Re: [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!
Cameron Hissey wrote: now that you have some background, I am having no luck installing these two cards - i have already confirmed they are on their own IRQ etc, and if i run genzaptelconf, they are coming up fine and with no errors, and i can see them in the zapata-auto.conf file, however i cannot see the E1 card when typing ztcfg -, and i cannot get a link light on the E1 card either. This looks like something other than straight Asterisk. So it may be different, but I'll try to help you the best that I can. First, stock Asterisk/Zaptel needs 2 files to be correct for your E1 card and TDM card to work. You need a valid, correct, and complete zaptel.conf, usually in /etc (although it can be different depending on your config). zaptel-auto.conf may be #included in your zaptel.conf, or it may need to be renamed. I honestly have no idea. For all of my systems, it would have to be renamed, as I try not to #include in anything except my sip.conf, extensions.conf, and iax.conf. But that's just me. Secondly, Asterisk needs a valid, correct, and complete zapata.conf, usually in /etc/asterisk, although it can vary depending on how things were configured when the system was set up. If you have both of these files in the right place, and with the right stuff, make sure you reload zaptel and asterisk. I know I don't have to, but when I make big changes after hours, I tend to reboot. Especially when I'm tired and can easily make mistakes. But that's a matter of personal preference. Hope that helps. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Speakerphone
Doug wrote: At 08:38 11/12/2007, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Excellent speakerphone. Extremely cumbersome to configure. I agree about the speakerphone, and disagree with the claim about configuration. The XML is extremely to generate through scripts, and once the framework is built, I find it to be far simpler to manage the deployment than other IP phones. Of course, YMMV. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on polycom 501
Jerry Geis wrote: I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. Google is your friend: http://www.google.com/search?hl=enq=polycom+daylightSavingsbtnG=Google+Search Top hit fixed it for us. For the archives, in case the top hit is no longer the top hit (i.e. I'm feeling unlucky), the result is: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/996/10627_f.SAL_PUBLIC_1_2.html and contains this: Description *Technical Bulletin 17803* *SoundPoint® and SoundStation® IP phones require configuration changes due to changes in daylight saving time (DST) dates.* *This information applies to:* • SoundPoint IP 300, 301, 430, 500, 501, 600, 601, 650 desktop phones and SoundStation IP 4000 conference phones *Note:* This information applies to the SoundPoint IP 650, where software releases exists to support the IP 650 (see Software Release Notes for platform compatibility). *SYMPTOMS* Beginning in 2006, all parts of the State of Indiana will observe Daylight Saving Time along with the rest of the United States. The majority of the state will now be in Eastern Time, but there are several counties near Chicago that will remain in Central Time. The United States Congress passed a law in 2005 that changes the dates when US Daylight Saving Time begins and ends starting in 2007. This affects all US states except Hawaii and Arizona, which do not observe DST. As of this writing, the Canadian provinces of Ontario, Manitoba, Quebec, Prince Edward Island, New Brunswick, Alberta, the Yukon and Northwest Territories, British Columbia, and Nova Scotia have indicated that they will adopt the same changes, and other provinces and territories will continue with current procedures. *RESOLUTION* With respect to the State of Indiana, no special configuration is required to support this change, but any special configuration that had been made previously to exempt phones in Indiana from DST needs to be removed. *Note:* The following change cannot be safely made until 6 November 2006, as the old settings are required until that date for 2006 DST to be calculated correctly. To configure phones for the new DST rules, the SNTP configuration section from sip.cfg (or ipmid.cfg in older versions) needs to change as follows: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 *Note:* /These changes should be made in the *ipmid.cfg* configuration file for SoundPoint IP phones running the MGCP application./ *Note:* /There is an error in the display and setting of the DST ‘Start/Stop Day Of Week’ if the web server interface is used to set the DST rules. When the start date is set to 1 (Sunday) in the *sip.cfg* or *ipmid.cfg* file, it is displayed as Monday in the web server interface. If you use the web server interface to set the DST start/stop dates, select Monday to obtain a setting of Sunday. This discrepancy will be fixed in a future software release./ *STATUS* *Polycom recommends that this configuration change be made.* ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring group containing external 10-digit numbers
Ryan Stille wrote: Don Pobanz wrote: Ryan Stille wrote: I have a ring group setup that I'd like to ring a bunch of local extensions, plus a few outside lines. I want recipients to confirm the call by pressing 1 before they are connected. But when I add in an external number to this ring group (such as 5551212#), none of my internal extensions ring any more. Any ideas? You did not indicate how you were connected to the outside world but if you are connected through fxo channels, the line would be answered by the phone company very quickly. Once 1 phone is answered, I believe all the others would stop ringing. Could this be the issue? Don Pobanz Sorry, I am connected to the outside world via a SIP trunk. I am letting all the phones ring for this testing - no one is answering. But for some reason when I add that external number, none of the internal extensions ring. -Ryan I think Don was on the right track. First, check your dial plan for any stray Answer directives where you dial out. And keep in mind that some SIP providers Answer the call first thing. My advice: think about GrandCentral and how it works: Press 1 to accept the call, press 2 to ... whatever. Or find a SIP provider that does not answer the line immediately. Many SIP providers will give you a few cents to play with before you buy an account. Just a few calls should be ample to figure out how it's doing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote: Joseph Begumisa wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I have a TE410P that does it. It may not be wise, but I just ignore the orange blinking LCD display (or light, depending on the model). I did try reseating the card, and it works for a few weeks, and then goes back to the same old thing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: Internal LAN echo problem]
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote: Let me screw this thread up by top posting now. Could echo be caused by late packets if jitterbuffer is on or something or would that just cause lag? Thanks, Steve So, does this qualify as an in-line reply, or a top post? Maybe it's a medium post ;) If both calls were in the LAN, chances are good that the phones will have re-invited to go around the SIP server. If that's the case, then it shouldn't be a problem. Now, if dial options, recording, or SIP settings prevent reinvites, then this might be part of the problem. kevin bergner wrote: On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Jonn Taylor wrote: Eric ManxPower Wieling wrote: Any echo you hear on pure IP calls is caused by the endpoint phone. You cannot do ANYTHING about it on Asterisk. Jonn Taylor wrote: Any ideas ? Jonn Original Message Subject:[asterisk-users] Internal LAN echo problem Date: Wed, 24 Oct 2007 08:34:32 -0500 From: Jonn R Taylor [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi all, I have an internal echo problem on my LAN only. I replaced the LAN switch with a new linksys 2024 with QOS and seemed to help but not fix the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are cheap that are known for echo problem in the handset. I have one remote user that never has a problem. I have a remote test server at home connect via IAX with no problems, also a PAP2 with no problem. External faxing from the rest of the world via our voip provider is working great. One strange thing that I noticed is that we can not fax to our iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure why either. That does not make sense. I can any one of these ata's or phones and connect them to the public ip side and they work fine. It can make sense or not make sense, but you cannot have echo on a pure VoIP call unless the endpoints introduce it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users i have seen this when the headset volume is too high and simply lowering the volume addressed the problem as others have said an echo is simply not possible ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: Internal LAN echo problem]
On 10/24/07, David Gomillion [EMAIL PROTECTED] wrote:
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
Let me screw this thread up by top posting now.
Could echo be caused by late packets if jitterbuffer is on or something
or would that just cause lag?
Thanks,
Steve
So, does this qualify as an in-line reply, or a top post? Maybe it's a
medium post ;)
If both calls were in the LAN, chances are good that the phones will have
re-invited to go around the SIP server. If that's the case, then it
shouldn't be a problem.
Now, if dial options, recording, or SIP settings prevent reinvites, then
this might be part of the problem.
Sorry, I need to clarify my own post. By part of the problem, I mean
magnifying the effect. The real problem is the handset leaking, probably too
much sidetone.
Anyway, the more the delay, the more noticeable this echo will usually be.
kevin bergner wrote:
On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Jonn Taylor wrote:
Eric ManxPower Wieling wrote:
Any echo you hear on pure IP calls is caused by the endpoint
phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject:[asterisk-users] Internal LAN echo problem
Date: Wed, 24 Oct 2007 08:34:32 -0500
From: Jonn R Taylor [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Hi all,
I have an internal echo problem on my LAN only. I replaced the LAN
switch with a new linksys 2024 with QOS and seemed to help but not
fix
the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
Asterisk 1.2.24 /FreePBX, 2-NIC cards, one with a public ip and
one with
an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that
bt's are
cheap that are known for echo problem in the handset. I have one
remote
user that never has a problem. I have a remote test server at home
connect via IAX with no problems, also a PAP2 with no problem.
External
faxing from the rest of the world via our voip provider is working
great. One strange thing that I noticed is that we can not fax to
our
iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax.
Not sure
why either.
That does not make sense. I can any one of these ata's or phones and
connect them to the public ip side and they work fine.
It can make sense or not make sense, but you cannot have echo on a
pure
VoIP call unless the endpoints introduce it.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
i have seen this when the headset volume is too high and simply
lowering the volume addressed the problem
as others have said an echo is simply not possible
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
On 10/19/07, Kevin Smith [EMAIL PROTECTED] wrote: Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin We have a medium-sized network (120 polycoms of various persuasions, and 80 workstations), and we haven't had any real problems with phones ruining QoS. We have the phones on separate VLANs than the workstations. Actually, every switch has 4 VLANs defined: 2 voice, 2 data, so no VLAN has more than about 12 devices (about because sometimes we have to put a pocket switch in a room where the people want to add yet another computer). The echo from SIP to SIP with people using cheap headsets has affected us far more than any problems with PCs trying to suck the bandwidth. If I remember correctly, recent firmwares on the Polycom phones pretty much do the right thing, giving priority to the phone traffic. To summarize: works OK for us. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk incomming call huntgrroup
On 10/17/07, satish patel [EMAIL PROTECTED] wrote: Dear all I want to configure Huntgroup for my company like i call on 1100 extention i will transfer to avalible group extention i got some document on voip-info website but this is not working for me http://www.voip-info.org/wiki/view/Asterisk+Hunting+Groups+for+incoming+calls Why not just use a queue? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to
do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of fancy things like call
queues, parking, forwarding, followme, voicemail to email etc. etc. how
practical is it to type all this in by hand making sure to get ever single
space, ., ,, {}, [] etc. exactly right which NEVER happens. So
then
you have to spend more time debugging the conf files.
Even with a bunch of pre-made templates it seems like an awful lot of
unnecessary heavy lifting when a GUI can make it so much easier and
efficient.
You're welcome to do it however you like. But please don't suggest that
using a GUI will make things more efficient. Someone with experience
scripting can easily write a system to generate a well-formed, valid .conf
file, with appropriate comments. I, for one, have done this.
The reason many seasoned Asterisk admins prefer using the .conf files
instead of using a GUI is that no GUI can possibly conceive of every way to
do something. So, at some point, if your PBX does anything interesting,
you're going to have to integrate your changes with what the GUI generated.
And not let the GUI stomp on the changes. But make sure everything will be
in contexts that can access what it should, and not access what it
shouldn't.
Now, as far as how practical it is to create the dialplan by hand, I can
tell you that it only takes about 2-3 minutes to full configure such a
simple PBX as you described. Most GUI systems take far longer than that to
install, much less configure. Also, I can more easily manage systems
remotely via SSH than through many of the GUIs out there.
