! We use a package called Nagios to monitor our servers, which works quite well. It has the ability to track service and host dependencies so you don't get flooded with a bunch of service down alerts when the real cause is a bad switch (or similar). It would seem logical for someone (hah!) to write a res_snmp.c for asterisk that would expose a lot of asterisk's internal data. This would seem a logical step toward writing fully functional monitoring applications as well. The module would allow clients to add themselves to the list and receive traps, as well as check for the current status of various variables. brainstorming Okay, this may be over the top, but here goes. Write an asterisk application that sends (and receives) status information to another box over the PSTN. My idea is not only to use this as a way to verify that * is running, but as a way to RELIABLY tell that a remote * box is actively accepting incoming calls. It wouldn't have to be anything complicated, just a heartbeat and some basic details to let the caller know that yes, I'm alive and accepting calls over this line. Simplified protocol: 1) Monitoring box calls up and says (in DTMF): #my CallerID#extension I am trying to reach#I'm a machine, so reply in DTMF instead of voice#the secret code is# 2) The remote box says #your CallerID#Your DNIS#yes I will accept a call to that number# 3) Monitoring box acknowledges and disconnects 4) Remote box disconnects 5) Monitoring box decides whether it likes the answers it received and performs actions accordingly. /brainstorming --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones Sent: Wednesday, April 21, 2004 10:16 AM To: asterisk users Subject: [Asterisk-Users] Questions about alarm reporting in Asterisk I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most common problems are on the infrastructure network but are reported as phone system problems because that is the piece that the customer directly interacts with. I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. Any good ideas would be appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail storage in DB
What about using NFS or AFS for this? --Ernest From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren NaySent: Monday, April 12, 2004 10:35 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Voicemail storage in DB Hey all, Quick Question. I have heard mention that Asterisk has the capability to store voicemail inside a database, instead of storing each voicemail in a separate file under a spool directory. Is this true? If so, does it (or can it) use MySQL? Is there any documentation available showing how to do this? The problem that we are having is that we need redundant voicemail servers and in order to do that we would need to replicate the voicemail "spool" directory to each redundant server ... we haven't been able to find an efficient -yet- cost effective method for this. However, if we can use a mysql database for voicemail storage then I can set up mysql database replication and our problem is solved. Thanks so much for your help! Darren Nay - [EMAIL PROTECTED]
[Asterisk-Users] T100P specs
Does anyone have the physical spec sheet for the T100P from Digium? The one on the website doesn't have what I need. Things like 3.3 or 5v operation, uses n IRQ channels, requires 32-bit PCI, must be installed while standing on one foot and reciting the GPL, etc. Also, if anyone is selling a used T100P or TE4xxP I'd like to talk... Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 200
having problems with snom phone installstion Please tell us what's up. I recently installed several SNOM phones and worked through many minor issues. Let me know and I'll tell you what I can :) --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Exception flag set - snom200
What version of asterisk are you using, and what version of the SNOM firmware? --Ernest From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jcSent: Tuesday, March 30, 2004 10:20 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Exception flag set - snom200 Sorry I forgot the subject in the last post. When my snom200 receives an inbound SIP external sip call, it somehow rejects the call and with a busy tone. The debug shows the following error: channel.c:1142 ast_read: Exception flag set on 'SIP/sipphone-7796', but no exception handler what does this mean and how can I debug it further?? Thanks JC
Re: [Asterisk-Users] Snom 200
At 11:39 AM 3/22/2004, you wrote: Progress It seems I can't hear the Say Time, due to RTP Double NAT I'm guess this is both problem 1 and 2 really issue. My config: IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server ANyone know of work arounds the double NAT? or other methods to route RTP with snom 200s, to work around this? The best workaround I know of is to use a VPN or other encapsulating technology to avoid the NAT problem entirely. I don't know that there is any reliable _and_ universal way to deal with double NAT and RTP. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk application
At 09:52 AM 3/16/2004, you wrote: I need to setup asterisk so that users can dial into asterisk using normal phone lines and and enter a number when prompted then this number should be accessable to a backend app. is this possible in asterisk. any pointer would be helpfule Yes, this is possible. You could use AGI (see the examples in the CVS tree) and save the number to a database, text file, etc, or your can use the built-in asterisk database if you only need to store one number at a time. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk application
At 10:16 AM 3/16/2004, you wrote: and what would i need to connect asterisk to 2 normal phone lines You would need two FXO cards from Digium to connect to two Telco lines. You would need two FXS cards from Digium to connect to two telephones. Telephone - FXS + Asterisk + FXO - Wall (Telco) If you buy the cards from Digium they will assist you in installing them and getting things up and running. There is an excellent website http://www.voip-info.org/ with information that will assist with AGI development (and everything else as well). --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold sound goes off if environment is silent
At 08:37 AM 3/11/2004, you wrote: Music on hold works if the environment is noisy. But in case of silence the sound goes off. If I scratch continuously on the mikrofone, then the replay works without any interruption. Q: is there a parameter which influences this behaviour? Whatever phone or softphone you are using, you need to disable silence suppression. Why? Dunno exactly. In the newest version of Xten, the feature is Advanced System Settings - Audio Settings - Silence Settings - Transmit Silence - Should be Yes. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP connection broken
At 07:07 AM 3/2/2004, you wrote: Ahhh, you must have upgraded to firmware version 4.2. I had the same problem because I didn't find the new parameter that they added in this release for broken RTP connections. Here is how I fixed it: BROKENCONNECTIONEVENTTIMEOUT = 36 That did it, thanks! Hey, if you get RFC2833 DTMF bridging to work on that gateway, let me know. I currently have a bug report open on them because Asterisk doesn't seem to interoperate with the Audicodes in that respect. I've tested what you describe with the MP-108, a SNOM 200 phone (2.03o firmware) and the most recent CVS of asterisk. No problems at all. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
At 08:33 AM 3/1/2004, you wrote: I'm curious what distro of linux you used. I also can't seem to find a listing of dependancies asterisk requires, even though they are probably staring me in the face. I used Fedora Core 1:) Basically (I'll try to find the instructions when I get a chance) you install all the necessary RPMs to a directory other than the root. Then, create an image and upload it to a flash card. There are some other tweaks (like using tempfs for the /tmp partition) but otherwise everything worked like a charm. --Ernest Thanks -Matt - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 10:25 AM Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage At 08:01 AM 3/1/2004, you wrote: Hello All, I was wondering if anyone is successfully running asterisk on a system with solid state storage, such as a compact flash card? I'm looking for some pointers on doing this. I've gotten this working on a Soekris Net4501. There are several Soekris distro's available, just search the web for Soekris and linux. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
