Re: [asterisk-users] Voicemail Configuration
Thanks, that's given me some ideas. I don't think I can totally roll my
own, since I also make use of the MWI features of voicemail. Another
thread pointed out the existence of minivm, which I hadn't realized was
available. I just need to find the time to play around with some of the
proposed options.
John
On 5/14/2011 10:46 PM, virendra bhati wrote:
this will help you..
; DIY VOICEMAIL
*[ck987_vm_record]*
;start recording after the beep. Press # when done.
exten = 1,1,Playback(/home/ck987/asterisk_sounds/vm-record-start)
;build this call's recorded message file nameuniqueID_phone number
;every call is assigned a unique id.
exten = 1,n,Set(record_file=${UNIQUEID}_${CALLERID(num)})
; records into my vm_msg folder.
;Ends if # is hit, silence for 2 secs, or recording lasts for 60 seconds
exten = 1,n,Record(/home/ck987/asterisk_sounds/vm_msg/${record_file}.wav,2,60)
; 2 to review message, 3 to re record, or hang up
exten = 1,n,Background(/home/ck987/asterisk_sounds/vm-record-end)
exten = 1,n,WaitExten(5)
exten = 1,n,Playback(/home/ck987/asterisk_sounds/bye)
exten = 1,n,Hangup()
exten = 2,1,Playback(/home/ck987/asterisk_sounds/vm_msg/${record_file})
; press 1 to re-record, or hangup if satisfied
exten = 2,n,Background(/home/ck987/asterisk_sounds/vm-record-again)
exten = 2,n,WaitExten(5)
;no response, hang up on person.
exten = 2,n,Playback(/home/ck987/asterisk_sounds/bye)
exten = 2,n,Hangup()
;go back to the record option
exten = 3,1,Goto(ck987_vm_record,1,1)
;--DIY VOICEMAIL ADMIN--
*[ck987_vm_admin]*
; pass is 9988, jump to n+101 if authentication fails, expect 4 digits
exten = 1,1,Authenticate(9988,j,4)
; get number of voicemail messages
; SHELL function returns the output from a system command
;ls -1 lists visible files in a list.wc -l will count how many lines
there are. 1 line per file!
exten = 1,n,Set(num_messages=${SHELL(ls -1
/home/ck987/asterisk_sounds/vm_msg/ | wc -l)})
;you have...
exten = 1,n,Playback(/home/ck987/asterisk_sounds/you-have)
exten = 1,n,SayDigits(${num_messages})
;...messages!
exten = 1,n,Playback(/home/ck987/asterisk_sounds/messages)
;get file names.sed command trims off any .wav exten.tr command trims off
whitespace and line feeds.
exten = 1,n,Set(file_names=${SHELL(ls -m /home/ck987/asterisk_sounds/vm_msg/
| sed's/.wav//g' | tr -d' \n')})
;start message counter
exten = 1,n,Set(msg_counter=1)
; 1 to repeat message, 3 to go to the next message, 7 to go to the previous
message
exten = 1,n,Playback(/home/ck987/asterisk_sounds/msg-options)
exten = 1,n,Goto(ck987_vm_play_message,1,1) ; playback loop
;try again if password is wrong
exten = 1,102,Goto(ck987_vm_admin,1,1)
*[ck987_vm_play_message]*
exten = 1,1,Background(beep)
exten = 1,n,Set(current_message=${CUT(file_names,\,,${msg_counter})})
exten = 1,n,NoOp(${file_names} ${current_message} ${msg_counter})
exten = 1,n,Background(/home/ck987/asterisk_sounds/vm_msg/${current_message})
exten = 1,n,Goto(3,1)
; next message: add 1 and go back to the top of the loop
exten = 3,1,Set(msg_counter=$[${msg_counter} + 1])
exten = 3,n,Set(msg_counter=${IF($[${msg_counter}
${num_messages}]?${num_messages}:${msg_counter})})
exten = 3,n,Goto(1,1)
; previous message: subtract 1 and go back to the top of the loop
exten = 7,1,Set(msg_counter=$[${msg_counter} - 1])
;make sure number never goes below 1
exten = 7,n,Set(msg_counter=${IF($[${msg_counter} 1]?1:${msg_counter})})
exten = 7,n,Playback(/home/ck987/asterisk_sounds/previous_message)
exten = 7,n,Goto(1,1)
;delete message
exten = *,1,System(rm /home/ck987/asterisk_sounds/vm_msg/${current_message})
;number of files has changed. reload number of files and file names
;ls -1 lists visible files in a list.wc -l will count how many lines
there are. 1 line per file!