So, as I said before, do whatever works best for you. But please don't
insinuate that editing configuration files cannot be a good idea.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On 10/9/07, Matt [EMAIL PROTECTED] wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm Fascinating. Not really. Anyway, how is this related to Asterisk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing contexts on the fly
On 9/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Sep 28, 2007 at 05:28:21PM +0100, Ade Vickers wrote: Hi folks, I've been playing around with an Asterisk server in my office for a few weeks now, and I've got it pretty much nailed down the way I want it, which is nice. One of the features I'm using is the ability to switch different contexts in out of the dialplan on a schedule. So, for example, I've got the official tel number ringing my desk phone between 9.00-17.30 mon-fri; and out of those hours any caller gets a recorded message + sent to voicemail. However, I'm quite often working later than 17.30, and would quite like to be able to easily flick a switch which tells Asterisk that, actually, I'm here in the office, and I'd quite like to receive calls. Currently, I have to alter dialplans.conf, comment out a couple of lines uncomment another; save then re-load the dialplan. I'm guessing I've got 3 options open to me: 1) Convert from using the various .conf files, to using a realtime config, then write a small front-end to the DB so I can access the settings from a simple switch on my Windows desktop 2) Write some kind of script which I can execute on the Asterisk box which makes the same changes I'm currently making manually 3) Some other option I've not thought of... 4) Use a condional dialplan. e.g GotoIfTime or other uses of GotoIf . Now, add a flag that allows your calls to be routed as either: 1. Default - route according to the schedule 2. Open - give me the calls, to heck with the time 3. Closed - leave me alone. Yes, I know what time it is, but I don't care. Put this before the GotoIfTime stuff, and it can override however you'd like. We did this, but added a few fancy things, like ClosedForHurricane mode. It allows us to record a message as to which dates patients have been rescheduled to, says the time of the last update, and a few other goodies. Have fun with it. You can do just about anything you can dream of. Except solve the halting problem. Ah well... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. Create a new MOH class with one large file consisting of every message you want heard, in the order you want them heard. Since there will be only one file, you know which will be first ;) We actually do this with some of our queues, so I know it works. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1/PRI pricing
I know this borders on commercial, so I apologize. I will take this off list as soon as possible. Someone a couple months ago claimed to know how to get PRI or T1 voice circuits significantly cheaper than going through the ILEC. I would appreciate that person contacting me (off-list) at this email address. Thanks, David Gomillion [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
On 9/18/07, Atis Lezdins [EMAIL PROTECTED] wrote: On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu Actually i believe you can do it yourself. X-Lite is windows, right? There are a bunch of programs, allowing to edit internal resources of executable files. So, just grab a resource editor (i prefer XN Resource Editor), open .exe file, edit the menu - disable (and hide) items you want to forbid changing for users, and give them the executable. I'm not certain that X-Lite's executable is not packed/crypted, but editing SJPhone was very successful some time ago. Of course, there's always an option for user - to take another softphone, but whatever softphone you choose - they will have the same chance. I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
On 9/10/07, Ira [EMAIL PROTECTED] wrote: At 02:11 PM 9/10/2007, you wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). dd-wrt or Sveasoft on a Linksys router though I understand there are better choices in routers today. Don't expect too much out of traffic shaping. While it should work nearly perfectly upstream, there's only so much you can do to control the downstream (from your ISP to you). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meridian S1 to Asterisk via T1
On 9/7/07, Michelle Dupuis [EMAIL PROTECTED] wrote: This is going into an emergency response facility...where they currently have a Nortel Option 61 (I think). They want to slowly phase into VoIP. They will need 1000 phone set capacity (assuming full migration). This can be done, and I am a proponent of Asterisk. But I don't think I would recommend it in this situation. Frankly, having a big company like Nortel to blame if/when downtime occurs would be worth the money difference to me! My fear of connecting their PBX directly to the PC (PCI card) is the potential for a PC crash. If that somehow takes down their PBX there will be hell to pay. (If it were just a regular office environment it would be ok). Most media gateways will not work if/when the SIP server (i.e. Asterisk or SER) goes down. Now, some SIP phones can register to multiple hosts, etc, but I'd still push towards a COTS system in this case. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
On 8/23/07, Ed Pastore [EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
On 8/23/07, Ryan M. Colbert [EMAIL PROTECTED] wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? That would be very interesting to see, if you get it working. Last I checked, though, speech-to-text didn't work very well without a very small language to choose from, far smaller than English. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
On 8/21/07, Steve Prior [EMAIL PROTECTED] wrote: Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Does this spark anyone's imagination or ideas to implement? Sparks my imagination thusly: Suppose you have a speaker phone in every room. When the phone is onhook, Asterisk automatically opens up a call to the speaker and places it in the automation context. When you pick up the phone, it grabs a different line, and drops the automation connection. Now, you can address Asterisk by saying, Computer, raise lights 20% and impress all of your trekkie friends when the lights turn up. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as ISDN PRI Proxy
On 8/20/07, Gustavo Felisberto [EMAIL PROTECTED] wrote: I have a costumer with a Siemens PBX installed, and I would like to setup a Asterisk system that would act as a kind of Proxy between the Siemens PBX and the operator network. The current setup is: Siemens PBX 2*PRI - Operator what I want is: Siemens PBX 2*PRI - Asterisk BOX - Operator This is not unusual. For the Siemens PBX the Asterisk Box would be a standard Telephony Operator, and the Asterisk box would either route the calls normally, or would route them via another system via SIP or IAX. I need to know if this is possible, and what kind of hardware do I need on the Asterisk Box to do this. I know I'll need some PRI cards to connect to the Operator, but do those cards allow me to masquerade as a Operator to the Siemens PBX? look at pri_cpe vs pri_net -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbund Route via Extension
On 8/16/07, Nhadie Ramos [EMAIL PROTECTED] wrote: Hi All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source. i'm using freepbx for this. You could easily just put the extensions in different contexts. I'm not sure how freepbx handles this, but Asterisk certainly supports having different handsets in different outgoing contexts. This is probably your best option Another option is to have gotoif statements, based on the callerID of the handset placing the call. Another option, although it's bad in my opinion, is to train everyone in Company A to dial a 9 for outgoing calls, and everyone in Company B to dial 8. That give 0 security, so again, it's bad, but it could work. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experimenting- Sip dialing with Zap
On 8/16/07, John Meksavan [EMAIL PROTECTED] wrote:
line yet. The phone simulator only allow 3 digit dialing. Now, I get this
message on the Asterisk CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-006fd1a0,
Zap/g0/{EXTEN})
in new stack
[Aug 16 20:22:34] WARNING[14292]: app_dial.c:1106 dial_exec_full: Unable
to
create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/200-006fd1a0' status is 'CHANUNAVAIL'
Just a guess here, but it looks like Asterisk is unable to create channel of
type 'Zap', and that everyone is busy/congested at this time.
Now, figure out if you have valid Zap channels defined in both zaptel.confand
zapata.conf. Make sure you have the right signalling, and the right
indications. Stupid question that I don't have to ask, but will anyway, you
do have the TDM400P actually installed, right?
With these basic questions, you may be better served reading a book about
Asterisk, trying what is in there, googling for answers to any questions you
may have, and then asking the list after you have exhausted all other
resources. We're here to help, but I think that these steps may help give
you a better foundation. And we like it when people have at least tried to
figure out solutions.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forking from a dial plan?
On 8/9/07, Andres Paglayan [EMAIL PROTECTED] wrote: On Aug 9, 2007, at 4:32 PM, C F wrote: Local channel should be able to accomplish that, what exactly are you trying to do? this is one of the specific scenarios, I got many others, (for the shortsighted, this can be used in any case you don't want to wait for a chain of steps to complete before triggering another) e.g. 1./ a call goes on queue, 2./ no agents are logged in, 3./ then, an announcement is played, 4./ other options are played 5./ then, the call is transfered to the voice mail, 6./ manger gets a ring with a record,(no-one-is-logged-in-damn-it) Why wouldn't you put #6 as #3, and have it generate a .call file to make a new phone call? forking out of the dial plan, Some commands, like Background, will allow execution to continue. So, it obviously can be done, when needed. But most things that need to happen should happen in order. If it's complicated enough to be done out of order, then AGI is probably more appropriate anyway. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma PRI
(top-posting because Julian did, and I'm too lazy to fix it all) Last I checked, the replacement with the new firmware is only for those who bought the card in the last year (i.e. the card is still under warranty). Those of us who were early adopters cannot enjoy the improvements of the upgraded firmware without buying all new cards. Hopefully, I'm wrong and someone will correct me. On 8/6/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: And what of all the folk that have a v1 card (I've got 2 quad-ports sitting here) ? And can you cross-ship a v1 card for a v2 card replacement ? Julian. Steve Totaro wrote: Kevin P. Fleming wrote: Stephen Bosch wrote: The only way this will ever happen is if Digium completely redesigns the card, which is a long way of saying that you will buy a new card before you have that request filled. That is incorrect. The TE4XXP cards with v2 or later firmware *can* be upgraded in the field, but we have not released an upgrade for those cards that warrants distributing it to end users (there is a v3 but it is only necessary for the PCI Express variants). This may change soon, though, as there is work to produce some improved firmware for all the TE4XXP cards in process right now. Unfortunately cards with v1 firmware will not be able to be upgraded in the field. Steve Totaro: We regularly allow users to cross-ship (advance replacement) cards for firmware upgrades; you should not be required to have your system out of service for any length of time longer than what it takes to swap cards. Who do I contact for this. Is the firmware upgrade still free? My last email to the lady responsible (forgot her name) never replied or her email went into /dev/spam/null. Can you get the ball rolling or give me an email address please? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-recorded first and last names audio database
On 8/4/07, John Vogel [EMAIL PROTECTED] wrote: Hi! My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. Hi, John Brown. Your book is due back at the library. So I need John and Brown in audio files along with LOTS of other names - Do such databases of sound files already exist or do I have to record my own? I'm not sure how many first and last names I'd have to record but it seems like thousands for both genders first names and then thousands more for last names to cover a significant proportion of the people in the USA - I haven't seen something like this, but if you figure it out, I'd like to know. There's a piece of software called HouseCalls that reminds people of appointments. The proprietary software prompts the person setting up the automated reminders to record each name individually. In the beginning, it's a bear, but over time, it gets better. I guess something in Asterisk would have to do the same, right? I mean, a general list of John Jon Jonh in some person's voice, and the rest of the prompt in another, wouldn't be much better than having Festival say the name, would it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open up firewall ports for Asterisk - safe?