None of this is mine, but it's useful all same :) http://www.xs4all.nl/~hreuver/net4501-try1.html http://www.antlinux.com/staticwiki/LinuxOnSoekris.html At 08:49 AM 3/1/2004, you wrote: At 08:33 AM 3/1/2004, you wrote: I'm curious what distro of linux you used. I also can't seem to find a listing of dependancies asterisk requires, even though they are probably staring me in the face. I used Fedora Core 1:) Basically (I'll try to find the instructions when I get a chance) you install all the necessary RPMs to a directory other than the root. Then, create an image and upload it to a flash card. There are some other tweaks (like using tempfs for the /tmp partition) but otherwise everything worked like a charm. --Ernest Thanks -Matt - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 10:25 AM Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage At 08:01 AM 3/1/2004, you wrote: Hello All, I was wondering if anyone is successfully running asterisk on a system with solid state storage, such as a compact flash card? I'm looking for some pointers on doing this. I've gotten this working on a Soekris Net4501. There are several Soekris distro's available, just search the web for Soekris and linux. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
At 08:51 AM 3/1/2004, you wrote: I have 5 BT phone lines coming into my office. We use four for international calls, and one for local/mobile calls. We have just obtained another call carrier, and now we would like to be able to make calls from any phone to any carrier, without having to remember what details to tap into the phone. I would like all calls to be prefixed with the relevant codes, so that my employees can all dial direct. Also, incoming calls, I want them all redirected to just one phone, the one in reception, and then diverted as required. Is the above possible?? Absolutely; it's exactly what we do here at our office (in the US). --Ernest
[Asterisk-Users] RTP connection broken
We have an Audiocodes MP-108 that keeps dropping connections to voicemail after exactly ten seconds. All other calls are normal, and voicemail works fine from SIP devices other than the gateway. The reason given for dropping these calls is RTP Connection Broken. I suspect that the gateway is sensing the lack of audio from Voicemail and is panicking. Any suggestions? --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail cutting off messages on SIP
We have a situation where voicemail coming in (i.e. FXO-Asterisk-Voicemail) through a Mediacodes MP108-FXO are getting cut off a couple of seconds early. I recall a thread about this quite a while back where this was happening due to silence detection on ZAP channels... Has anyone experienced this and/or found a solution? The MP-108 is using Polarity reversal, but no silence detection. Also, this problem doesn't happen on internal messages (from SNOM 200 phones). Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Voice Quality IP Hard Phones
At 05:26 AM 2/25/2004, you wrote: I am in the middle of getting my self some hard phones. Anyone care to comment on the *voice* quality of the following phones: Cisco 7960 Siptone II SNOM Budgetone I have seen a few reviews, but none go to deep into the voice quality issue. I have not received any complaints here about the voice quality on 13 SNOM phones that I installed a few days ago. There have been several complaints about other things (mostly unrelated to the phones themselves), so I'm sure I would have heard :) We are using the g711a codec in-house. --Ernest Thanks. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Voice Quality IP Hard Phones
At 05:49 AM 2/25/2004, you wrote: The Snom 200 phone mostly functions well, however the phone's logic is more oriented to european telephony and several of the functions do not work in a manner that one might consider 'standard' in the US. It's light-weight, pulls across the desk when the handset cord is stretched, handset is somewhat different when compared to old analog phones, some of the front panel keys have multiple functions (mostly undocumented) that can get one into trouble, etc. Still a good phone with many good characteristics including voice quality, but if you let a non-technical user eval both the Snom and Cisco, the user will pick the Cisco just about every time. multiple functions (mostly undocumented) that can get one into trouble. Oh, lord can they! We discovered the hard way that the transfer button means transfer when only one call is on the phone, and conference when two calls are on the phone! The staff here have declared that they just won't use the transfer button and will use the # button (an Asterisk feature) instead. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detection of extension
At 09:21 AM 2/25/2004, you wrote: Ernest W. Lessenger wrote: At 08:15 AM 2/25/2004, you wrote: This may sound silly but how can I say to asterisk that new number have been dialed and that it has to treat these as a new extension ? I mean: I have received a call, and now I want that asterisk execute the command, by example call forwarding, recording... that I can do when I dial a precise extension ... What do I have to do ? Asterisk see the digits, but never uses them as an extension... http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Thanks but it doesn't explain what I looking for: the ability to press to press keys in the middle of a call and to have asterisk that try to match such extension... Oh, I see what you mean. What about using the # transfer feature? The alternative, I suppose, would be to hack the Dial command so that it loads an AGI script in the background. I don't think there's any existing way to do this, but others are welcome to correct me :) --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Voice Quality IP Hard Phones
) never use the SNOM's built-in transfer button. Instead, they press ESC to hang up and use the # key to initiate a transfer. ABP also tried to sell us a Power-over-Ethernet device that would provide power to the SNOM phones during a blackout (in conjunction with a UPS of course). This is a problem that affects all VoIP phones, but our wiring is not PoE friendly. So, we went with the external power supplies for an extra charge. When we recommend/resell SNOM phones to our customers, we intend to sell the PoE system as well, wiring permitting. All-in-all the office staff here is settling down to the new phones and there have been no show-stopping issues so far. Asterisk: Wonderful, it does everything I want. A few of the things it does: - Voicemail - Private and public extensions - Directory Service - Call Queues - Music on Hold (donated by our local High School jazz band) - Works with Cisco, SNOM, Pingtel, Budgetone, X-Lite and more - Automated attendent and IVR - A simple app that I wrote to play a message based on our network status (we are an ISP) - Fully customizable ring groups (i.e. ring all phones after hours, or only the receptionist during hours) - Conference Calling and Three-way-calling (enhanced by the SNOM phone's conference call feature) - Automatic Failover from PSTN to Internet and vice-versa - Support for multiple VoIP dialtone providers for low-cost long-distance - Transfer to cell phone VoicePulse: We use VoicePulse for our outgoing long-distance, and so far have not had any complaints. We were using NuFone, but are turned off by their lack of a web interface for refilling our account and viewing CDR. Both NuFone and VoicePulse have recently had (reported on the list, but not personally confirmed) outages that did not affect us. In both cases, setup and ordering were quick and easy (more so with VoicePulse). ABP Technology Partners: After an initial testing period with Asterisk and X-Ten, we purchased a single SNOM phone from ABP (I don't recall how we found them). They were very helpful and were willing to sell us a single phone, which is always a pleasant surprise when dealing with distributors. The single SNOM phone we received worked well, so we went ahead and purchased an additional 12 phones and the MP-108 gateway on the recommendation of our salesperson. Our experience with them has been very good, and the (uncompleted) RMA process of one defective unit has gone smoothly so far. --Ernest W. Lessenger OACYS Technology OACYS TECHNOLOGY is a 23-year company, founded in 1982 to develop and deploy computer solutions. Based in a semi-rural community, the company has long been accustomed to operating independently and developing self-reliant solutions with minimal external dependencies. We have developed our own solutions to address and manage substantial competition and adversity, and we continue to do so in the course of our own daily operations. For our consulting clients we explain step-by-step what we have done, why we have done it, and how to do what we did (or would do or avoid) to resolve any of the many challenges facing today's independent ISP/WISPS who are interested in winning their own battles and wars. http://www.oacys.com/