exten = *,n,Set(num_messages=${SHELL(ls -1
/home/ck987/asterisk_sounds/vm_msg/ | wc -l)})
;get file names.sed command trims off any .wav exten.tr command trims off
whitespace and line feeds.
exten = *,n,Set(file_names=${SHELL(ls -m /home/ck987/asterisk_sounds/vm_msg/
| sed's/.wav//g' | tr -d' \n')})
exten = *,n,Goto(1,1)
it's the actual voicemail example which is use by asterisk it self
On Tue, May 10, 2011 at 2:53 AM, John Marvin jm-aster...@themarvins.org
mailto:jm-aster...@themarvins.org wrote:
On 5/9/2011 3:08 PM, Roger Burton West wrote:
You could use Monitor to record the whole call, then use an AGI
to do
something with it on hangup if the other conditions haven't been
satisfied...?
I understand how to do the first part, and I at least understand
that I could do something fancy with the AGI capability. But what I
don't know is how I can take the recording and insert it into a
voicemail box such that it can be retrieved through the normal
VoiceMailMain mechanism
[asterisk-users] Voicemail Configuration
Hi, I can't figure out a way of achieving what I want to do with the voicemail feature. I thought I'd ask here to see if there are any creative solutions that I have not considered. What I want to do is have a message that says Press 1 for Dick, or 2 for Jane. Then, depending on which number is pressed, have the caller sent to the appropriate voicemail box. I know how to do that without any problem. I want to keep repeating that message after a timeout, i.e. not send the caller to a default voicemail box if nothing is pressed. I can handle that also. However, I want to record what is said during that time and send it to a third voicemail box once the caller hangs up without having pressed 1 or 2. I want this ability in order to handle robot callers. I'm not interested in most robot calls, but sometimes they contain useful information (school closures, your order is ready ... etc.). The problem is if I just timeout and send the caller to a 3rd mailbox I will usually have lost the beginning of the message, and I don't want a short timeout because I want someone to be able to listen to the prompt and make a proper choice (if they're not a robot). Any ideas? Is this possible? Thanks, John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Configuration
On 5/9/2011 3:08 PM, Roger Burton West wrote: You could use Monitor to record the whole call, then use an AGI to do something with it on hangup if the other conditions haven't been satisfied...? I understand how to do the first part, and I at least understand that I could do something fancy with the AGI capability. But what I don't know is how I can take the recording and insert it into a voicemail box such that it can be retrieved through the normal VoiceMailMain mechanism. Would the asterisk voicemail app dynamically notice something new being dropped into the voicemail mbox directory? Would it only be noticed once Asterisk is restarted? Most importantly, would it send out the notifies to the phone associated with that voicemail box? I can probably fake the last part if necessary, but making the voicemail retrievable through the normal voicemail mechanism is what I really need to achieve. Thanks, John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP ban list by country
One possible advantage of the fact that IANA has depleted its pool of /8's (class A) is that if you are only filtering at that level the data is static now. It should never change again for IPV4. John On 2/13/2011 11:54 PM, Steve Edwards wrote: On Mon, 14 Feb 2011, Bruce B wrote: What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. I compiled this list a few (6?) months ago by typing class A address blocks into Arin.net's 'whois' web page and noting which Regional Internet Registry it was allocated to. http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks After plonking this into a couple of production hosts, attacks of all ports dropped dramatically. I note there have been changes since then (128.0.0.0 was assigned to RIPE back in November), so if anybody wants to 'refresh' and post changes, please do. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registering with a static peer?