On 7/19/07, Ryan Stille [EMAIL PROTECTED] wrote: Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. How many developers? And what kind of developers? If they're developing things for your phone system, then you may want them on their own development boxes instead. If you're a software shop and they're just users, then that's different. Is this safe to do? I've got the Allow Anonymous Inbound SIP Calls box unchecked in freePBX. Is there anything else I need to do? Isn't there an issue with the extension/secret being passed in clear text? I'm not the most knowledgable on what freePBX does, as far as the check box. My guess is that it's just tweaking the SIP users/peers in the sip.conffile. This gives only a minimal level of security, in my opinion. It looks like I need to open port 5060, and whatever ports are inbetween the rtpstart/rtpend values in /etc/asterisk/rtp.conf. Is that right? Right now thats ports, I've read that you can chop that down to 20 ports for just a few calls. We want to have 5-6 simultaneous calls, so if I set rtpstart to 10001 and rtpend to 10100, then open up those ports, is that adequate? If it were me, and I had 20 remote users or less, I would create a VPN and have them join my network that way. Then, no SIP ports would be open to the world. And the NAT problems would pretty much disappear. You may have a slight reduction in sound quality, depending on how you set up the VPN. I really haven't had major problems with it, but again, it depends on your type of VPN. We're using a site-to-site hardware-accelerated IPSec VPN for each of our remote sites (including my house), and I have not had any problems. Except when the underlying medium (the Intarweb) has latency/jitter problems. But then, straight SIP would have issues too... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On 7/3/07, J. Oquendo [EMAIL PROTECTED] wrote: Reposted to this list: (http://lists.virus.org/voipsec-0610/msg00046.html) That's exactly the type of thing that needs to be stopped. If Dell outsourcing calls me from India, the CLI must be their number in India not a faked-in number of some office in the US. That to me is exactly the purpose of this proposed law. It is equivalent to the law regarding FAX calls that has been around for a long time. Here is the single biggest issue facing anything anyone on this list can speak about: Validation. Let's be realistic here using (again) Dell. We know based on someone's accent and lack of proper use of grammar, they are not speaking to us from a location in the USA. How can we validate that such instance is illegal. It would be hearsay because all we have is a notion without factual evidence. So how does anyone propose addressing a situation such as this. If Dell owns the number, it's not spoofing. Point-to-point T1s and such have been allowing companies to use toll bypass for years. VoIP just makes it easier and cheaper. Now, if someone pretends to be Dell in order to sell you Dekk computers, then that's fraud, spoofing, etc. This is one of the dangers I am speaking of regarding security. Let's take this situation right now, supposing I dislike you and have enough information about you. I set out to make life disruptive for you so I change my CLI to your phone number. First I want to call the bank (with your information) hopefully I can get someone insane enough to use caller ID as a source of information. Then, I decide to call the credit card companies in hopes they're going to bring up your information based on caller ID, and the scenario goes on and on. Should a company make a decision based on caller ID? Would you irrate by their actions? I know I would. We are already protected by fraud from everything you mentioned by other laws. And yet it still happens. So, what purpose will another law serve? I presume from your comment that you, like others in the Internet/VoIP arena I have corresponded with, believe that the PSTN did everything wrong and that VoIP is doing everything correctly. I don't think the PSTN did anything worse or better than VoIP, in fact I would prefer to rely on the PSTN than VoIP for certain reasons. 1) With the PSTN, any utility company, emergency service company knows with 100% accuracy that a copper line with the number 12035551212 is coming from 1 Main Street, New Haven as opposed to VoIP's 12035551212 being registered via some pre-filled out form, stating at the point in time that the form was submitted, it was at 1 Main Street however, it truly might not be at that location anymore. Someone may have moved their ATA or server. And yet, the Bells sometimes got the address wrong. And when a PRI got moved for a company I did work with, their local carrier failed to update the address in the 911 database. So, it can be screwed up, no matter what technology is used. Look, we can spoof CID through our PRI. So what? We've been able to do it for years. Have we? No, we have no need to. I'm sick and tired of all these news stories about how people can suddenly spoof CID. It's been going on for years. And anyone who gives out personal information when receiving a phone call deserves whatever happens to them. When I got a call from my CC fraud department, I simply asked for a reference number, and said that I'd call back on the number on the back of my card. Turns out it was legit, but it only took me an extra ~30 seconds to be sure. As for things VoIP has done better? The only thing that comes to me thusfar is saved someone money. Anyhow, I think this was a pretty good discussion on the topic, but bottom line if you ask me, Truth in Caller ID does nothing more than give a politician something to boast about during election time. Nothing more. Hear hear! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On 7/3/07, mlists [EMAIL PROTECTED] wrote: Keep in mind that this law is proposed by the Senator who thinks the Internet is a series of interconnected tubes which can get clogged. ommm, isn't that conceptually what a DoS attack is? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On 6/28/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 Now, I am tying to dial from 1st PC to 2nd PC I am trying to dial from 1st PC to 2nd PC through asterisk server ... 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] Why? Shouldn't you just pick up Phone1 and dial 11? If you dial it by the IP address, why would it go through Asterisk? It is not being called through asterisk server running on linux m/c. It is calling directly. As, I am running sip debub but no packet dumping is taking place. Can anybody will tell me the error I am doing. I am going to assume that the typo is in the above paragraph, and you really mean sip debug. If not, that's another problem. Thanx and regards sanchal Hope that helps, David ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+squid
On 6/27/07, rozsa [EMAIL PROTECTED] wrote: Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? I usually pass VoIP traffic without it going through the proxy. It can be dangerous, but if you set up your rules right, it should be OK. The only real exposure is that other things can hop on those ports. But then again, the safe_ports has the same challenge... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: Sorry, I forgot to mention that I want to route between VLANs without an external router and do some simple ACLs to allow PCs on the data VLANs to access the web interface of the Trixbox on the voice VLAN. thanks The only reason to route the voice VLAN is if you need the phones to access the Internet and/or vice-versa. If you only need to worry about the computers on the data VLAN accessing Trixbox's web interface, I would suggest using the Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on Trixbox, and have the default vlan for the port on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and the phones. The other option is to put in another NIC, one for the voice VLAN, the other for the data VLAN. I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). They're running around $400 right now. If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. But then again, I got a bunch that don't work that I could sell you ;) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: This is a follow up to an earlier post. Looking for a means to individualize incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to enter the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. With all due respect, this project should be handed over to whomever has authorization to administer the Asterisk box. We can tell you how to do it in Asterisk, but if you can't take our advice, our ability to help you will be severely limited. Now, we have many, many fax machines. We have our incoming through PRI, and then redirect to a channel bank. We have no problems with fax reception. When we used a Sangoma card, we did, but now that we're back on Digium hardware, we've been doing well, thus far. Probably had to do with the echo cancellation, but without infinite time to troubleshoot, we just had to get it working. I would not recommend passing fax data across the PCI bus between cards. I'm probably just superstitious, but I wouldn't do it. But it would be very simple to do with just a Dial statement. Basically, just go out and try it. Your business requirements and what you're allowed to do obviously drive where your decision is going to go. If you get stuck, and can't find answers through Google or the Wiki, then ask this list. But you can't expect us to tell you what's going to work in your business when you aren't empowered to follow our advice. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
This install uses a Sangoma card.
Could you expand on redirect to a channel bank? Could you illuminate
the connectivity for me?
A single T1 connects to??? Is the Digium card smart as in, can it
break out DS0 line(s) on a second port (to go to the channel bank)?
What we did is have 1 PRI (over T1 in the US) coming in from the telco, into
a 4-port T1 card. Then, we have 2 channel banks coming off of it: 1 has an
external echo can, and goes to our in-house phone extensions. Cordless
phones, wall-hanging phones, and anywhere that we couldn't get 2 pair into.
Basically any required analog that a person would be on. The second channel
bank has no echo can on it, and connects to our modems, fax machines, etc.
Assume for a moment that your incoming lines are in zaptel group 0, your
voice channel bank lines are in group 1, and your other channel bank is in
group 2
exten = 55,1,Goto(default,1000,1) ;go into the internal context to
route the call
exten = 56,1,Dial(Zap/25) ;ring one phone
exten = 57,1,Dial(Zap/G2/${EXTEN}) ;go out group 2, starting at
highest channel number, since the incoming calls probably start at the
lowest channel numbers, and best not to have any contention, in my opinion
In this way, Asterisk will establish a new call going out Group 2 and dial
your number. The box receiving the faxes will get the extension, and as long
as you've left your DID in there, that's what will get passed. So, it will
appear to your fax box as if it were sitting on the PSTN. Asterisk just has
to know which DIDs should have the calls passed along.
Now, in practice, we do a lot more than the above snippet, and use macros
extensively. But this should get you pushed in the right direction.
Hope that helps,
David
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: The only reason to route the voice VLAN is if you need the phones to access the Internet and/or vice-versa. If you only need to worry about the computers on the data VLAN accessing Trixbox's web interface, I would suggest using the Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on Trixbox, and have the default vlan for the port on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and the phones. The other option is to put in another NIC, one for the voice VLAN, the other for the data VLAN. I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). They're running around $400 right now. If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. But then again, I got a bunch that don't work that I could sell you ;) Ahh, interesting idea…if I understood correctly, you're basically using a layer 2 switch and trunking the voice and data VLAN to the asterisk box and doing the routing and ACL work there? Advantage is lower cost because you don't need a layer 3 switch anymore and don't have to learn a new CLI or other config method.? Here's a bit more information…the client is a building owner who occupies the first floor and is renting out the rest of the building. In addition to his own voice/data network (which would be on separate VLANs) they want to offer the building tenants the ability to use their PBX and internet connection. Due to a quirk in the service providers SIP ALG all IP phones in the building must be on the same network (VLAN) which I don't see a problem with, but each tenant's data will be in a separate VLAN. I'm thinking I could trunk the voice VLAN and all of the individual tenant data VLANs to the Trixbox to allow them access to the web interface? Any other ideas out there based on this scenario? We do something somewhat similar. Each switch has 2 data VLANs, and also is part of the Voice VLAN. Each VLAN for data is routed, but the voice VLAN only carries voice traffic. Our Asterisk server does not route packets between the networks. So, aside from some nasty attacks that sniff and replicate VLAN headers, our voice network is pretty secure. So our network has 20 different data VLANs (again, 2 per edge switch), 1 server VLAN, 1 voice VLAN, 1 wireless VLAN, and one DMZ VLAN. The data and server VLANs are all routed, and everything else is not. They have to go through some type of bridge between the networks. For wireless, that's our wireless switch. For the DMZ, it's our firewall. The voice VLAN can only reach our Asterisk box. If you use a SIP provider, you may have to either take another approach, or realize that all SIP traffic will have to remain on the host (i.e. reinvites are bad when you don't have a network path from A to B). But we're strictly IAX between offices, and PSTN thru PRI. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More FAX over T1
On 6/26/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. That's definitely an idea. If you don't need the Caller ID on the fax (and in most cases, you probably don't), this might be your best solution. Assuming, of course, the faxmodems on Hylafax are picking up the caller ID and you have Caller ID from the phone company. That would take up 2 of your PRI channels, though, per fax reception. I think I read that you have 4 fax lines. If this is correct, then the calculation is thus: 4 lines * 30 per month = 120 per month. channel bank = $200 new (for faxes, I have never had a problem with zhone CBs found at http://www.digital-loggers.com/CB.html) 2-port T1 card = $900 new, for a total of $1100 in equipment (a one-time cost) So, if you keep the solution going for more than 9 months, you'll come out ahead just buying the equipment. If you pay more than $30 per month per phone line, your break-even will be much quicker. Also, if you ever needed to add more lines, you already can have 24 faxes through Asterisk, and your fax server would be the bottleneck. This is why we run all of our fax lines off of our PRI, even though our local dialtone provider tried to convince us to buy POTS lines for each one. At around $30 per line (not including taxes), it just doesn't add up. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSkype
On 6/21/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, I recently installed chanskype on my asterisk box and it works like a dream, can phone out. But no idea how to setup the incoming calls, every time I phone my skype name it just connects and disconnect the call right away. ... Any one got some advice ? My advice: contact the developer of ChanSkype. You have to pay for that, right? Hopefully, it comes with some support. In the mean time, make sure your incoming call's context exists, ensure that you have an s and i extension in that extension just in case the number comes in differently than how you expect, and put some no-ops in, maybe have it echo the EXTEN variable. You know, basic troubleshooting. Good luck, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN on Asterisk
On 6/18/07, Dominik Zalewski [EMAIL PROTECTED] wrote: On Monday 18 June 2007 03:09:40 pm Biju wrote: Somebody sugested that we can do this with open VPN . 1st Asterisk PBX - install OpenVPN and configure it to run as a server 2nd Asterisk PBX - install OpenVPN and configure it as a client That assumes, of course, that there's Asterisk PBXes at each location. The post refers to phy phones, which may refer to hard phones, or could mean my phones. Not really sure. Anyway, you can create an SSL VPN that tunnels network to network connections with hardware network devices. There are tons out there, starting with little linksys boxes, all the way up to Cisco PIXes. Or you can use OpenVPN and some IP tables routing, or just about anything else you can imagine. If you have softphones, you can connect the PCs through a VPN, as mentioned before. But I have yet to see a sip hardphone that has an integrated VPN client. Although it would be nice... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN on Asterisk
On 6/18/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Dominik Zalewski [EMAIL PROTECTED]: wouldn't it be simpler just to run voip on some other port that is not blocked like 80 or 110 etc ? Then again if your network provider is doing things like that already what guarantees do you have they are not going to block vpn or whatever else you try ? How can you do business in a country like that where you have absolutely no guarantees your business will function from one day to the next ? Pop quiz: name me one country who has never changed the laws in a way which affect business. Now, back to the technological side, there are ways to block SIP and IAX beyond just simple port blocking. And it's really easy to sniff traffic. So, if you're doing something that the government expressly forbids, I would think that a VPN would be a bare minimum for privacy/security. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Kernel
On 6/14/07, Remco Post [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? Regards Bilal not to be rude, but what does this have to do with asterisk? From what you are telling us, I guess you need to find some fedora or general linux support medium... That's true. But isn't it easier to tell him to check his /boot/grub/grub.conf file? And only one line... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridge bug in 1.4?