Re: [Asterisk-Users] Calls always parked on 701
At 11:48 AM 2/25/2004, you wrote: No matter what I put in parking.conf for parkpos, I find that the first call is always parked on 701. Is this a bug? With recent CVS builds I've been able to specify 7000 and 7001-7200 as the call parking lot. I haven't tried any other numbers. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls always parked on 701
At 12:34 PM 2/25/2004, you wrote: On Wed, 25 Feb 2004, Ernest W. Lessenger wrote: With recent CVS builds I've been able to specify 7000 and 7001-7200 as the call parking lot. I haven't tried any other numbers. The parking lot is assigned by the user or by the system? I found that my * is assigning 'lot' 701 for my parked calls (as I don't have any other parked calls, it will always be assigned to the first lot which is 701 here). Did I make a mistake or the space is really assigned by *? The parking app always assigns your call to the first available slot. There is not currently any way to specify that you want to park a call at, say, lot 702. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thread-safe applications
I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Thread-safe applications
At 08:31 AM 2/23/2004, you wrote: On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote: I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Could you elaborate, please? What specific applications are not thread-safe and what aspect makes them not thread-safe? Whoops, you're right, the String Manipulation function I was looking at is thread-safe (but some it it's variants aren't). Regardless, do Applications need to be thread safe? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Thread-safe applications
At 09:14 AM 2/23/2004, you wrote: Why would you program something that isn't thread safe? From what I can tell, it isn't much extra effort to do things the right way instead of debuging crap later. I wouldn't, and generally don't. But sometimes (rarely) you need to include functions that aren't thread-safe (ex. specialized operations from vendors who charge a lot of money for poorly-written APIs) and it's good to know what the requirements are. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Connection - Voicepulse
At 05:15 PM 2/19/2004, you wrote: I usually use [EMAIL PROTECTED] they do eventually get back to you. We operate a call centre and have offered them an inbound package, but it seems they are not interested. Matt P.S. Our DID line hasn't been working for around a month nowin the process of signing up with other companies What other companies have you found? We've used NuFone, but aren't too impressed by their payment and CDR interface (i.e. email the salesperson). Otherwise they seem to be stable and knowledgeable. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ultra-cheap asterisk box
At one point I had Asterisk running on a Fedora Core 1 based embedded system using a Soekris embedded device. Once the OS is running, the only hard part is finding a source of timing for the MOH and conference calling. However, I think the new Soekris units have a timing source on them (USB). --Ernest At 03:06 PM 1/15/2004, you wrote: Hi, I'm interested in participating on the embedded side. One of our RD labs is working on a number of embedded server solutions, including servers that are built around a 3 square PCB, linked to a 2 square PCB with a compact flash interface. It's robust, and up to military standards (but it's within the civilian domain, so there are no import/export restrictions). I'm looking to build a solution, with a custom Linux dist (that's not my domain, so I'm looking forward for other people to take this up!), which can be built into a number of sizes: - 1u 19 rackmount, but only 400mm deep, so circa that of a router or a switch. Think - Cisco - 1u 8.5 rackmount, mini-Lan cabinets for residential applications - 3u 19 rackmount, only 400mm, but with front loading for drives, compact flash (two interfaces for swapping Asterisk loads) and LCD status or LED status (basically, enough room inside to have two PSU's for redundancy and space for two or three E1/T1 PRI boards. - Robust IP66 grade outdoor unit - for emergency applications, and for temporary backup solutions We have the capability to manufacture these - so I see potential in developing some robust solutions for the small-biz, and even medium-biz markets. Contact me offlist for specific's, or onlist for more group-orientated specifics. Ad. On 15 Jan 2004, at 7:10 pm, [EMAIL PROTECTED] wrote: Message: 14 Subject: Re: [Asterisk-Users] ultra-cheap asterisk box From: Nicolas Gudino [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: House Internet S.R.L. Date: Thu, 15 Jan 2004 15:52:43 -0300 Reply-To: [EMAIL PROTECTED] On Thu, 2004-01-15 at 14:31, Chris Albertson wrote: I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. I'm also looking at this. I was thinking on a system without a hard drive, booting from a pendrive or flashdive. I want to avoid moving parts, they always break or get dirty and are noisy. If there are other people working on this, we might join efforts and work together and came up with a small linux version with asterisk included, that can boot from a pendrive or a cdrom. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
At 06:53 PM 12/28/2003, you wrote: Side note, and probably not related, but what's the SB live card for? You don't actually use this computer, do you? It's a server, let it be one... Asterisk requires a timing source to play music on hold and conference VoIP channels. The SB performs this function. However, I thought an fxo card was supposed to provide timing... --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: SIP vs. Skinny protocol
At 11:10 AM 12/24/2003, you wrote: Skinny phone functionality is 'richer' than SIP phone functionality. First off, on a skinny phone, in hands free mode, you can start dialling and the phone will automatically go off hook. Sip requires you to manually hit the speaker button, hit new call, or pickup the phone before dialling. (One extra confusing key stroke I have a hard time getting over). Um, that's a feature of the phone, not of the SIP protocol. My SNOM 200 lets me dial before picking up the handset no problem. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fedora core 1 install problem
At 09:20 AM 12/17/2003, you wrote: Hi, I am trying ti install an asterisk system on fedora core 1. During the make of asterisk I got the folowing problem: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe Does anybody know how to solve this? David I can't, but I can tell you that I installed Fedora Core 1 with Development Tools, Editors and Kernel Development, and that asterisk installed without any problems at all. --Ernest \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o tdd.o tdd.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o acl.o acl.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o rtp.o rtp.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o manager.o manager.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe [EMAIL PROTECTED] asterisk]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] On Hold - Talked about before
At 08:45 PM 12/9/2003, you wrote: Ok - Here is where I am at. I know this topic has been discussed before, but never a solid answer was set in place. Is anyone aware of any phones that can put a caller on hold and the caller hear MOH by the user pressing the hold button. I understand most phones are only muting the speaker and handset. The SNOM phones can do this, and are also excellent phones generally. Install the 1.6x software build for now; the 2.x build changes their behavior a bit and breaks MOH with asterisk. This is being worked on. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom X MOH