I've been using a couple of Polycom 501 phones in my home Asterisk setup. I set up each phone in sip.conf to be static, i.e. host=phone ip address so that registration wasn't required. This has worked fine for me for a couple of years. Now I just bought a Polycom 335. Since the 501's are now obsolete, I had to go through the steps required in order to have separate sip.cfg's, etc. Anyway, I got the Polycom 335 to almost work, except for the dialplan. When picking up the handset (or hitting the line 1 key) it put me in url-dialing mode, i.e. it prompted me with Enter URL and didn't follow the dialplan I had set for the phone. It took me quite some time to figure out what was going on. I finally found a paragraph in the phone's users guide that said: URL or IP dialing is not supported on registered phones. If your phone is intentionally unregistered, the default dialing mode is by URL. At least that explained the cause of the problem. I could not find a way of changing the default dialing mode for an unregistered phone. So, I figured the only answer was to register the phone. Just setting the phone's register parameter to 1 (voIpProt.server.1.register=1) fixed the problem, i.e. the phone would now prompt me with Enter Number rather than Enter URL and allowed normal numeric dialing, according to the dialplan. But of course, if I turned on verbose logging in Asterisk, it complained about the phone not needing to register. The only way I could get it to shut up and properly register was by specifying host=dynamic. I'd rather not do that, so I'm currently using the hack of telling the phone to register and it failing every 30 seconds. This doesn't seem to break anything, but I can't say I'm really happy with the solution. So, a couple of questions: 1) Does anyone know of a way to disable the default URL dialing mode for an unregistered Polycom 335? 2) Is there a way of configuring asterisk to allow static ip address registration in sip.conf? I can understand that this doesn't make much sense, but it appears the Polycom phone really wants to be registered. I prefer the static configuration since it is a little more secure and reliable in my opinion. 3) Assuming the answer is no to 1 and 2 above, is there something I am overlooking with my hack solution? Should I just go ahead and set the phone up for dynamic registration? John ___ -- 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] Persistant Dialing
Is there any way to get Asterisk to keep redialing automatically if a number is busy? How about the case where it is very difficult to get through, due to a large number of people trying to simultaneously call the same number? In particular, I am asking these questions for an outgoing SIP connection. Is there a way via SIP to have multiple simultaneous calls in progress, each attempting to call the number, and then stopping all of the extra's when one succeeds in getting through? A related question would be (i.e. if I decide to write some custom code for this application): what happens when the destination number is on a crossbar or step office? I'm assuming that when the destination is VOIP or a more modern POTS switch that the busy condition is reliably detected and a proper SIP status is eventually returned. But if the destination is crossbar or step is it possible that the connection will actually be made in order for me to hear the remote office generating a busy signal? John ___ -- 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 power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets
Stephen Bosch wrote: 3. I thought I might save some clutter by putting these cables between the midspan and the patch panel, but then I discovered that the male end of the cable is keyed, just as in the default AC cables provided with the phones, meaning that they'll only work if plugged directly into the phone itself. The reduction in clutter with this set-up is, unfortunately, not what I had hoped, though anything is better than nothing. I imagine it would work if I sanded away the plastic post on the connector, but that says nothing about how it might behave if a non-compliant device were plugged into it. Better safe than sorry. Actually, I did exactly this with the default AC cables. I plugged my Polycom wall warts into my UPS near my household patch panel, filed off the tabs on the AC cables and used them as patch cables between my (non POE) switch and the patch panel. I use a standard patch cable for the phones. I don't think this would be a good idea for an office environment. I'm not sure what would happen if I plugged something other than one of my Polycom phones (501's) into the non standard powered ethernet jack. I'm fairly safe in my home environment, since I am usually the only one messing with ethernet cables in the house, and I have told my family specifically not to ever unplug one of the phones in order to plug in something else (they can always plug into the back of the phone instead if they need a temporary connection). John ___ --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] Zaptel version for asterisk 1.2.16
Kevin P. Fleming wrote: There is no need for any 'map'; any Asterisk 1.2.x release should be usable with any Zaptel 1.2.x release, but of course we'd suggest using the latest releases of both. There are no API changes or feature additions (generally) in release branches, so frequently you can update _only_ Asterisk if you are happy with the version of Zaptel you have installed and running. This may be true most of the time, however, Asterisk 1.2.16 won't build without a recent version of zaptel headers (codec_zap.c has missing dependencies). John ___ --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] Building a new voicemail system... Testers needed!