On 6/12/07, Tony Plack [EMAIL PROTECTED] wrote: I have GXP-2000 phones running against Asterisk 1.4. All phones are running G729 and this is witnessed by the fact that the phone shows the G729 codec. I dial the first phone, place it on hold, dial the second phone, press CONF and the other line. The first connection goes away and the second remains connected. I have seen phones that only allow ONE g.729 stream to be operating at a time. You may want to check the documentation and see if that's what the licensing on the GXP-2000 allows. Hope that helps nudge you in the right direction, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write to multiple databases as redundancy scheme
On 6/8/07, Justin Moore [EMAIL PROTECTED] wrote: On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... If it's only for the unlikely event that a DB server is unavailable, why not have it log the CDR in text and in MySQL? If the DB server is unavailable, the records could be parsed from the text file and the database updated. Of course, if you had to do this more than once or twice, it would get a bit annoying, I'm sure. But then again, write the script to do it, and use it to populate the other databases? Dunno, just thinking out loud here. I've written a few parsers, and the format appears to be easy to parse. It really wouldn't be too big of a deal. The hardest part will be kicking it off (I'd use cron), parsing the file (my personal preference would be perl or PHP), updating the database, and making sure you don't insert duplicates. I think I would use the UniqueID as they key, and then just use INSERT statements. You may need an IGNORE in it to allow it to keep going, even when there are duplicates. It's been a while since I wrote something to update a DB where I was unsure of the data hygiene. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card
On 6/5/07, Noah Miller [EMAIL PROTECTED] wrote: Most small/medium companies have a T1 for all their phone needs. Internally there is a need for some analog lines. * Fax Machine - FXS * Security System (most ask/demand two lines) FXS * Paging - FXO * Dialup systems I think he's asking why both T1 and FXS/FXO need to be on a single card. Probably for a small server with only one available PCI slot. Back to the original question from Arun: I'm pretty sure there are no hybrid PRI/TDM cards. At least, I'm pretty sure there are none that would work directly with Asterisk. As one with experience with TDM cards, I highly recommend buying a channel bank and connecting it to a T1 card. That way, you can get a 2-port T1 card, a channel bank, and have your T1/TDM environment. I'm not saying you can't do it another way, but I find that my lines connected through channel banks are more reliable than those connected through TDM cards. YMMV ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Multiple Network Interfaces
On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. Or route the subnet and put it behind NAT. But yes, your solution is certainly viable. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? I haven't had a problem. Each of our Asterisk servers are multi-homed, and each talks SIP and IAX on all of the various networks without problems. Make sure you set Canreinvite=no to people on the outside network or you'll have audio problems. Other than that, it should be really straightforward. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Anything is possible. But I haven't seen one off-the-shelf. It really won't be a big deal to write, though. We created a timeclock application and toyed with allowing people to clock in via phone, and I even wrote the extension logic, but we opted to not enable it because we don't trust our employees that much. This was years ago, when we were running pre-1.0 code. We've switched servers a few times, so the logic is long gone, but it only took an afternoon to write and debug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, Nitesh Divecha wrote: I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Are you by chance referring to chipsets that provide hardware timing / Real-Time Clock functionality used by Asterisk? Unless I'm very much mistaken, he's referring to a Time and Attendance system. The idea is to capture times that a person clocks in and when the person clocks out, to simplify running payroll. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: This is all definitely possible by using Asterisk database interfaces, but I cannot find an existing implementation of something of this nature. It is an unusual and clever application of Asterisk. :-) Don't know how unusual. When I do contract work, most of the jobs I do have a phone number to log in and out thru. By the way, when I wrote the module, I cheated and used a System call (although I would use the TrySystem if I were to do it again) and called a very simple PHP script. Oh, and I authenticated within the dialplan so that I could easily play useful error messages without checking the returned value of the PHP script. Not the best system, but it worked in my testing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Echo
We experience echo too from time to time. It's usually headset-related, but not always. I ran a persistent ping on one of the phones, and we diagnosed a wiring problem with it. Other phones needed a new handset. The problem is that these problems need to be fixed on the phone NOT hearing echo. On 5/22/07, Asterisk [EMAIL PROTECTED] wrote: How could I check if bandwith or/and latency is causing it? If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way to gather more detailed info on SIP calls and latency? * box is connected to the 1Gb switch with 1Gb connection, and clients have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones connected to the * box. Thanks, Alex -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Alexandre VERNIOL Sent: Tuesday, May 22, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Echo Hi, Could be bandwith or/and latency ... Many causes... Alex Asterisk a écrit : Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Echo
Are your phones reinviting? Do you have any strange routing weirdness, or are they all on a single subnet? On 5/22/07, Asterisk [EMAIL PROTECTED] wrote: I tried with the ping ... all of the phones respond in ca. 0.3ms, so network seems to be OK. More than 90% of CPU on * box is idle even in peak times, so this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, but did anyone ever experience any handset problems with Polycom IP SoundPoint 430 :-) ? I will check the headsets and any possibilities of acoustical echo. Besides that, if we rule out the network, and the CPU on the * box, is there anything else that could be causing echoes on internal SIP calls? -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *David Gomillion *Sent:* Tuesday, May 22, 2007 3:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP Echo We experience echo too from time to time. It's usually headset-related, but not always. I ran a persistent ping on one of the phones, and we diagnosed a wiring problem with it. Other phones needed a new handset. The problem is that these problems need to be fixed on the phone NOT hearing echo. On 5/22/07, *Asterisk* [EMAIL PROTECTED] wrote: How could I check if bandwith or/and latency is causing it? If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way to gather more detailed info on SIP calls and latency? * box is connected to the 1Gb switch with 1Gb connection, and clients have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones connected to the * box. Thanks, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandre VERNIOL Sent: Tuesday, May 22, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Echo Hi, Could be bandwith or/and latency ... Many causes... Alex Asterisk a écrit : Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling conflicts with Clear channel
On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling conflicts with Clear channel
On 5/21/07, Vieri [EMAIL PROTECTED] wrote: --- David Gomillion [EMAIL PROTECTED] wrote: On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? I'm sure you're right because the following yields no error: # misdn-init stop # rmmod wctdm # rmmod xpp # rmmod wcte12xp # rmmod zaptel # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v snip I guess I'll have trouble getting all three cards to work together on the same box. You should still be able to get all of the cards working together. Just be sure you define your channels in the right order. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
On 5/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Sounds like DHCP to me. I've not had this problem with a GXP-2000. If it were me, I would try setting the IP address to a static IP to rule out any kind of DHCP weirdness. Now, if it still happens, then it's not really losing its IP; instead, it will be losing the connection. Or it could be unregistering with the Asterisk server for some reason, and then re-register 30-60 seconds later. That's still bad, but it's different than losing one's IP. One way you may be able to more accurately diagnose the problem would be to run Ethereal or some other packet sniffer and see if the voice packets get through your router. If not, fix or replace your router. If so, you'll need to do more detective work to see if you have a phone problem, configuration problem, cabling problem, bad port on the switch, etc. Hope that helps, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration
On 5/17/07, Tim Litwiller [EMAIL PROTECTED] wrote: We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. If it were me, I would: 1. create a conference room 2. create a .call file that dials into the speaker system 3. create .call files to dial the participants, muting them This can obviously be scripted very easily. A simple cron job copying the .call files should do nicely. If the disabled persons wish to no longer participate, you can simply delete that .call file; conversely, if you need to add someone, you can just create another one. Then you won't have to worry about incoming phone numbers and coordinating with the private school, you can bring it in when you need it, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event
On 5/16/07, William Moore [EMAIL PROTECTED] wrote: On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote: Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 This indicates an unstable d-channel. Try changing dchan in zaptel.conf to hardhdlc. If that fixes it, you are missing interrupts for one reason or another. I would also advise that you call Digium's tech support. I've seen this be a problem with the LBO value being wrong in the zaptel.conf. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside lines are just not happening...
On 5/15/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Two problems, possibly related:
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
I recommend that you put in a group, like group=2
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
if you put in the group, you can dial out via:
exten = _9NXX,1,Dial(Zap/g2/${EXTEN:1}) to start with the lowest
available channel, or
Dial(ZAP/G2/${EXTEN:1}) to start with the
highest available channel.
This will let you make more than one outgoing call at a time.
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
I don't know if it's relevant or not, but dialtone stops after I press
9, which is not what I was led to believe would happen with the
ignorepat directive.
Dial tone is generated by the SIP phone. You'll need to configure it
directly on whatever SIP device you're using. Now, if your analog phones
(like on ports 1-4) stop dial tone, you might need to be concerned.
Problem A: Dialing in. If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context. So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute before dropping the
POTS line. If I pick that SIP phone back up, and dial an outside
number, I can reconnect to the dangling call, which will hear the
tones after the 9... The outside caller will finally get dropped after
about a minute of waiting.
This is normal when dealing with POTS lines. You can try to get disconnect
supervision, try to trick zaptel into guessing what the state of the line
is, but in my experience, it just comes with the territory. Disconnect
supervision is, by far, the best solution, but most telcos stick their
fingers in their ears when it's requested...