At 12:23 PM 12/8/2003, listas iPfone [EMAIL PROTECTED] wrote: I updated my snom200 to 2.02t and now MOH from * don´t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I am having the same problem. You can resolve it temporarily by downgrading to the 1.6.x series of SNOM. I am BCC'ing this email to a SNOM representative who is working on this issue. --Ernest
Re: [Asterisk-Users] Problems with voicepulse.com
At 01:32 PM 12/8/2003, you wrote: Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get Registration Refused errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are trying to use IAX instead of IAX2 as the protocol. I have pasted the exact error message (with account info deleted) and their recommended configuration files below. Any and all assistance would be GREATLY appreciated. Stupid question: How long did you wait before calling support? We got Registration Refused messages for about two hours after we created our account, but everything's been fine since. You might check with them to make sure your account was created properly. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XBOX as and * Dedicated Server
At 07:17 AM 12/5/2003, you wrote: I guess for the XBox you would need some external gateway. Audicodes or Mediatrix come to mind but they start at $500. A year ago, I installed Linux on Playstation 2. I had to purchase it with the hardware for about $200. (40GB, keyboard and some network adaptor). It actually worked. And its a much more open community than Xbox. Now that you mention it I will revisit this. http://playstation2-linux.com/ I have asterisk running on a $300 Soekris motherboard. Works perfectly. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
At 02:34 PM 12/4/2003, you wrote: However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Eight quad-span T-1 cards from Digium: $8,970 Three reasonable-quality asterisk servers: $1,000 One T-1/DS-3 MUX: $5000 Total system cost: $14,970 That actually sounds quite reasonable to me. However, if I were doing this myself I would look hard at getting a MAX TNT with VoIP capability off eBay. The price would be equivalent or less, the interface would be more complicated, but all the DSP would be done by the MAX. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
Correcting an idiot-math error (24/4 != 8 and 1000*3 != 1000) ... At 02:34 PM 12/4/2003, you wrote: However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Six quad-span T-1 cards from Digium: $8,970 Three reasonable-quality asterisk servers: $3,000 One T-1/DS-3 MUX: $5000 Total system cost: $16,970 That actually sounds quite reasonable to me. However, if I were doing this myself I would look hard at getting a MAX TNT with VoIP capability off eBay. The price would be equivalent or less, the interface would be more complicated, but all the DSP would be done by the MAX. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and G.729a
At 10:37 AM 12/2/2003, you wrote: Does asterisk support G.729a or do you have to add something (is there an open source one) Yes, Yes, and Maybe (i.e. it's not free, but you can license one through Digium, and there is a reference source available but absolutely NOT open-source). Check out this page for Digium's pricing on g.729a channels: http://www.digium.com/index.php?menu=asterisk_g729. Also, search the mailing list archives for several dozen threads on this subject. --Ernest
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
At 10:59 AM 11/22/2003, you wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? What kind of phone do you have? MOH depends first on the phone, as it is the phone that decides what to do when you press the hold button. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review Comment
Sounds like a great idea! I'll gladly help if requested (I'm a technical writer). Comment: I don't see anything on echo cancellation. That's a big enough and common enough issue that it deserves some discussion. --Ernest At 10:46 AM 11/21/2003, you wrote: Hi Steven, I think this is a great idea and the best way to make users more familiar with Asterisk and its configuration and usage. I can and will provide input for all H.323 related sections. Michael. Steven Sokol wrote: Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an ebook that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly Co.). Below is the outline for the book. We REALLY need as much input as we can get. I would like to completely flesh-out the outline, then I would like to start accepting submissions from the user community for each of the sections/chapters/topics covered in the outline. I have to stress here that I AM NOT AN ASTERISK GURU. I need help from the real gurus (especially: Steven Critchfield, John Todd, Tilghman Lesher, Olle Johansson, and where possible/necessary Mark and Martin). If this works, it will help Asterisk achieve the same kind of global success as Apache, Samba, and other Linux staples. If you want to chat, I am lurking in the #asterisk-doc channel on Freenode IRC. I'm 'ssokol'. Others on this project (so far) are Jared Smith and Leif Madson. A living copy of this outline can be found at: http://www.sokol-associates.com/outline.htm . I will try to update it daily with your suggestions. It will also be the basis for the project outline (completion %, assignments, etc.). Thanks, Steve Sokol Sokol Associates, LLC [Outline Guide] The following outline describes the layout for the book. 1. - Section a. - Chapter 1) - Sub-Chapter i. - Topic Heading *. - Sidebar Heading 1} - Graphic or Chart 1 - Table [Outline] 1. Introduction to Asterisk a. Introductory letter from Mark Spencer 1) Whatever Mark has to say... 2) Digium Reference Information i. Web Site ii. Phone Number b. The Business Case For Asterisk [Somebody From The Business Side Writes This] c. General concept of asterisk 1) Asterisk: Swiss Army Knife of Telephony 2) PBX, IVR, ACD 3) What To Expect i. Asterisk Is Not A Turnkey System ii. Don't Like It? Change It Yourself. iii. Opensource, GPL and LGPL Licensing d. Asterisk architecture 1) The Big Picture 2) Channels 3) Codec Conversions 4) Etc. e. Key components 1) Asterisk software i. Asterisk (Main PBX Channels) ii. Zaptel (Drivers for Zaptel Hardware) iii. Libpri (ISDN PRI Drivers for Zaptel) 2) Zaptel Hardware i. Overview ii. X100P - Single Port FXO Line Interface iii. S100U - Single Port FXS USB Interface iv. TDM400P - 4 Port FXS Analog Interface v. T100P - Single Span T1/E1 Interface vi. TE410P - Quad-Span T1/E1 Interface 3) Channels i. Zaptel Devices/Channels ii. The IAX Protocol iii. SIP iv. MGCP v. Skinny vi. H323 4) Applications i. Dial and Other Basics ii. Voicemail iii. Dial-Plan Scripting 5) Extensibility i. AGI ii. Custom Applications f. Add-On/Optional Components 1) Software i. Gnophone ii. VoIP Soft Phones iii. DIAX iv. Gastman v. Open H.323 2) Hardware i. VoIP Hard-Phones ii. VoIP Gateways ii. Channel Banks 2. Installing Asterisk *. Asterisk Quickstart 1) Install PC Hardware 2) Download Asterisk Software 3) Build Asterisk 4) Install Asterisk 5) Configure Autostart a. Requirements *) Picking A Solid System 1)
Re: [Asterisk-Users] 4 Port FXO cards
At 07:26 AM 11/20/2003, you wrote: Probably too late to ask for, but for us reversal polarity detection (far end answer supervision) is very important for billing and pre-paid purpose. Don't the X100P cards already support this? I believe it's called KewlStart. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] OT : For the SQL gurus..
At 02:08 PM 11/13/2003, you wrote: Now... for the self empowered type... You can go to http://lists.digium.com and remove yourself... but I still would like to see what is meant by or ELSE. Presumably he means or else I'll have to actually look at the instructions printed at the bottom of every post to this list. --Ernest Tom Walsh Network Administrator http://www.ala.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
At 11:07 AM 11/10/2003, you wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 (code field not indexed) = 47.183s
Query1 (code field indexed) = 45.731s
Query2 (code field not indexed) = 109.321s
Query2 (code field indexed) = 2.302s
Query2 has additional overhead in the script as well because it has to
itterate through the number and build up the query..