Olle E Johansson wrote: Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. ... I've wanted to do exactly this for quite some time, but I haven't had the bandwidth to be able to do any work on it. This idea makes perfect sense. I've always felt that the current voicemail implementation was too restricted and didn't allow enough customization. Anyway, I don't have any time to help or test right now (perhaps I will in a couple of months), but I just wanted to at least thank you for doing what you have done already, and encourage you and others to continue this work. Thanks! John ___ --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] Cordless SIP Phones
Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn’t cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. It depends on the features you are looking for. The Aastra probably has the best support of VOIP features on the handset. The additional handsets are a little pricey ($99), and only four are supported (which is probably more than enough in most cases). If you don't care about VOIP features in the handset, the Uniden UIP1868 might be a good option. This can be purchased in an unlocked version from various online voip equipment sellers (e.g. www.voipsupply.com). The advantage of the Uniden set is that it uses the same handsets as their 5.8 Ghz cordless POTS phones, which means that you have a variety of handsets you can use (including a waterproof/submersible handset), and they are cheaper. The UIP1868 also supports up to 10 handsets (probably way more than you want, but the feature might be useful in some situations). There's also a much better chance you can buy a new or replacement handset a couple of years down the road. Otherwise, as others have suggested, you might consider just using an FXS adapter and using an analog cordless phone system. This is what I am currently doing (although I am seriously considering buying the UIP1868). The main problem I have with this solution is the delays introduced by caller id, especially if you want distinctive rings based on the caller. With an analog system asterisk has to send the caller id between the first and second ring. Then, at least with Uniden phones the handsets won't ring at all until the caller id has been received (if you've enabled the distinctive ringing feature). In my house the cordless phone typically doesn't start ringing until the SIP phones have already rung twice. One final note. It is possible to unlock the vtech 8100. Do a google search for cyt35. CYT is a program that was written to unlock various TI AR7 based devices, and it is known to work with the vtech 8100. I have no experience with this myself, but you might want to look into it if you still have the vtech 8100, and you are not planning on using it with Vonage. John ___ --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] Is there any Asterisk controllable thermostat?
Doug Crompton wrote: John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. Similar, in terms of a wireless transmitter -- no. But they have both a serial and a USB computer controller that works over the power ($50-$70 for the controller). It works for both X10 and Insteon protocols. Why isn't that acceptable? And yes, some essential X10 replacements are not yet available. I have two of the X10 spotlights myself. But Insteon has a lot of interest from a lot of companies, so I expect to see a lot more variety in the next year or two. Note: There is opensource software available for the controller, so you don't have to pay the extra $70-$200 or more for the various non opensource software packages available. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Well, the answering machine was a digital one that had multiple (3) VM boxes. It had a separate message waiting light for each box. That is the feature that my wife misses. I'm not sure what if any SIP phones provide multiple message waiting indicators. Besides, it is a moot point for me at this time, since I've already finished building the hardware, now it is just a simple matter of programming to get it to work. ___ --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] Is there any Asterisk controllable thermostat?