That's one of the main reasons we use PRI where it makes sense, and have
people hang up the phones where it doesn't.
snip
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I'm not sure on this one. It could be a bad line, the line may not be fully
reset from the previous call, or something completely different.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [asterisk-users] Trixbox problems
On 5/15/07, Marco Vescovi [EMAIL PROTECTED] wrote: Just to be clear: 1) I do not use to configure the * config files with trixbox GUI but I manually edit the file 2) from my point of view, the main advantage of trixbox is to have an * installation uprunning in half an hour, then it's up to you use the GUI or manually edit files 3) I did not ask 'what I have to change in my configuration' but the question is different and it's 'how can I troubleshoot the problem'. Troubleshooting it's distribution/GUI independent task so if you don't want to help other people just relax and watch a film on the tv, don't waste your time writing unuseful mails. Regards marco If you've edited the files directly, then you undoubtedly know why you're getting the responses you are getting. There's a few different files that all come together to form each configuration file. And sometimes it's not easy to see what will override others. Another challenge when dealing with trixbox installs is dealing with the permissions. Trixbox, rightly in my opinion, changes who owns files from the default root:root. You just need to be careful when you start monkeying around with the installation files. Dropped calls are not usually easy to narrow down. You need to make sure the line is good. You need to make sure that some of the more esoteric options in zapata.conf are turned off, as I've seen them cause problems. And, frankly, we've had problems out of the 400-series cards, so we only use them in low-traffic areas. But checking IRQ misses, your hard drive DMA settings, and all of the standard troubleshooting techniques may help. I've had the ringing problem before, but for me it was an indications problem. But another time, I always had to use an Answer() before dialing out the Zap interface. Since you're headed to a POTS line anyway, it will be Answer()'d as soon as the call is dialled anyway, and putting the Answer() before the Dial() gave some of my more clueless SIP UAs a hint as to what's going on. Otherwise, they'd disconnect after 1 minute of ringing and leave the line out off-hook. So, the answer to your question is this: do the normal troubleshooting steps. But, since it very well could be configuration-related, you may want to try the Trixbox list, as this may have come up on other installations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feasibility Request
On 5/15/07, Jeremy Mann [EMAIL PROTECTED] wrote: I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. Yes, I've already done it. Just make sure you use a T1 cross-over and get the signalling correct (use pri_net instead of pri_cpe) 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. I've done that too, using the same PRI as part 1. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. This isn't really a big deal. Just have a fall-through when PSTN lines are full/down. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). You can get local presence by having a provider who can sell you a DID from your local areas and trunk them to a PRI/T1 in another area, or deliver them over SIP. The challenge with having only one analog line in a city means you can't receive 2 calls at the same time... definitely sub-optimal! I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. We're healthcare too, but in Ophthalmology. So 5-9's aren't really required here, although we've had it. I haven't really had any problems with Asterisk reliability. In the setup you propose, you're probably going to see more challenges in keeping your Internet connections up with good latency than a well-built Asterisk system. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). Buy decent servers, with redundant power supplies, raid-5 arrays with a software mirror across different array controllers, keep a warm-standby at each location, install separate diesel generators in each location, move your offices into underground bunkers in secret, nondescript locations, hire armed trolls to guard the server and pummel anyone who attempts to approach, etc. The point is, you can have as much reliability as you're willing to buy. I'm located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. Spent a couple of years in Addison, and I grew up in Houston. But I can't really offer too much on-location help, as I've moved to FL. Ah well, can't win 'em all, right? But if you get the trolls, I may be willing to make the trip ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent
Your best bet would probably be to remaster Trixbox then. You can create new RPMs to install your custom web interface and have it automatically installed. Add to that the RPMs already built and tested by the Trixbox community, and you should be good to go. I remastered a few distros years ago, but have not done so in a long time. Again, I recommend you hire a consultant who specializes in such things. Those who do it often will be able to build a fully-working install CD. I would probably start with the participants in the Trixbox project, as they'll be most familiar with its packages. You'll want to remove a lot of the parts, I expect, and knowing the dependencies before install time will make it actually work. I know I've screwed up in that department a few times... On 4/25/07, Khaled Chehab [EMAIL PROTECTED] wrote: Dear David I want customized packages to be installed from cd with no need every time to install packages and my personalized web interface , Regards *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *David Gomillion *Sent:* Tuesday, April 24, 2007 8:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent I don't really understand the question. Why do you want to do this? What do you hope to accomplish? Do you just want customized packages to be installed, or do you expect the configurations to come too? Do you want to auto-run from the CD, or just have it install? If it's so urgent, why don't you hire a consultant with experience in remastering OS installations? On 4/24/07, *Khaled Chehab* [EMAIL PROTECTED] wrote: Dears its too urgent Can anyone guide me …… I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent
I don't really understand the question. Why do you want to do this? What do you hope to accomplish? Do you just want customized packages to be installed, or do you expect the configurations to come too? Do you want to auto-run from the CD, or just have it install? If it's so urgent, why don't you hire a consultant with experience in remastering OS installations? On 4/24/07, Khaled Chehab [EMAIL PROTECTED] wrote: Dears its too urgent Can anyone guide me …… I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billion ISDN problem
I don't really know what Billion ISDN is, but some basic Asterisk troubleshooting seems to be in order. What does your zapata look like? Looks like you have some errors in there... Next, you have PRI, right? did you compile libpri after installing zaptel? Finally, you need to make sure your zapata.conf makes sense with your zaptel.conf, as far as channels, signalling, etc. Hope that helps you know where to start looking, David On 4/23/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 30 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 31 of zapata.conf -- Registered channel 1, PRI Signalling signalling Apr 23 15:27:23 WARNING[2205]: chan_zap.c:1099 zt_open: Unable to specify channel 2: No such device Apr 23 15:27:23 ERROR[2205]: chan_zap.c:7241 mkintf: Unable to open channel 2: No such device here = 0, tmp-channel = 2, channel = 2 Apr 23 15:27:23 ERROR[2205]: chan_zap.c:12011 setup_zap: Unable to register channel '1-2' Apr 23 15:27:23 WARNING[2205]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 Apr 23 15:27:23 WARNING[2205]: loader.c:554 load_modules: Loading module chan_zap.so failed! can you help me please??? thanks a lot ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording
I did this with a Nortel MICS a few years ago. No problem.
The dialplan was something like:
[incoming]
exten = _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten = _X.,2,Monitor(filename)
exten = _X.,3,Dial(Zap/G2/${EXTEN})
[outgoing]
exten = _X.,1,setvar(filename) ; If you want to record outgoing calls
exten = _X.,2,Monitor(filename); use these two lines, otherwise, just skip
them
exten = _X.,3,Dial(Zap/G1/${EXTEN})
Obviously, this isn't production code, but you should get the idea. If
you're in a 2-party area, you probably need to make your employees sign a
disclosure, and play a sound file to your callers to warn them that the call
is/may be recorded. While it will waste space, I recommend starting the
recording before the file is played. That way, if you're ever challenged,
you'd have something to back up your position that the caller knew. Add the
signed disclosure, and you may be OK.
Of course, I am no lawyer. And you probably ought to talk to one before you
do this. We did, and he had some helpful pointers on what to include in the
disclosure.
There are some areas that will require you to play an annoying beep to
callers. We didn't have to do that, so I'm not sure of the best way to go
about it.
Good luck,
David
On 4/20/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also, could the calls go into the cdr for retrieval/browsing later?
What hardware/server would you recommend?
Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing CallerID
On 4/19/07, Forrest Beck [EMAIL PROTECTED] wrote: I thought of maybe adding a key for each extension to the astdb and have a Macro query the astdb. Any other ideas? That's how we do it. We created a MySQL DB that maps DIDs to extensions, and a php script to write our configuration files for us (a file called did.conf, which is #include'd into extensions.conf), as well as push the DID into the Asterisk DB. Actually the DB holds all of the information for our phones, and all of the files we need are generated each night, including sip configs, provisioning files for our Polycoms, the dhcpd configurations to give static addresses, and a few other miscellaneous files. And it creates our phone list. The nice thing about building the DB yourself is that you can do anything you want with it. On one of our boxes, I went a step further and created individual outgoing contexts, one for each device. The context set the caller ID. But it was definite overkill; I haven't done that since. Thanks. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird extension behavior
On 4/1/07, Mark Hennessy [EMAIL PROTECTED] wrote: Hi, I'm using Asterisk with two Cisco 7960 phones using SIP. I'm seeing the following weird behavior: SIP Phome 1 is extension 4002 SIP Phone 2 is extension 4003 So, did you name your SIP user/peers 4002 and 4003? It doesn't matter, but the word extension really means more about what you see in extensions.conf. You can check this by looking either at the phone's configs or in sip.conf, or better still, both to make sure they match. I call 4002 from 4003 and that works fine. I call 4003 from 4002, and it rings locally to 4002, never gets to 4003. This sounds like a problem with the extension.conf file. Without the relevant portions of it, though, there's little we can do to help troubleshoot. I'm able to send a config query packet to 4003 from the asterisk console and get a response, when I send one to 4002 there is no respone. I know that both phones pull down their config via TFTP properly, I look in the network settings and see that 4002 has been given an IP of x.y.z.201 and 4003 has been given an IP of x.y.z.202 and the asterisk box is running on x.y.z.74. The next step would be to run sip show peers and sip show users at the Asterisk CLI to see how/if the phones registered with the expected IPs. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simplify
On 4/2/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(2)
exten = 2,103,Hangup
exten = 20100,1,Dial(SIP/20100,30,Ttm)
exten = 20100,2,Hangup
exten = 20100,102,Voicemail(20100)
exten = 20100,103,Hangup
exten = 20200,1,Dial(SIP/20200,30,Ttm)
exten = 20200,2,Hangup
exten = 202000,102,Voicemail(20200)
exten = 20200,103,Hangup
exten = 20300,1,Dial(SIP/20300,30,Ttm)
exten = 20300,2,Hangup
exten = 203000,102,Voicemail(20300)
exten = 20300,103,Hangup
exten = 20400,1,Dial(SIP/20400,30,Ttm)
exten = 20400,2,Hangup
exten = 204000,102,Voicemail(20400)
exten = 20400,103,Hangup
Yes, 2 ways:
1. Use a macro:
[macro-whatever]
exten = s,1,Dial(SIP/${ARG1},30,Ttm)
exten = s,2,Hangup
exten = s,102, Voicemail(b${ARG1})
exten = s,103,Hangup
2. Use pattern matching
exten = _20[0-4]00,1,Macro(whatever,${EXTEN})
Is that simpler?
By the way, I took the liberty of adding b for busy to the macro. But you
may want to consider using the standard extension macro provided with
Asterisk instead. It allows people to press * to check their voicemail, and
a few other handy features. Why reinvent the wheel?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power
I was reading somewhere that Polycom cables only work when the power is over pins 1236 (signal pairs). I won't swear to it, but that's what I read. Anyway, most PoE injectors (not all, but most) inject power on pins 4578 (non-signal pairs), meaning it won't work with things that need the power on the signal pairs. For that, you'll need a switch that supports PoE. For simple testing, Netgear has some cheap unmanaged switches that provide PoE that's 802.11af-compliant. We're using a couple of them in really small satellite offices, and they've been holding up pretty well. But I wouldn't use them in a large-scale deployment, since they don't support VLAN trunking, QoS, and what-not. On 3/29/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Noah Miller wrote: Hi Mike - I have a 501 with traditional power and a 301 with PoE. I rightfully assumed that the traditional power from the 501 would work on the 301. How do I get the PoE to work? Do I use the Polycom PoE cable in addition to whatever PoE injection method I use? I have a Cisco PoE injector that works on my Cisco AP350 and my 7960. No combination of this injector, the Polycom cable, and the phone result in success. I have 18v PoE injectors that I use for other things, but I hear that 802.3af is 48v, therefore probably wouldn't work. How do I use Polycom PoE? You'll probably have to get different injectors, or a new PoE switch. The newer Cisco PoE switches do speak 802.3af, but many of the older Cisco PoE products do not. The original Cisco PoE implementation was proprietary and does not conform to 802.3af. Polycom has cables available to support Cisco PoE and 802.3af PoE. They are, however, different cables. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. YES! Would creating a public database, managed by users be worthwhile to anyone? I'm not sure the technical issues will be as easy to work out as one would hope. When creating such a system, care must be taken to keep the information accurate and up-to-date. And where would you get the information from in the first place? Thanks – Any input is greatly appreciated. What I would like to see is a distributed system that allows for updates to be rsync'd in, so that those of us who keep our servers off the Internet can move it through a QA process and then push the update through. Some type of a mirror system, where the packages can be updated from time to time (like daily). -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com http://www.xstreamhost.com/ - Web Hosting http://www.SophMedia.com http://www.sophmedia.com/ - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail
I am having an issue with 1.4 where we can't successfully transfer a call
directly to a voicemail box. We hit Transfer on the phone and dial the
mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and starts to play its message and at this
point. We should then hit Transfer again at this point the person doing
the transfer should drop off the call. However we just continue to hear the
voicemail message and the caller continues to sit on hold.