Query1 is far simpler to use in a script becasue the query does not have
to be built up..
Since you only need to do a simple lookup, why not either (a) build your
own db or (b) use berkely DB or some other fast database engine? Since all
you really need to do is a prefix search on a key:
struct node {
char num;
struct node* p0;
struct node* p1;
struct node* p2;
struct node* p3;
struct node* p4;
struct node* p5;
struct node* p6;
struct node* p7;
struct node* p8;
struct node* p9;
char* desc;
}
That's 48 bytes per record (not counting the description). Memory usage
will depend on how much data you need to store, but lookups would be O(k),
where k is the length of the key.
--Ernest
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Re: [Asterisk-Users] OT: Document Control System?
At 10:28 AM 11/11/2003, you wrote: I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The documentation also has to be written on a Windows desktop platform.. preferably would like to somehow use MS Word. If anyone has any suggestions it would be greatly appreciated. I saw some software called Laboratory Document Control System, but I don't have $999 for a single user license :) (or any money at all actually) Check out the site http://www.voip-info.org/tiki-index.php?page=Asterisk. If you are looking to create * documentation, I recommend that you get an account here and starting adding to it. For general purposes, you can find the TikiWiki site at http://tikiwiki.org/. This particular app provides security, versioning, submissions, etc. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Text entry by DTMF
At 06:51 AM 11/11/2003, you wrote: Ummm.. kind of. I mean, it says Enter the first 3 digits of the persons last night and you enter them via the keypad, it then searches for the names, and says, Calling so-and-so. I think I've seen this feature on a phone system I called once, but I can't remember exactly how it worked. I'm pretty sure you just entered the persons last name in by digits. Yes, that is precisely how it works on our existing KSU. Each extension has the first and last name as text in a database somewhere and you can get a list of matching names by entering up to 8 chars of the person's last name, then pound. i.e. for SMITH you'd enter 76484#. That'd match Smith, Smithers, Goldsmith, etc. Seems pretty easy to do, too, building up a list of potential strings and then comparing them to the database, or the reverse even. That is indeed how the existing * application works. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Text entry by DTMF
At 10:33 AM 11/10/2003, you wrote: Steve Underwood [EMAIL PROTECTED] wrote in news:3FAE487A.7000508 @coppice.org: Hi, I've kind of ported a DTMF text extry method I wrote some time ago for Dialogic. It is now a semi-working Asterisk app. I've still got to clean up some stuff in how Festival is used to read back what is entered, and then I think it should be OK. Any useful input much appreciated. Not sure if it's useful, but could this be used so that you could create a database of names and extensions, then use this to enter in the first 3 digits of the name or whatever the case may be? Could be useful... I believe this feature already exists (undocumented, of course). It uses the voicemail config file to match mailboxes against names and reads the say your name audio clip out of the user's voicemail box. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
At 01:08 PM 11/8/2003, you wrote: So what do people think about adding the call rate to the CDR structure?? Sounds great, but there's one problem. How does asterisk know what the current rate in effect is? I can think of several ways to do this, but they all involve some fairly significant C coding. or is their a better way to do this. I just use a perl script. The CDR record tells me the number dialed and the channel that handled the far end of the call. Based on that I can tell which provider was used, what number was called, how long the call lasted and when the call was made. That's more than enough information to calculate rates. Parse that file once an hour/day and there you go. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
At 07:01 AM 11/7/2003, you wrote: Hi ! Now I can hear nice mp3 through my phone... Great :P And many thanks for your posts. Now it's working fine... hmmm almost !!! In fact, I m using DialenMP3.agi. It's a real nice agi script... For those which would not know, you will find it here : http://asterisk.gnuinter.net/files/digium/asterisk-ng/agi/ What the agi script does, it's EXEC MP3Player \$key\\n; in which the key is the mp3 to play. Well the things weird, really weird, it's that the process, launched by the agi script, is never killed !!! Any ideas to fix that ?!? You need to... 1) Set up a signal handler to handle the case when the mp3 player dies before you are ready (SIGCHILD) 2) Kill the mp3 player before you exit the AGI script (kill procid) I'll help you with this if you need it, it's really not all that hard. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
At 09:20 AM 11/7/2003, you wrote: I though to it also, but really I don't know how can I get the pid of a process ran by asterisk. I mean, the only think I do it's :print EXEC MP3Player \$key\\n; Then asterisk take the hand with mp3player applications that will launch mpg123, etc... You're right, I thought you were using EAGI... That app seems to work fine for me. MPG123 does seem to be giving asterisk trouble, generally. I tried a few different changes, but nothing resulted in a fix. Is MPG123 maybe not responding to a SIGKILL properly? --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Putting call on hold
At 03:12 PM 11/7/2003, you wrote: Is there a way to put a call on hold and play music on hold with out using the park app? There is a MusicOnHold extension that is like park, except that you can never take them off hold. Most SIP phones also have the ability to put a call on hold and tell * to start playing music. The SNOM 200 does this in version 1.16, but not in version 2.0beta (so far, I haven't tested it recently), and the Xten softphone does as well. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
At 04:00 PM 11/5/2003, you wrote: We are Mediatrix's US distributor and have used them with Asterisk in our lab and have had several resellers purchase them to use with Asterisk. They seem to work well with Asterisk, but I have to agree that the configuration leaves a lot to be desired. Their SIP units use SNMP exclusively and the way that their MIB is arranged, it is a little like configuring a Windows PC via the registry editor. Thankfully their are only 6 or so settings that need to be changed from the default to get it working so once you know where everything is, it is not that bad. How well does the echo cancel, hangup detect, etc work? We're experiencing some nasty echos with *, and I need to know whether we can expect better or worse when we move to a gateway. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intel Performance Primitives
Hey all, For those of you who are really worried about asterisk performance, I thought I might alert you to a toy you might play around with. The Intel Performance Primitives contain a number of optimized functions for use in digital signal processing that could help with echo cancellation, codec transformations, etc. I don't have any idea how useful this would be in Real Life (actual performance gain, license compatibility, etc), but there you go... http://www.intel.com/software/products/ipp/ipp30/ --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk: Reloaded
At 10:23 PM 10/31/2003, Bryan Nolen wrote: System execute asterisk -rx reload ? Yes, correct. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Saturday, 1 November 2003 5:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk: Reloaded Hello, Pretend I had a Perl script that did something to an Asterisk conf file... How can I [from Perl] ask Asterisk to reload? ;) Ben __ Benjamin Wakefield [EMAIL PROTECTED] http://www.dcsi.net.au/ DCSI - We do Internet. 64 Queen Street Warragul, VIC 3820 AU Ph: (+61) 1300 665 575 Fx: (+61) 1300 556 595 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound PSTN Calls