Doug Crompton wrote: I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --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] Dialing from Placed Calls on Polycom IP501doesn't always work
Noah Miller wrote: I never ran 1.6.6 for any length of time. 1.6.7 and 2.0.1 don't seem to suffer this issue. 2.0.1 has some buddy watch problems, so you may not want to use it, but 1.6.7 should be OK. I've been running 1.6.6 for quite a while, and I have been quite annoyed by this bug. However, the release notes for 1.6.7 did not mention fixing this problem, so I did not have any motivation for upgrading. But, since you said that you did not see the problem on 1.6.7 I decided to upgrade and see if the problem was fixed. It appears to have fixed it, although I can't be sure yet, because sometimes a call placed from the placed calls list did work on 1.6.6, so I don't have enough of a sample size yet to be sure the bug is gone. I sure hope it is. Thanks for the info. John ___ --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] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
Jay R. Ashworth wrote: On Thu, Oct 26, 2006 at 01:00:18PM -0400, Henry.L.Coleman wrote: My position is that there are only two reasons for wanting to change an outbound CID: 1. to deceive the called party 2. to validate the calling party I don't know how much notice people take of CID but obviously if if it can be used to mis-represent or for fraudulant purposes then it will become useless. People take *lots* of note of it, or they wouldn't be thinking about a bill. As far as I'm concerned, the issue just now is can I trust the CNID handed to me by my LEC on my PSTN connection. That is, and really, can only remain, the province of the LEC's and IXC's that send calls there. Firewalling incoming CNID from PRI's and VoIP carriers is the business of people who accept incoming end-user calls; it has to be done there in a network architected the way the PSTN is. There seems to be an underlying assumption here that each outgoing call is always tied to an incoming phone number. This isn't always true for the PSTN, and it is certainly not true for voip. For my home Asterisk setup I have a single PSTN line, and then I use a variety of different voip providers. I use two different providers for my DID's (one toll free, and one normal). I use yet a different provider for terminating outgoing calls. So, when making an outgoing call via voip, what number should I use to identify myself? I currently use the number of my PSTN line, since that is our public inbound number. If you have your way, what kind of controls would be put in place? Each voip provider would have to verify that I have the right to use a particular number, even if that number isn't provided by that provider? How would they know if I got rid of it? Would I even be allowed to do what I am doing now? How much is this going to cost the voip termination providers to properly administer this? John ___ --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: Choice of soundfile format
Martin Joseph wrote: Transcoding is a bigger hit then mixing as i understand it. If all the conference members are using ulaw for example, then having the playback material encoded in ulaw is the big winner. If there are different codecs connecting, then there is a lot of decoding/mixing/recoding that will need to occur. There is a misunderstanding here. The only time transcoding is not required is when data is just being passed through, either from a data file (recording, Music on Hold, Voicemail, etc.) or from one channel to another, AND the format is the same. However, once mixing is required, i.e. for a conference, transcoding will always be required because you can't mix compressed data. All channels have to be decoded to sLinear, mixed, and then reencoded. John ___ --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 2.01 sip issues
James Andrewartha wrote: Jessee J Holmes wrote: As far 1.6.7 firmware supporting multiple presences (48 i think), maybe I was wrong on that; however, I remember reading the 2.0.1 firmware release notes and they mentioned that feature was fixed within the 2.0 firmware. Maybe they fixed it before that and just never documented it or maybe I misread it. If it works through in 1.6.7, great! Thanks Douglas. The release notes for 2.0.1 say: 2.2 Version 2.0.0 (Beta Release Only) Note: The 2.0.0 Release does not include the changes and corrections from SIP releases 1.6.6 and 1.6.7 Note that the notes for Version 2.0.1 says it does include all the changes from 1.6.6 and 1.6.7. The notes for 2.0.0 shoulw probably say does not include ALL the changes, because it does include some. It specifically mentions the 48 buddies feature, which then leads some people to assume that the feature was new for 2.0.0, whereas you are correct in that it was actually introduced for the first time in 1.6.6. And 1.6.6 is the first release to support 48 watch buddies on the 601. FWIW I'm still using 1.6.6 here. The only problem I've noticed is that pressing dial from one of the call lists (when accessed via the arrow keys) doesn't always work. Yes, I am running 1.6.6 on my 501, and I have noticed that bug also. Its not just when accessing it via the arrow keys. It doesn't work sometimes when you access via numeric keys or the select softkey and then explicitly pressing the dial softkey. I see no reference in any of the release notes to this bug being fixed. I wonder if Polycom knows about it? John ___ --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] Extension Numbering