I've not worked with 1.4 much yet, but I'd try changing my dialplan to:
exten=_*40XX,1,Answer
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
That way, I would know that the channel is answered, which is what often
will stop IP phones from allowing the attended transfer to complete.
Hope that helps,
David
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I use sip gateway of PCI card?
I don't think it really matters. I'd go with which ever is cheaper. On 1/30/07, Robert Augustyn [EMAIL PROTECTED] wrote: Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail Storage
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 25 January 2007 6:30 pm, David Gomillion wrote: I mean that I would like to have a system in place so that Asterisk, as a privileged service, can gain access to Courier's IMAP storage. Having to keep track of all of our users' passwords in the Asterisk configuration is going to provide a ridiculous amount of administration, as we force them to change their passwords often in our single-sign on environment. How do they log on to check their voicemail? Is your SSO system entirely numeric? -A. I'm not talking about setting the voicemail password. I'm talking about not having to put my users' email passwords in the voicemail.conf file. Asterisk, if I understand correctly, needs each user's email password to deliver the voicemail, to integrate messaging into the IMAP server. Or it needs a general user that has rights to deliver and read any mailbox, which I don't know of existing in Courier. You said you had done some testing. What model did you use? Did you put each user's email username and password in the voicemail.conf, or were you able to come up with a general user for Asterisk to use when delivering every voicemail? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queuing Problem with Asterisk
On 1/25/07, George C. Attopany [EMAIL PROTECTED] wrote: Hi, description of problem cut out for brevity member = Zap/9-1 member = Zap/10-1 member = Zap/11-1 member = Zap/12-1 member = Zap/13-1 member = Zap/14-1 member = Zap/15-1 member = Zap/16-1 I don't think you want the -1 on the end of each line. Try: member = Zap/9 member = Zap/10 member = Zap/11 member = Zap/12 member = Zap/13 member = Zap/14 member = Zap/15 member = Zap/16 Hope that helps, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail Storage
Since you've done some work with Courier and Asterisk's IMAP voicemail, is there a place you documented your findings? I'm interested in merging the two. Is there any way to do it without having to ask all of my users for their passwords? On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 25 January 2007 3:27 pm, Bruce Reeves wrote: I am doing some testing with 1.4 and the imap storage and a exchange 2003 server. I have not had any positive results so far using the notes on the wiki or the docs in the release. My current settings are I've done some work with the IMAP voicemail storage and Courier-IMAP, and have had it working. It does seem like it just cannot get to your IMAP server; have you tried the imaptools test program (the name escapes me, it's the only binary produced by the imaptools package), giving it the same IMAP connection string as what Asterisk reports? My development box is offline at the moment or I could give you specific details. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail Storage
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 25 January 2007 4:48 pm, David Gomillion wrote: Since you've done some work with Courier and Asterisk's IMAP voicemail, is there a place you documented your findings? I'm interested in merging the two. Is there any way to do it without having to ask all of my users for their passwords? There really weren't any findings; I wrote a small patch which corrected how the IMAP connection string was built, but other than that it just worked. As far as not asking all your users for their passwords -- I'm not sure what you mean -- Asterisk needs to know the voicemail passwords, and those are stored in voicemail.conf. I'm not using IMAP server passwords at all. I mean that I would like to have a system in place so that Asterisk, as a privileged service, can gain access to Courier's IMAP storage. Having to keep track of all of our users' passwords in the Asterisk configuration is going to provide a ridiculous amount of administration, as we force them to change their passwords often in our single-sign on environment. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on IP phones...
We have this from time to time. It's usually someone using a cheap headset that's turned up too high. Polycom's have some settings you can tweak to cancel out the echo, although they're not supported. We used them for a short while, but they seemed to interfere with the echo can on our Sangoma card, so we had to set them back to default. You might want to see if Aastra phones offer some type of internal echo can, if there are cheap headsets being used (the person not hearing the echo should turn his/her volume down), or if there's one phone in particular causing problems (could be a bad network cable or NIC on the phone). On 1/23/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be echo on the SIP phones when they are all using ALAW/ULAW and are on the same local network. I even have QoS configured on the Linksys SRW224P switch to give priority to the voice services. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
On 1/15/07, chester c young [EMAIL PROTECTED] wrote: g option to Dial only continues the dialplan if the destination (called) leg of the call hangs up. It will NOT cause the dialplan to continue if the source (calling) leg of the call hangs up. When the calling channel hangs up, Asterisk will send the remaining leg of the call to exten = h. this is exactly right and is exactly the problem. when the called leg hangs up the dial plan does not proceed to the next priority. Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, well, that's the problem. We have this with a few of our TDM400's, as well as an old X100P. callprogress=yes did not seem to fix them much. So, the result is that our phone system always thinks we are the ones hanging up. Sometimes that causes a bit of a problem when a person is in a queue and hangs up before they get to an agent. In those cases, the agent gets the dead line. But, when they hang up, the line is freed. In that case, you would just have to use the 'h' flag, and put the rules there, and realize that your system will always believe you hung up. The other option is to get a line with disconnect supervision from your phone company, or some type of digital trunk (PRI, etc). Hope that helps, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX '' '||' Grandstream GXP-2000 problem
On 1/15/07, J. Espinal [EMAIL PROTECTED] wrote: Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHzBox... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to hold and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the hold in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any... Thanks, Are you using G.729? Last I heard, grandstreams could only have one call via G.729 at a time. It had something to do with the licensing that they used, I think. Just a thought... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On 1/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I don't have any experience with an Audiocodes Meidant 1000, but I'll try to help you I am using Polycom 501's and 601',s We have a lot of these and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. the PSTN party hears dead air until the call is answered or goes to voicemail. I would start by making sure the Music on Hold actually works, and that the SIP phones are properly configured to use a MOH context that actually exists. If those things check out, I would try using a blind transfer and see what happens, try transferring when the 3rd party answers (VM or whatever), and watch the console carefully with as much verbosity as possible. I'm not really sure where to start my troubleshooting. Any one have any experience with this type of setup? Hope this helps, David Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: PRI Problems]
Tell us about channels 45-47... We have 45-47 defined in zapata.conf, but not in zaptel.conf. Probably not the problem, but it might cause some confusion... Also, look at your timing sources. They don't look quite right to me. Spans 1 and 2 are both marked as primary timing sources. Hope that helps, David On 1/4/07, Rob Schall [EMAIL PROTECTED] wrote: Correction in my zapata.conf file I used Hey Everyone, So this is a problem I've been having for sometime now. I sent a few messages to the list with no luck. The problem is that when people dial into the Asterisk system using DID numbers, it works the first time or 2, then I get busy signals. A friend recommended I clear out the zapata and zaptel, start over, and recreate my wanpipe stuff. He thought the problem was with the spans themselves. However, when I use the following information, and restart wanrouter, I get an error. Zapata.conf [channels] language=en context=internal switchtype=national pridialplan=unknown signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=yes group=0 channel = 1-23 group=1 signalling=em_w channel = 25-47 Zaptel.conf span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 em=25-44 -ERROR Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: Clear channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) Channel 25: E M (Default) (Slaves: 25) Channel 26: E M (Default) (Slaves: 26) Channel 27: E M (Default) (Slaves: 27) Channel 28: E M (Default) (Slaves: 28) Channel 29: E M (Default) (Slaves: 29) Channel 30: E M (Default) (Slaves: 30) Channel 31: E M (Default) (Slaves: 31) Channel 32: E M (Default) (Slaves: 32) Channel 33: E M (Default) (Slaves: 33) Channel 34: E M (Default) (Slaves: 34) Channel 35: E M (Default) (Slaves: 35) Channel 36: E M (Default) (Slaves: 36) Channel 37: E M (Default) (Slaves: 37) Channel 38: E M (Default) (Slaves: 38) Channel 39: E M (Default) (Slaves: 39) Channel 40: E M (Default) (Slaves: 40) Channel 41: E M (Default) (Slaves: 41) Channel 42: E M (Default) (Slaves: 42) Channel 43: E M (Default) (Slaves: 43) Channel 44: E M (Default) (Slaves: 44) 44 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Any quiet 24 port POE switches out there?
Sorry, they were the NetGear 24-port rack-mount 1U switches (7326P is the part number, if I remember correctly). It's the brand that was mentioned right before my reply... On 1/4/07, Allen Casteran [EMAIL PROTECTED] wrote: David Gomillion wrote: We bought 7 switches, and 3 of them failed after one year. It took quite a bit of doing to get the off-shored customer support to read their own literature to cover one switch under warranty. Never could get the other two covered... When I got the warranty status updated, they told me to reload the flash and call them back. When I called back, they had lost the incident. I gave up, threw them away, and bought the linksys 24-port PoE switches. Haven't had a single hiccup from those... What were the 7 switches you had some many problems with? NetGear? LinkSys? McSwitch? Allen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HowTO configure voice T1
T1s can use many different signalling types. You need to find out which one is running, what the line encoding is, etc. PRI vs T1 are not the only distinctions... On 1/4/07, Mark Greene [EMAIL PROTECTED] wrote: Alright guys here is my question. What is do I need to set switchtype, and signalling to in zapata for a voice T1. This is not a PRI. I cannot say that enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice channels. There is not a D Channel. It runs from one office to another and USED to plug into two opt. 11c but now one end is going to plug into an asterisk box. - Mark ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
We bought 7 switches, and 3 of them failed after one year. It took quite a bit of doing to get the off-shored customer support to read their own literature to cover one switch under warranty. Never could get the other two covered... When I got the warranty status updated, they told me to reload the flash and call them back. When I called back, they had lost the incident. I gave up, threw them away, and bought the linksys 24-port PoE switches. Haven't had a single hiccup from those... On 1/3/07, Jerry Jones [EMAIL PROTECTED] wrote: I suspect any 24port will have a fan. The Netgear FSM7326P are not too bad and we have had good luck with them. ps - I also load their open source software. On Jan 3, 2007, at 4:51 PM, John French wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Match a Numer - then continue with, dialplan
I think you're making it far too difficult.
What I do is something like this:
[outgoing]
include = internal
include = longdistance
;Always include internal first, as matches from the first include
;will be used first. This allows you to make sure your internal
;extensions don't go out your trunks.
[longdistance]
ignorepat = 9;
include = default; already included from local, but putting here for
clarity
include = local;
exten = _91XXX,1,Macro(trunkout,${EXTEN}) ;Medium Distance
exten = _91XX,1,Macro(trunkout,${EXTEN}) ;Long Distance
Then, I have:
[macro-trunkout]
exten = s,1,Set(cname=${DB(showname/${CALLERIDNUM})});
exten = s,n,Set(cnum=${DB(shownum/${CALLERIDNUM})});
exten = s,n,GotoIf($[foo${cnum} = foo]?6); //if calling from ZAP
channel that set caller ID already
exten = s,n,Set(CALLERID(name)=${cname}|a);
exten = s,n,Set(CALLERID(number)=${cnum}|a);
exten = s,n,Dial(${TRUNK}/${ARG1:${TRUNKMSD}});
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Congestion(30)
exten = s-CONGESTION,2,Hangup
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-BUSY,2,Hangup
Why is this important? It's not. But it is fundamentally different from
what you're asking. You want to match a partial extension dialed and
then continue appending digits. What you really need to do is wait for
the whole number, then decide what kind of number it is, do the
processing, and send it on its way. It's just a slight change in the way
you're thinking, because you understand that there's a class of numbers
to treat differently. And that's OK. Just don't do anything with it
until the whole extension has been entered!