At 07:42 AM 11/1/2003, you wrote: Hi All, Is it possible to show which line a call has come in on in *. Yes, absolutely. In asterisk each line is a channel. The channel information is VITAL to the call and is available (and used) everywhere in asterisk. Channels look like this: ZAP/1-1, which means Zaptel card, line 1, call 1. My scenario is 8 incoming lines, 6 lines are trunked to one number and the other 2 are individual lines. I assume you mean that they are six analog lines set up with a rollover. If you use eight FXO cards, then each line is a separate asterisk channel. Configure each channel with a different default context in the zaptel.conf file. I believe the same is true if you use a channel bank, in which case each T1 will be 23 channels (1-23, 24-47, etc) context=default signalling=fxs_ks channel=1 channel=2 context=notdefault channel=3 The total system for a start will consist of, 8 PSTN (analogue lines and 25 extensions, with the possibility of expansion for remote SIP phones globally). If you use a VoIP gateway, then you need to configure the gateway with a different user for each group of lines. I can't help you with this, as it depends on the gateway, but I'm told it's possible (and I'll be doing it myself soon). --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
At 08:54 AM 11/1/2003, you wrote: P.S.: Looks like I have to post this once a day now. You should post this (or I'll do it for you, with permission, as I already have an account) on the Asterisk wiki at www.voip-info.org. You might still have to post, but at least it will be out there... Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on remote end when using NuFone
At 10:42 AM 10/31/2003, you wrote: I have the same problem and it was solved setting: # Uncomment for aggressive residual echo supression under # MARK2 echo canceller # KFLAGS+=-DAGGRESSIVE_SUPPRESSOR This creates a very nasty click when I talk into the SNOM (but no long echo!). It's like having a conversation with a compulsive interjector who never finishes his sentences :) Do you have this problem? If so, do you recall how you solved it? Thanks, --Ernest in the makefile of zaptel and recompiling. miklos - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 4:21 PM Subject: [Asterisk-Users] Echo on remote end when using NuFone I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a separate issue, I am hearing a bad echo when using my Digium X100P to connect to the PSTN. I've tried tweaking the tx/rx gain to no real effect. I've also tried changing the volume on the SNOM phone, changing the codec to g711u, and decreasing the packet size. Any other things to try? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
At 05:15 PM 11/1/2003, you wrote: Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... Asterisk works VERY well under RH9. Be sure to install kernel-sources and keep them up-to-date along with the rest of the system. BTW, where would I find a useful FM? Um, yeah. (1) Search the mailing list archives. (2) Check out http://www.voip-info.org/tiki-index.php?page=Asterisk. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quick Question
At 05:03 PM 11/1/2003, you wrote: Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on remote end when using NuFone
I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a separate issue, I am hearing a bad echo when using my Digium X100P to connect to the PSTN. I've tried tweaking the tx/rx gain to no real effect. I've also tried changing the volume on the SNOM phone, changing the codec to g711u, and decreasing the packet size. Any other things to try? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Video
Is anyone using Asterisk as the gatekeeper/proxy for videophone calls? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH Mixing tool
Does anyone know of a command-line tool that I can use to mix my own MOH tracks? Specifically, I want to be able to do this: 1) Record a Your call is valuable to us... advertisement 2) Specify a number of song files to be played randomly/in sequence/whatever 3) Insert or overlay the advertisement every n seconds I would like this to be done live (via a configuration file) if possible, but I'd be just as happy to build a 60 minute mp3 file if it gets the job done. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Luxon Communications
Has anyone successfully used a Luxon VoIP gateway with *? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newb - want to create a Dialpad like system
Check out the software at http://www.xten.com/. Their XTen-Web and XTen.NET products may help you out. Allowing people to dial a landline is actually quite simple, and can definately be done with Asterisk. --Ernest At 08:00 AM 10/22/2003, you wrote: Resending this. Any help appreciated. Thanks, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 6:36 PM Subject: [Asterisk-Users] newb - want to create a Dialpad like system Hi all, i am planning to create Dialpad like system for fun. i want to build it in such a way that one can use either web based app or GnoPhone / MsnMessenger to connect to my server and then dial a land line. i did a search on the archives but couldnt find any good pointers. i would appreciate if someone could let me know whether this possible to create this app using asteriskand send me some pointers. i hv the following RH 7.3 Dialogic D41 thanks a lot, -Balaji Do you Yahoo!? The New Yahoo! Shopping - with improved product search Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] SNOM 200 beta build + MOH
I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Send to VoiceMail button
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
Here is what I use with a SNOM 200...
exten = _2,1,Voicemail2(u${EXTEN:1})
Then, configure any of the SNOM's redirect options (automatic or using key
mapping) to redirect to your extension with a 2 prepended. Works perfectly.
--Ernest
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[Asterisk-Users] Audiocodes gateway and asterisk
Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with voicemail, etc? My experience with loopstart on the X100P is that voicemail never stops and I end up with two-hour long messages. Any comments or suggestions? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is somthing broken?
At 12:33 PM 9/29/2003, you wrote: Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram) Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). That solved the problem. --Ernest Mark On Mon, 29 Sep 2003, WipeOut wrote: WipeOut wrote: Hi, I updated my live server yesterday(after testing on my Dev server first, all works on the Dev server).. Here is the setup.. SIP_UA---[NAT]---Asterisk1---PSTN(chan_capi) The SIP_UA is able to recieve calls from the server with no problems.. Initiated from the PSTN or my Dev Asterisk box which is connected to Asterisk1 with IAX.. When the SIP_UA tries to make calls out via the PSTN or to Voicemail on Asterisk1 or another extention there is no sound.. The definition in sip.conf is fairly standard(included below).. This config has been working fine for months.. the last update was about 1 month ago so sometime between then and now it seems that SIP has changed and so stopped working.. Hopefully this can be solved quickly becasue it is a problem.. Later.. Definition from sip.conf [2014] context=users type=friend secret=magic nat=yes canreinvite=no dtmfmode=info ; Grandstream host=dynamic mailbox=2014; Mailbox for message waiting indicator ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Looks like there is a problem with SIP, I rolled back to Thurday last weeks CVS and the SIP UA behind NAT is now working.. I will be interested to see if anyone else has this problem when updating to the current CVS.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speaking of Outlook
Does anybody have a reasonable solution for an Outlook MAPI plugin that works with asterisk? At very least, I would like Asterisk to push incoming call information to the computer, which should then open an Outlook form, launch a web browser, etc. Beyond that, it would be cool to have Outlook initiate outgoing calls. Shouldn't be too difficult, and I know some of you are working along similar lines. Outlook itself is only an example. What I'm looking for is a simple incoming call, launch a browser with the callerid in the query string type app for windows. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me
At 06:48 PM 9/16/2003, you wrote:
cell phone into the call (or my office number, etc.) I understand the
selected numbers part of it, but not how to get it to use the three way. If
I send it to Nufone first, I'm paying for a call to a local number (my
cell) that I don't need to.