Norbert Zawodsky wrote: quote ... (Well, almost. Extensions must be shorter than 80 characters long, and you shouldn’t use single-character extensions for your own use, as they’re reserved.) ... /quote O.k. - This answers my first question (if there is disadvantage if we use only 1 digit extensions). But what for are single-character extensions reserved ? As others have mentioned the single character extensions are reserved for special extensions in the dialplan, i.e. 's' for start, 't' for timeout, 'i' for invalid, etc. However, the single digit extensions are not reserved, so you can use them if you want. The only problem with using single digit extensions is the reduced flexibility in your dialplan, i.e. things like: 1) no room to expand your number of extensions without having a mix of 1 digit and 2 (or more) digit extensions, 2) having to have a prefix number for dialing an external number, etc. Of course, if you don't have a dialout digit (e.g. '9') then you have limited flexibility also. Since I don't have a dialout digit and I allow both 7 digit (NXX) and 11 digit (1NXXNXX) dialing, I don't have that many options for extensions either (they either have to start with 10, 11 or 0). Right now I'm using three digit extensions starting with 10 and 11 (i.e. 10X and 11X) but I am running out, and am thinking of moving to three digit extensions starting with 0 (i.e. 0XX, except 011 would still work for international dialing). I don't have any use for just 0, since I don't even know what dialing an operator should mean in the context of a mixed PSTN and multi provider VOIP setup (not to mention I haven't called an operator of any type in over 20 years). John ___ --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] Extension Numbering
Lacy Moore - Aspendora wrote: I have a suggestion regarding dial plan. When I first started I saw no reason to have to dial 9 first for outside calls. Because I wanted to be able to dial out from the missed calls list, I chose to eliminate the dial 9 requirement. I'm now regretting it, primarily because 711 is a valid number, but also a parking spot. I can handle 911, and I can handle 411, but 711 is going to be a pain. Just wanted to give you something to consider when you are starting out. Something I didn't consider. Why not just move your parking spot range to something else? It doesn't have to be the default 701-720. John ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
The problem is that most people aren't going to be able to answer this question without trying it. Most voip providers (including Teliax) advertise a rate to all New Zealand Mobile service providers, i.e. +64 2, not specifically +64 21xxx. Note, I just tried a +6421 mobile number via Teliax from the U.S. and it worked. So either 1) Teliax can't reliably connect to those numbers, 2) They can connect to some subset of those numbers, or 3) they fixed something since you last checked. John ___ --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] sound file length
Race Vanderdecken wrote: If it is a wav file you can take the size of the file in bytes and subtract 44. This is a hot button of mine. This is a heuristic that may work for many wav files, but is not true for others. A wav file header can be larger than 44 bytes, the wav data doesn't have to start right after the header, and many wav files contain meta data after the wav file data (and techically meta data could come before the audio data, but I've never seen that before). Some very poorly written wav file playing programs will play right through that meta data as if it was audio data. John ___ --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] Looped message playback
Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Well, I don't have a solution for the general case (looped playback), but if you are only playing a sine wave, couldn't you use Playtones() instead? It has the ability to play a tone indefinitely until you tell it to stop. John ___ --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] A Caller ID question (UK)
Gordon Henderson wrote:
Heres an oddity. Call comes in on a ordinary BT POTS landline, and in the
UK, (to my knowledge!) the caller ID can be one of 3 things: The number,
withheld and unavailable (which usually means an international call
from a non caller-id co-operating telco, or someone in the UK using some
el-cheapo telco to place outgoing calls)
This works OK. On my DECT phones I get the right thing on the display.
It's basically the same here in the U.S., except that it is normally
Private instead of withheld.
Now stick an asterisk box inbetween the incoming line and the DECT
phone,and arrange for incoming calls to be forwarded to the DECT phone.
(TDM400P card)
Now, the display on the DECT phone still displays the right thing - ie.
the number, withheld, or unavailable. But Asterisk internally just gets an
empty ${CALLERID} variable with either withheld or unavailable calls.
(When a number is presented it gets the correct number)
What version of Asterisk are you running? I'm very surprised that the
phone still is able to differentiate between withheld and unavailable.
In Asterisk, at least version for 1.2.10 which I am running, here is how
callerid is handled by chan_zap.c (the code that supports your TDM400P
card and my Sangoma A200D card). When a call comes in on the incoming
line, the low level code determines if the CLID is private (withheld),
unavailable or there is a number present. If it is private or
unavailable it sets a flag indicating that and then it removes the
privacy/and or unavailable indicator from the number string. But then
the calling code loses the information in the flag field so from that
point the only thing you have is whether or not the number field is
empty or not.
Then, when that caller ID information is passed to your phone, the code
determines if the number string is empty, and if so it sets the
unavailable indicator. There is no provision for setting the privacy
indicator because that information is lost.