You'll notice that, anything not going through the trunkout macro
doesn't get tweaked, and anything that goes through there will read from
the database. I could just as easily set a single value, but I have some
users that I want to go out as themselves, and different departments
that have a general number, etc. I found the Asterisk Database to be the
easiest to tweak, as I have some scripts to allow admins to change the
effective CallerID on the fly.
I hope this helps! Asterisk can do what you're asking, and it does every
day.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Penalty
I have seen a few requests to allow people to ring different groups depending on how long a queue call has been left unanswered. I have a suggestion on how this can be accomplished. Suppose for a moment that calls that timed out were treated the same as calls that had come back with a busy or congestion message. Then, when a call has been unanswered in a queue, it will move on to the next group. At least this is the way I would conceptually believe it should work. As of my testing (on 1.2.5), this NOT the way it works. Take for instance: [myQ] music = muzak strategy = ringall context = lemmeout timeout = 15 weight = 0 wrapuptime = 0 announce-holdtime = once member = SIP/007 member = SIP/008 member = SIP/009 member = SIP/107,1 member = SIP/108,1 member = SIP/109,1 member = SIP/207,2 member = SIP/208,2 What this should do is ring the first 3 extensions. If they are busy and/or don't answer, ring the first 6. If they are busy/don't answer, then go ahead and ring them all. Has this been fixed in a later release? Would this break any body's application? I'm interested to hear any thoughts on the subject. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Encrypted IAX termination
Does anybody know anyone who offers encrypted IAX termination at reasonable rates? I googled, searched the WIKI, but didn't find a whole lot of information. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP MWI
If it's already been covered, please forgive the repetition. I searched Mantis, but couldn't come up with anything. We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped working on SP IP 300s and 600s. All of them. I tried splitting the friend entries in sip.conf into user and peer. I made sure the context of the voicemail box was on the end of the mailbox option in the sip.conf file. I checked and rechecked the config files for the phones. Nothing worked to restore the MWI's until I reverted to 1.2.5. Then everything just worked like it should. Has anyone else seen this? Is there an open bug, or a fix already merged into svn? Thanks, David Gomillion ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config File Management
Sorry for thread breaking... I'm on digest. I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? I'm using CVS. I only have one server right now. I use it on other clusters to sync files and it works for me.. Instead of doing this, I ended up creating a MySQL database and a few scripts to generate the config files for each of my servers. All I have to do is update the database, and the correct server pulls the information from the DB, generates the file, reloads, and sends reboot messages to the proper phones. Very specific to my needs, but extremely fast and effective. And all it requires on each Asterisk server is cron, PHP, and php-mysql. I had to customize a few of the variables inside the PHP scripts for each server, but by putting them close to the top, it's not a real big deal when I update the scripts to customize them for my servers. Mind you, I only have 4 servers on this system, but we don't anticipate growing beyond one more server for a while. One thing to mention that I have found: use lots of macros. Some of my macros require 6 or 7 arguments, but they are extremely flexible and trivial to generate on the fly through these tools. Each extension fits in only one line in the dialplan (calls a macro). Entries in the DB turn on and off features, sets the timeout, forwards to another extension or sends to voicemail, etc. Just what I'm doing. Hope it helps. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID, Attended Transfers, Polycom
We have contracted with an outside call center to provide sales for a certain product. We want to be able to transfer people over to those dedicated sales agents using an attended transfer (so we can prepare them with as much information as we have), to a regular extension. So far, so good. All of this is working just great. We want the caller's information presented as the CallerID so that the outside staff can use the information for tracking the calls. When the call leg is created that starts the transfer sequence, the CallerID is set to our outgoing CallerID by the Polycom phone. Are there any good tricks to determine how to set it, such that it will match the caller ID of the number that called? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Buttons (Key system behavior)
Since this issue has raised its ugly head again, and I still don't know a very good solution, I wanted to bounce a few ideas off the gurus on this list. Scenario: You have an administrative assistant who need to be able to take calls for a PHB Desired Behavior: Assistant has a line button that shows status of the boss's phone. Pressing the button, no matter the state of the call, allows the phone to join the conversation. Or maybe it only allows joining if the boss isn't on the line. I've seen it both ways. Solutions: 1. Program the SUBSCRIBE-NOTIFY model alluded to in the Polycom manual. Pro: probably the right way to do this. Con: hasn't been done up until now, so it probably isn't at the top of any of the programmers' list. I wanted to throw out another possible solution for comments: 2. Set up a macro for extensions. What it does is this: - Set 'hint' to get the lights on - place call in a private MeetMe conference room - if there's a call in the conference room already, then the line is 'busy' - put a call file in the spool so that the intended callee is invited to the conference - if the callee doesn't join the conference within a set timeout, pull the caller out and send him on his way (i.e. voicemail, etc) The next part is where it starts getting a little fuzzy. For someone else to be able to join in, the line button must actually have an automatic off-hook extension. It would do one of 2 things: IF: the conference room is not empty, we should join that conference the conference room is empty, go to a meta-space, where we can dial an outgoing phone call. This would completely break the default behavior of most SIP phones. On-hook dialing couldn't happen, nor could Dial soft-buttons. But I don't know how else to get the assistant on the call by simply pressing the button next to the flashing light... The more I think about this, the more I think the complexity in the dialplan is not worth it; however, it's preventing a few installations here, and I'm sure there are others around that this is a deal-breaker on. I'm a programmer by training, but I've been so busy with IT garbage that I don't think I'll have the time to learn the SIP channel well enough to implement #1. But I'd be willing to put a bounty on it, if others want the feature too. Thought? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Calling on all Polycom Experts
From: Ryan Stark [EMAIL PROTECTED] Subject: [Asterisk-Users] Calling on all Polycom Experts To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error message: There was an error proccessing the config file, Error of type 0x4020. Then I used the config file that came with the new release to write a new config for that phone, rebooted, same error. I did the 468* reset and it did the same thing again. Any ideas on what that error is and how I fix it? (Polycom logs quoted bellow sig.) Thanks, -Ryan I wouldn't call myself an expert, but I don't see in the logs where the phone successfully requested the config files. We had the same problem when upgrading. It had to do with our FTP server's firewall. They changed the way the FTP stuff is done when requesting the phone's cfg file. Hope that helps get you on the right track. I didn't discover what the root problem was until I moved the FTP files to an un-firewalled box all together to see if the FTP server itself was whack. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: astGUIclient installation problem
Hi everyone: I try to install astGUIclient for my call center. I'm interesting to put in work the monitoring client, i follow step by step the installation from scratch but when i try to run the application from my Windows XP astGUIclient i got the follow error: Client does not support authentication protocol requested by server; consider up grading MySQL client at astGUIclient_1.1.0.pl line 4704 At the risk of being a jerk, did you try to find the answer on your own? http://www.google.com/search?hl=enq=Client+does+not+support+authenticat ion+protocol+requested+by+server%3B+btnG=Google+Search I just copied the first bit of the error message into a Google search box. Lots of information. This error usually means you are running a 4.1+ version of MySQL server, and the client doesn't understand the newer authentication protocol. You need to set the password using the OLD_PASSWORD function in MySQL. Take a look at the top entry when you run the Google search, as it is directly from MySQL's manual. This should fix the error. Good luck. And in the future, you can save time by trying a really quick Google search on error messages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] # Transfers
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on hold for a couple seconds. Pick them back up and press #. They hear a beautiful, short, DTMF tone, nothing more. Is this a bug, or did I miss something in the configurations? Has anyone else had this problem? As far as the transfers, I found a message at http://lists.digium.com/pipermail/asterisk-users/2004-September/062080.h tml but there were no more messages in that thread. The other zombie channel transfer questions didn't seem to fit the problem, but I may be wrong. Any suggestions would be greatly appreciated. Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Escape context and queue application
I am running stable 1.07. I have tweaked the queue app a little, so I'd like verification from someone running stock... When the queue tries to announce the position, I go ahead and read the options for the escape context, which are in the queues.conf as the thank-you. However, pressing any key during the announcement is ineffective. As soon as the announcement is over, I can press the key. Has anyone else run into this? Or is there a better way to announce the escape sequence to callers than putting it in the announcement? I did that so that it would play once and then repeat every 90 seconds, starting from the beginning for each caller. I commented out all the code in app_queue.say_position(struct queue_ent *qe) that told the caller the position and hold time (as we ONLY want the announcement to play). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom DTMF
Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to inband. Without making any configuration changes on the phones, I changed the DTMF mode to rfc2833. The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the right way to fix this, as Polycom's configurations, by default, will encode DTMF in the active RTP stream. There may have been a change in the sip channel's code that is causing this. Others on the list have indicated that they worked around the problem by reverting the version of the sip app to an older version. As the new code usually fixes other problems, the solution of reverting seemed to be counter-productive, so I tried other DTMF signalling modes. Thankfully, the stock Polycom configs will work with Asterisk's sip.conf rfc2833 DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread *-1.0.7 DTFM = Not working from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. Sincerely, David Gomillion ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom/sip.conf/voicemail configurator
I have just created a very rough (read hack-ish) version of a Polycom SIP phone configurator. It allows you to define phones, create registrations, and such. By describing stuff about users, I am attempting to divine what the configuration should be. This is a VERY early first step in that direction. Right now, it: 1. Parses the phone1.cfg file included with whatever version software you use. 2. Reads the phone and registration entries to determine the file names to use, create the registrations, create a customized XML file, generate the sip.conf, and also generate a very basic voicemail.conf file. 3. Generates a random password for both the phone configuration and the sip.conf file (yes, the SAME random password as appropriate, and yes, DIFFERENT passwords for each registration) 4. Outputs 2 files per phone: .cfg, and extension.cfg, as well as 3 global files: voicemail.conf and sip.conf, and voicemail.conf It is not very powerful right now. It is written wholly in PHP and MySQL (I did it in an afternoon), except for a piece of shell scripting to glue it together and move the files around and stuff. Before I take a couple hours and clean up the code (i.e. remove passwords) I have created some shell scripts to kick off the file generation and then copy the appropriate files where they go. We close at night, so I have the flexibility of pushing all of the files out to the server, sending a reboot to the phones, and reloading Asterisk at night, every night. This means each extension will have a new password each day, but no user will ever know (or need to know) their own password. If there is sufficient interest, I will clean it up and put the files up on a server somewhere. It does not require anything special, but it will be of most interest to those who know a little about PHP, as the [general] section of each of the .conf files is hard-coded right now. I'm planning to put in files that will be parsed, yadda yadda yadda, but I have enough other things on my plate right now. Please respond to me directly if you would have interest in this, as a bunch of me toos on the list will do no good but to annoy everyone. If it works out that there's sufficient interest, I will send a single announcement to the list, and put an entry on the WIKI. It only works for polycom phones, so if you don't use a bunch of them, it probably won't be worth your time... Thanks, David Gomillion ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom and call waiting again...