This should work...
[default]
exten = s,1,Dial(Zap/3,20,t) ; This is your desk phone
exten = s,2,Dial(Zap/2/1234567,20,t) ; This is your secondary POTS line
calling your office
exten = s,3,Dial(Zap/2/3217654,20,t) ; This is your secondary POTS line
calling your cell phone
; I've never tried this one coming up, but I think it's worth a shot as it
works just fine for local extensions
exten = s,4,Dial(Zap/2/3217654Zap/3/3217654,20,t) ; This is your
secondary and tertiary POTS lines calling your cell phone anbd office
As long as none of these lines go to voicemail, they should fail over
properly in order. You can also make it more complicated with time-based
includes and gotos.
--Ernest
At 09:57 AM 9/16/2003 -0700, Ernest W. Lessenger wrote:
At 11:22 PM 9/14/2003, you wrote:
First -- Thanks to everyone who offered their help and tips on getting my
Cisco 7960 working with Asterisk -- this is great stuff.
Does anyone have any examples of Follow Me or other call forwarding with
a single PSTN interface? Or a pointer on what I need to read to figure it
out?
Is this what you need? Basically, the local trunk and the Nufone trunk
fail over to each other. So, if you have a forward set up and transfer to
a non-local extension, the call will go out even if the original incoming
call was made on the PSTN line.
[trunklocal]
exten = _NXX,1,Dial(${TRUNK}/${EXTEN})
exten = _NXX,102,Dial(${NUFONE}/1${AREACODE}${EXTEN})
exten = _NXX,203,Congestion()
[iaxprovider]
exten = _1NXXNXX,1,Dial(${NUFONE}/${EXTEN})
exten = _1NXXNXX,102,Dial(${TRUNK})
exten = _1NXXNXX,203,Congestion()
exten = _011.,1,Dial(${NUFONE}/${EXTEN})
exten = _011.,102,Congestion()
exten = _1011.,1,Dial(${NUFONE}/${EXTEN})
exten = _1011.,102,Congestion()
--Ernest
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Re: [Asterisk-Users] Voicemail feature
At 07:52 PM 9/14/2003, you wrote: Any chance of getting this feature added (preferrable as another option on each mailbox setting in voicemail.conf (after the pager # maybe))? I know it could be hacked, but I am trying to avoid those type of improvements. :) Asterisk already has an outgoing call queue. You could easily modify the Voicemail2.c file to write a queue record with the settings you want. Wouldn't even be a hack :) In fact, it should be even simpler than the original sendpage function. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905
Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SOME QUESTIONES (LOG, MySQL, Extensions)
At 01:01 PM 9/15/2003, you wrote: Hi all. I have some questions: 1) Is there a way to get a full log of the calls (incoming and outgoing) You can get the Call Detail Records which show you the incoming user and the initial dialed extension as well as the date, time, etc. They don't record the complete call flow. 2) How is the intregation of Mysql and Asterisk. At witch Aplicattions. Functional, but I've heard mixed reports. It's my understanding that MySQL support works for logging (cdr-mysql) and voicemail2. a) I have a Support Call Center. Almost all the time all the extensions are busy, and some calls at hold. Is there a way that when some one hang off, can take the next call in the hold. At this moment, i have that if all are busy, send the call to hold. And in 1 minute tray again to call any extension. This is what the Queues are for. Search the mailing list or look at the demos. Basically, incoming calls are put into a queue until one of the currently logged-in operators (or one of the defined extensions) picks it up. Then, the other extensions keep ringing until the queue is empty or until the timeout expires and the callers get bounced. b) Is ther any way.. that other person can take the ringing call of other extension at his phone?... Example: There is a ringing call at extension 100 which is alone and i'm at extension 150, So and i want to take it? so i dial some number and i take it... How to do this? I'm told this can be done, but I don't know how myself. What you can do is have several extensions ring at once, as though they were a single extension. This can be done with groups or by using the extext syntax in extensions.conf. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite + Asterisk
At 11:24 AM 9/12/2003, you wrote: Anyone configured X-Lite with asterisk? Any help on config, both on X-Lite and asterisk? Yep. Here's an acceptable SIP.CONF entry... [551212] type=friend username=551212 secret=12345678 host=dynamic qualify=1000 nat=1 mailbox=551212 context=default The X-Lite setup would be: Username: 551212 Auth Username: 551212 Password: 12345678 SIP Proxy: Whatever your SIP proxy is All other settings are default Important: If you have two NICs, make sure that X-Lite binds to the correct one. This can be configured in the Network menu. Also, I had some trouble with X-Lite choosing the wrong audio card in my system. I managed to have SJPhone working OK with * but not X-Lite. Oddly enough, I've had the exact opposite problem. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request for best practices
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to direct-dial each other's extensions. We want this to
work like a real centrex, in that seven-digit numbers should try (1)
local VoIP extensions, and then (2) local PSTN numbers. Ten-digit
numbers should dial (1) long-distance VoIP extensions, and then (2)
long-distance PSTN numbers.
Here's my plan so far, does anyone have a better way? Will Goto() work the
way I expect it to (i.e. will the extension I specify be pattern matched)?
==Extensions.conf==
[area555]
exten = _NXXNXXX, 1, Goto(extensions,555${EXTEN})
include = extensions
[area666]
exten = _NXXNXXX,1, Goto(extensions,666${EXTEN})
include = extensions
[extensions]
exten = 5551234567, 1, Macro(stdexten, 1234, SIP/user1)
exten = 6661234567, 1, Macro(stdexten, 1235, SIP/user2)
include = longdistance
[longdistance]
exten = _NXXNXX, 1, Dial(${Nufone},${ARG1})
exten = _NXXNXX, 2, Congestion()
[macro-stdexten]
... as in demo ...
==Sip.conf===
[user1]
...
context = area555
[user2]
...
context = area666
Thanks,
--Ernest
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Re: [Asterisk-Users] Having problems with S100U
My S100U also gets quite warm. I haven't had any trouble with it though. --Ernest At 01:31 PM 9/10/2003, you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jason A. Pattie wrote: | I have another observation for you. Do your S100U's get warm? I've | left this one plugged into the USB port and the red LED is lit on the | unit even though the wcusb driver is not loaded at this time. I noticed | that the unit gets warm whenever the driver is loaded. I've left it | running now for around 2 - 3 hours and it is still quite cool to the | touch. I will install the wcusb driver and see if the unit heats up. | Maybe all these issues that people have with these devices are heat | related? It's now almost 2 hours later and the device has been spitting out the error messages documented in my earlier posts. The S100U is it seems now at a stabilized heat level. It is not uncomfortably warm but feels like a transformer you might have plugged into the wall. It felt warmer fairly soon after I modprobe'd the wcusb driver, around 20 minutes. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/X4o1uYsUrHkpYtARAlniAJ9lQp7HsgIEQUtkvEB+vrUPP2l21QCfWrGU kS1G84KatsfXrK07YEC8LrI= =F5Ds -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has the allow=all function changed in sip.conf?