That's why I'm surprised that your phone still appropriately indicates
withheld when it is hooked up to your asterisk box. Have you actually
tested that particular scenario? If it does work then I am interested in
the version you are running, because I want a solution to this also. I
want to see if this is something that used to work and has changed, or
if one of the later versions fixed this.
Anyone know how to distinguish between the 2 inside the dialplan? (if it's
at all possible)
It's not possible as far as I can tell without modifying the code. You
could do a hack in the low level code that would replace the name with
Unavailable or Withheld and replace the number with some illegal or
known not to work numbers, and then use that in the dialplan.
I'm not sure what the right fix would be. I've thought about mapping the
flag (before the information is lost) into the channels CallingPres, and
then mapping that back when passing the information to the phone. But
I'm not sure of what all the side effects of doing that would be.
Perhaps then you would be able to look at the CallingPres in the
dialplan to determine the difference between the two, i.e. prohib vs.
unavailable.
John
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Re: [asterisk-users] Polycom related question
Kevin Smith wrote: Here is what the configuration looks like for one of the phones, the other is 284: [283](Empire-Defaults) [EMAIL PROTECTED] [283a](Empire-Defaults) [EMAIL PROTECTED] [283b](Empire-Defaults) [EMAIL PROTECTED] So actually you are trying to use one phone to monitor (receive notifies for) multiple boxes. It looks like the Polycom's have some support a different mwi for each registration, but I'm not sure how well it works. You didn't specify what username you specify for each config above, so I don't know if the notifies are going to one registration or to different registrations. The messages button on the phone only seems to show the status of one registration, but the indicator light seems to combine the different results together (and you can't clear the light with the clear button since that only applies to one of the registrations). Of course that assumes that you are sending the notifies to different registrations on the phone -- all bets are off if you are sending them to the same registration (which is controlled by the username value) since Asterisk is treating them as separate phones the notifies will collide with eachother. You would get more reliable behaviour if you did as Rich suggested and just specified something like this for just the [283] config: [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED] In that case Asterisk sums up the total messages in each of the boxes and the messages button on the phone will show you that total rather than the results for only one of the boxes. The polycom documentation is not very clear on how multiple mwi's are supposed to work, so I'm not sure what the right answer is. John ___ --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 Native Sound Distortion (ulaw)
shadowym wrote:
I found that the distortion was consistent. In other words it happened in
the same way at the same time in a particular file. I suspect it has
something to do with how Asterisk plays it back and not any sort of
hardware/IDE/interrupt issue. Kris, the developer of Astlinux didn't seem
to have any ideas why it would not work as well on Asterisk either.
If the distortion is consistant as you say, then you are probably seeing
the same problem I found a workaround for.
I was disappointed in the sound quality of the gsm files, so I was happy
to find the Native Sounds files. However, I then ran into the clicks and
pops when playing them. Reading some of the earlier comments in this
discussion someone mentioned an issue with Asterisk not padding files to
even 20ms increments when playing them. So, although that may be a bug
in Asterisk, I thought I would see if that was the problem by writing a
quick C program to pad all my ulaw files to multiples of 160 bytes.
Voila, all clicks and pops were gone. So, I don't know if that is the
only issue, and perhaps there are other problems people are having, but
padding the files fixed the issue for me. Obviously this should be fixed
in Asterisk.
If anyone else wants to try this experiment I've enclosed the simple C
program I wrote below. If you compile it and call it padulaw here is how
I fixed all the files:
find /var/lib/asterisk/sounds -type f -name '*.ulaw' | xargs padulaw
This program could be easily modified to pad .sln files to a multiple of
320 bytes (the files would be padded with 0x rather than 0xff).