Message: 10 Date: Wed, 26 Jan 2005 17:53:39 -0500 (EST) From: Sean A. Newton [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII On Wed, 26 Jan 2005, Noah Miller wrote: Have you tried adding SetGroup(), and CheckGroup() functions to the dialplan that rings the phone? It maybe something to try. I think the problem is that these functions only work from the dialplan. In this case, Sean is trying to get calls from a Queue (and not the dialplan) to the correct line on the phone. I was thinking about implementing a queue for our receptionists, but this problem prevents me from doing that, and I haven't figured out any way around it. Maybe the new 1.4.1 firmware provides a way to disable that horrid call-waiting feature? Has anybody gotten it to run successfully? I have a number of queues which ring to dedicated call appearances, if that's what you're trying to do. In my SIP config, I have: (sorry about capitalization... For some unknown reason, we had to standardize on M$ Outlook... *sigh*) [1234] Type=friend Context=whatever Host=dynamic Secret=password1234 Dtmfmode=inband Disallow=all Allow=ulaw [1234b] Type=friend Context=whatever Secret=password1234b Dtmfmode=inband Disallow=all Allow=ulaw Outgoinglimit=1 . . . Rinse, lather, and repeat for each queue you want on a phone, or as many call appearances as you have. Since we have IP600s, and nobody is in more than 5 queues currently, it works well for us. We avoid the call waiting issue using the outgoinglimit=1 directive, as the Asterisk server will only send one call to the phone at a time. I know that it is supposedly going away soon, but it's working right now. I just statically define the queues to have the appropriate call appearances like Member = SIP/1234b Then, in the phone1234.cfg file, I set each appearance to be 1234, 1234b, 1234c, etc. The problem with this is that each IP600 adds 80 lines to the sip.conf file, and each time we add queue members, I have to modify the queues.conf file. But it works for our needs. Exactly.. SetGroup was suggested by someone on the irc channel.. I looked at it briefly. I was then shot down by someone saying to save my effort, it didn't work. I suspected as much, due to the fact that the Queue function doesn't use the exten config for that phone. And it shouldn't.. The phone should be able to take care of this problem.. Yeah, I didn't think it would work, so I never went down that road either. I've unfortunately got myself into a bind because I've bought ~35 of these phones. :eek: Well, if you just can't use them, I could send you my address ;) If everyone thinks SetGroup and CheckGroup will work, I will spend the next days working with it, but I don't want to go barking up the tree of something that doesn't look like it will work. :| I'm also interested to try out the 1.4.1 firmware. Just need to procure a copy of it.. The 1.4.1 firmware is available now from a website that escapes me, but is linked from the WIKI. I've been testing it for about 12 hours, and so far so good :) --Sean Hope this helps, David Gomillion ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom POE Rumor
Message: 9 Date: Thu, 02 Dec 2004 11:33:16 -0700 From: Kevin P. Fleming [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Tim Jackson wrote: They aren't dumb hubs, they are dot1q capable switches. I do not understand why people are saying they need special POE cables for the IP500. Mine came with a cable that injected power into the cable, and from what I read, its Cisco and 802.3af compatible out of the box. My IP500s were $174/piece shipped. Maybe you will have better luck than I with them. I could be wrong on some of this info. The IP500 will not draw power from either an 802.3af or a Cisco powered Ethernet link without a special adapter cable from Polycom. Same goes for the IP300. Rumor has it that current phones are starting to ship with the 802.3af adapter cable included. As the starter of the rumor, let me apologize. I have received my first shipment of IP300s, and they do NOT include the cable. I was given bad information, it seems. It was said that when I said Power Over Ethernet Accessory Cable for 802.3af, it was somehow misinterpreted as a Category 5 Patch Cable, which the phone does come with. Why I would really care about a UTP cable, I don't know... I don't understand how, as I was very specific, but oh well, my reseller is trying to make it right for me... The IP600 has built-in 802.3af support, so it will work on an 802.3af link. It still requires an adapter cable to work on a Cisco powered link. I can verify the 600s working on an 802.3af switch, as I have 15 in production right now, working like a dream. ___ 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] Shared line appearances
Title: Message All right. It would appear that I am not the only one interested in shared line appearances. Many others have stated that they wish for the key-system-like feature of the blinking lights. Quite frankly, I don't think it's a good thing, but the people who use these systems are very resistant to change. I set up call queues, pickup groups, 1-touch transfers, and still nothing seems to placate them. If I could, I would just replace the users... I mainly use Polycom SoundPoint IP phones: some 300s and some 600s. Bottom line is that if I am going to be able to finish the rollout of the phone system, and switch away from having 2 PBXs vying for power (Asterisk and the Nortel NorStar MICS system), I am going to have to get this feature working. I have received authorization to offer a bounty to get it working in Asterisk, and to then contribute the source to the project. As I have studied the issue, I'm not sure it is within the "master plan" for asterisk. Searching the archives, it seems we only expect Asterisk to be a "clever UA". The people asking were advised to get a real SIP proxy. In passing, someone asked if chan_sip2 would support it, but I found no response. Many references to SER have been made. I have installed SER successfully. I then tried to make the feature work, but have been unsuccessful. Both lines will ring, but the first person to answer the call gets it, and the other phone's lights are as dark as can be. SER does not seem to do any better with "line-seize" than Asterisk. At least Asterisk has the hint to allow the lights to work (I have not yet implemented this, but since it does not meet the requirements, it does not really matter)... but neither system will allow the caller to press the blinking light on a call that was placed on hold to answer it. I am now looking at other SIP proxies. I am in the process of installing sipXpbx, which includes many different pieces of the Pingtel sipExpress system that have been open-sourced. I am not sure which pieces I will need specifically, so I will install the whole shooting match and see if the feature even works. If it does, I'll remove packages and try to reintegrate with Asterisk. Has anyone gotten shared line appearances to work with Polycom Soundpoint IP phones? Not just blinking lights, but the whole shebang: lights, pressing the button to seize the line, shared registrations, etc. Is it best to work with a 3rd party SIP proxy/router/whatever, or should we pool resources and get the feature integrated into Asterisk somehow? Looking forward to your thoughts, David Gomillion ___ 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] RE: Shared line appearances
Paul Rodan wrote: I don't think the nature of these phones would allow for such a thing. It was designed for transfers and such, to be a real PBX, not like having 4 phone lines from BellSouth and multiple 4 line phones. According to the admin manual, the phone supports shared call appearances (SCA) using the SUBSCRIBE-NOTIFY method in the 'SIP Specific Event Notification' framework (RFC 3265). The events used are: - 'call-info' for call appearance state notification - 'line-seize' for the phone to ask to seive the line (Polycom Administrator manual, version 1.3.0, page 135) The trick is that we don't offer shared lines, per se, but shared extensions. So, if there are 2 phones in the same office, they can share the same 2 extensions, allowing transfers without transferring, if that makes any sense. Only one party can have the call at a time, but any phone can see the status, and seize the line if the call is on hold or ringing. I couldn't imagine SER/Asterisk/any SIP proxy or program doing what is needed. This behavior is available from Cisco's Call Manager. The only idea I had to get asterisk to do it would be have the calling party thrown into a conference room right away, and then have it ring all the other phones. Whoever answers it would then be put into the conference room with the calling party. But I think the trick is, whenever a person calls in, they get thrown put into a conference room, and then the PolyCom's all have to auto-answer and place the calls on silent hold, so that everybody is thrown into the conference room. That shouldn't be TOO hard to rig, but how do you get all the phones to ring as well until somebody picks up, so that there is at least 1 active person in the conference with the calling party. Then any other phone should be able to bust in simply by taking that line off of hold. That might work, but there has to be a more elegant solution. Good luck with that :-) ___ 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] RE: Polycom IP 300 PoE?
Kevin P. Fleming wrote Noah Miller wrote: IP 300's don't support PoE even though their brochures say they do. Has anybody have firsthand experience with them? Is this true? None of the Polycom phones support PoE directly, but all of them support it via an external PoE adapter cable that Polycom makes available. It's about $40 retail, though. Close... The IP600s do support PoE directly. According to the PolyCom rep I spoke with, the new IP300s are shipping with the accessory cable which supports PoE. If that's true, then their brochure is right on. You may wish to have the reseller you are working with contact PolyCom and ask... That was the only way I was able to get in touch with PolyCom, and the way I was assured I would have the cables. If you're interested, I can let you know when they come in if they were indeed in the box. ___ 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] Possible to display which extensions are in use on the phone's display?
From: Steven Critchfield [EMAIL PROTECTED] On Wed, 2004-11-17 at 11:49 -0800, Tracy R Reed wrote: On Wed, Nov 17, 2004 at 01:13:57PM -0500, Noah Miller spake thusly: On our current phones (Iwatsu) we have a button on the phones for each extension that lights up when that This seems to be a popular request these days. Most places I've seen call this shared lines I thought this was impossible with Asterisk, And it seems to be something the developers are not interested in supporting. Whenever someone asks about this feature they are normally told that this is a feature of small-office key systems and that Asterisk has its sights set on bigger systems so this functionality is not pursued. Correct, it isn't on a list for TODO as it doesn't scale to the size of an average office. Also it is only supported via some SIP phones and proprietary digital key set phones for which we will not get access to the supporting cards. So time is much better spent aiming for something that can be attained. -- Steven Critchfield [EMAIL PROTECTED] If you want to talk about features, scaling, etc., we should look what the big commercial VoIP systems are offering. The one that I have experience with is Cisco's Call Manager. We used shared call appearances to allow our secretaries (excuse me, Administrative Assistants) to catch their boss's calls and screen them, if they were there. While the lines were not available to everybody, shared appearances were a very highly sought-after feature. I agree with Steven about not spending a lot of time to make things work with proprietary stuff. When standards are closed, there is no guarantee they won't change overnight with the next firmware release, and therefore all of the code wasted. But not all ways of handling shared call appearances are proprietary. According to the Polycom IP manual, the SUBSCRIBE-NOTIFY method (RFC 3265) is supported on the phone. Using the hint alluded to before, it appears that it should allow multiple phones to view the state (call-info bit for call appearance state notification). All that remains unknown (to me at least) is if the button can be used to take over the conversation (the line-seize part to ask to seize the line). For references on this, refer to the Polycom SIP Administrator Guide v. 1.3.0, pages 31,135,111,113,117-8. If these features are already implemented, then we need to let everyone know. I think this might be the right way to give the features people are requesting while still preserving scalability. In big systems, each extension would be, well, an extension. If someone wanted Asterisk to act like a key system, then lines and extensions would have a one-to-one correspondence, with a shared appearance on all phones. A happy medium. Thoughts? ___ 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] RE: Sending DTMF Digits for DID
I have a legacy PBX that is currently connected to an T1 carrying incoming DID channels (DTMF, not PRI). I'd like to install Asterisk with two T1 cards in between the PBX and the telco, and use it to split off a new block of DID numbers. The remaining (old) DIDs would need to be regenerated by Asterisk and sent on to the PBX. Basically, I want to do what David Gomillion describes: http://www.loligo.com/asterisk/misc/nortel-asterisk-0.2.pdf but without PRI (the legacy PBX doesn't support PRI without $$). Can Asterisk originate DID signalling. (Maybe it is as simple as dialing on a outgoing DID line?) I could be wrong (it's happened plenty of times), but as long as the systems connect, then in theory you should be able to just dial the number. For instance, suppose Zaptel group 1 connects to the PSTN, and group 2 connects to your PBX. You should be able to have something like: [fromgroup1] exten = 1024,1,Dial(Zap/g2/1024) This would then pass through the DID to the next line. I only have PRIs available, and only have 1 non-production server, so I don't have a very good test bed available. Can someone with a T1 test this and let us know if it works as expected? Thanks, David Gomillion ___ 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