At 11:26 AM 9/9/2003 -0600, you wrote: What is the general consciences for the allow=all statement? Should it be used, should it be specific towards those codecs supported, or removed? My understanding is that you MUST have at least one allow and one deny, or none at all. Just having one or the other causes problems. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
At 02:38 PM 9/9/2003 -0500, you wrote: That would be reinvite= and canreinvite= in the user entry for each SIP endpoint. Asterisk will allow the endpoints to talk directly to each other if both those settings are = yes (the default, I think) AND both endpoints use the same protocol (SIP) AND the same codec. So Asterisk will allow it... and if I set both to no, asterisk would act as a true proxy, using the most bandwidth efficient codec available for each leg of the call (i.e. GSM for x-lite and g.729 for Cisco et al)? Thanks, --Ernest On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote: I have seen this asked in the archives several times, but do not see a definitive answer anywhere. Is there a way to tell the Asterisk to act like a normal SIP Proxy, handling only the SIP messages, and letting the RTP go point-to-point? - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 09, 2003 1:40 PM Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet On Mon, 8 Sep 2003, Jim Mercer wrote: Can we bribe you? :) sure, pay my rent for 3 months and give me a 50 plasma TV to play in the background. Is that all? That sounds rather cheap, compared to the things direction that I'd have to go if I wanted to stick to the cisci CM route, with licenses for every endpoint that I want to connect. Realistically... I just can not comprehend how to get stuff to work correctly with Linux. I used to be a Linux nut years ago, but once I found FreeBSD with it's ports collection, I wondered why anyone ever bothered with Linux and it's completely messed up software install requirements. Right now, under RedHat 9.0, I have * running, but no hardware, and I can't figure out how to get h.323 operational so I can talk to my cisco gateway with the PRI interface... I'm only guessing that FreeBSD would be much easier for non-programmers like myself. -Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP LD carrier
At 07:57 PM 9/9/2003 -0400, you wrote: Travis Johnson wrote: I've called NuFone and was not impressed by their voicemail answering system (choppy) and was unable to even leave a message before the phone call was disconnected (in the middle of the recording). So your going to judge our system by making one phone call into my home asterisk system that runs on a fully saturated ADSL connection. One must take first impressions from something, and if that happens to be your answering machine... Typical, Unfortunately, yes. Seriously folks, NuFone has very reasonable prices, easy setup and decent tech support (or so I've heard - I haven't had any problems with them so far). I urge you to give NuFone a try, if only because (one of?) their big cheeses reads the * mailing list :) I recently checked out the competition, and here's what I found: NuFone: cool first impression, good second impression, good implementation Deltathree: good first impression, poor second impression Net2Phone: good first impression, excellent second impression, cool third impression Vonage: cool first impression, cool second impression In all cases I was evaluating for price, compatibility with * and ease of use/ease of billing. Of course, I had specific uses in minds, specific hardware, and my own company's cost structure in mind, so your mileage may vary. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
At 07:52 PM 9/8/2003 +0200, you wrote: Is there a way to configure Hylafax or sth one modem behind an ATA-186 to email faxes to different adresses depending on the called number ? I've looked into this myself, and I think the answer is yes, with some minor code changes. My thought is that you would use a separate HylaFax server with six modems in it, and add two Digium FXS cards to the * server. Configure * to send the faxes out the correct FXS port for each company, and configure hylafax to queue the faxes to a different folder for each line. User interface and notification are left as an exercise to the reader, as is the actual hylafax configuration :) The major downside to the above is all the POTS lines you have to run, and the waste of ports. An alternative would be to use only one or two POTS, and have * set the CallerID for each company. Then, have Hylafax queue the incoming faxes based on CallerID. The disadvantage to this is, of course, that you lose any real CallerID information. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension.conf and SIP phones.
At 03:17 PM 9/8/2003 -0400, you wrote:
We would like to setup in house SIP phones with numbered extensions for
demonstration purposes.
What is the syntax to associate a extension with SIP phone?
exten = 1234,1,Dial(SIP/username)
Does the Dial application have a SIP specific entry for example:
Dial,SIP/SIPphone/s|15
See above.
When I call from one extension to another I get User is on the phone.
The SIP phone needs to have logged-in in order for you to dial their
extension. The usual reason for not being able to dial a SIP phone is that
(a) they did not actually log-in correctly, or (b) you've rebooted asterisk
since they last logged in.
sip.conf
[user]
callerid=User Name
type=friend
nat=no
username=user
secret=password
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
defaultip=10.1.1.53
dtmfmode=rfc2833
mailbox=1115
voicemail.conf
1115 = 1234,User Name,[EMAIL PROTECTED]
extension.conf
exten = 1115,Macro(stdexten,1115,${user})
exten = 1115,1,Dial,SIP/user/s|15
exten = 1115,2,Voicemail,u1115
exten = 1115,102,Voicemail,b1115
exten = user,1,Goto(1115|1)
Try this:
sip.conf
[user]
callerid=User Name
type=friend
nat=no
username=user
secret=password
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
defaultip=10.1.1.53
dtmfmode=rfc2833
mailbox=1115
context=default
extension.conf
[default]
exten = 1115,1,Macro(stdexten,1115,SIP/user)
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,tT)
exten = s,2,Voicemail2(u${ARG1})
exten = s,3,Goto(default,s,1)
exten = s,102,Voicemail2(b${ARG1})
exten = s,103,Goto(default,s,1)
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[Asterisk-Users] Limiting the number of SIP/IAX lines
Is it possible to limit the number of lines provided by a given SIP/IAX connection? For example: I want to limit SIP extensions to only a single incoming line, even the phone itself can handle three. Or, I might want to prevent extensions from making more than one outgoing call at a time. Or, I might want to protect my bandwidth/call quality by limiting outgoing calls through NuFone to only three calls at a time. Any thoughts? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR not recording SIP username
In reading the source for the CDR_CSV module, I understand that it should use the SIP username as the account code for calls made from SIP devices. However, nothing is being recorded in the csv file for that field (i.e. blank value). Is there any way to add an account code for SIP users? I can always identify the SIP user from the channel identifier, but it would be cleaner to use an account code. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR not recording SIP username
At 12:21 PM 9/5/2003 -0700, you wrote: In reading the source for the CDR_CSV module, I understand that it should use the SIP username as the account code for calls made from SIP devices. However, nothing is being recorded in the csv file for that field (i.e. blank value). Is there any way to add an account code for SIP users? I can always identify the SIP user from the channel identifier, but it would be cleaner to use an account code. Hah! Undocumented (at least in the documentation I have) feature is that you can use the accountcode statement in sip.conf. Cool. Thanks anyway, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