John
#include stdio.h
#include fcntl.h
#include sys/stat.h
#include sys/types.h
#define ULAW_SILENCE 0xff
#define MS20_BYTES 160
unsigned char silence[MS20_BYTES];
void pad_file(char *);
main(int argc, char **argv)
{
int i;
int nfiles;
if (argc 2) {
fprintf(stderr,Usage: %s file name ...\n,argv[0]);
exit(1);
}
nfiles = argc - 1;
for (i = 0; i nfiles; i++) {
pad_file(argv[i+1]);
}
exit(0);
}
void
pad_file(char *fname)
{
int fd;
int i;
struct stat sbuf;
int filesize;
int remainder;
int nwrite;
fd = open(fname,O_WRONLY|O_APPEND);
if (fd 0) {
fprintf(stderr,Could not open %s for writing.\n,fname);
return;
}
if (fstat(fd,sbuf) != 0) {
fprintf(stderr,Could not stat file %s.\n,fname);
return;
}
filesize = (int) sbuf.st_size;
remainder = filesize % MS20_BYTES;
if (remainder == 0) {
close(fd);
return;
}
nwrite = MS20_BYTES - remainder;
for (i = 0; i nwrite; i++)
silence[i] = (unsigned char)ULAW_SILENCE;
if (write(fd,(void *)silence,nwrite) != nwrite) {
fprintf(stderr,Write Failure on file %s\n,fname);
return;
}
close(fd);
return;
}
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Re: [asterisk-users] Polycom related question
Rich Adamson wrote: If you look at the sample configs, you'll find: [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in the sip.conf.samples for v1.2 stable. That is the only way that I know of to turn on the mwi for two different phones (eg, extensions). Is that what you're using and its not working? I think that is the opposite of what Kevin is trying to do. The above config is for one phone monitoring multiple voicemail boxes. Kevin wants multiple (two) phones monitoring the same mailbox, i.e. he is probably specifying the same mailbox within the config for each of the phones that will be monitoring that mailbox. I'm not sure why there would be any problems with that. Kevin, have you tried just having one phone at a time do the monitoring, to make sure there aren't any problems with the phone's config? When one misses a notification, is it always the same phone that misses it? It's interesting that the problem is intermittent, it would seem that if Asterisk doesn't support this that it would only notify one phone each time and that the results would be consistant. John ___ --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 related question
Rich Adamson wrote: Phones don't monitor mailboxes. One needs to tell asterisk which phones are to be notified when a voicemail is left, and the sip statements above are the only ones that I'm aware of to accomplish that. Yes I am aware of that. Perhaps I chose the wrong wording, but I clarified it in the next sentence by talking about specifying the same mailbox in two different configs. On many phones, there is only one mwi function. If Kevin has one extn (eg, 111) on a phone set up with a mwi and then a second extn (eg, 222) on the same phone set up for mwi, one extn's mwi might turn the indicator on while the second extn will turn it right back off again. Since I don't recall Kevin saying what type of phone he's using, I can only guess that might be the problem. Yes, I can easily see that some phones won't behave well being notified for two different mailboxes. I'm not sure how a Polycom phone would handle that (which is the type of phone he is asking about). However, that does not appear to be the problem. Again, he is talking about two different phones being notified for changes to one mailbox, not one phone being notified for two different mailboxes. Your example was something like this: [101] [EMAIL PROTECTED],[EMAIL PROTECTED] which would mean that extension 101 should be notified for changes to voicemail boxes 1234 and 1233. That still would result in Asterisk only having to notify one phone when a change was made to a particular mailbox. He is doing (or should be doing) something like this: [101] [EMAIL PROTECTED] ... [102] [EMAIL PROTECTED] i.e. extensions 101 and 102 should be notified about changes to voicemail box 1234. His message appeared to be fairly clear that extensions 101 and 102 were two separate phones, rather than two different lines on the same phone. This should not be a problem with the phone unless there is a configuration issue (which is still certainly a possibility), since as far as the phone is concerned it doesn't know about the other and there should be no difference in the protocol. But it is possibly a bug in Asterisk, i.e. it may not properly handle having to notify more than one phone when a change is made to a particular mailbox. John ___ --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] What don't I get about SIP?
Mike wrote: Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. Actually, as soon as you hit 8 you will get the fast busy. Is that your full dialplan? What about an emergency (911) or other N11 calls? What about direct dial international calls (011...)? When I dial 9-555-5 and wait, nothing happens So, it looks like what you want is a global dialing timeout in the phone, which the Polycom phones don't appear to have once you break dialtone. But you may be able to kluge the digit timeout to give you that feature if you don't need it for what it is meant for. Right now you are using it to timeout when a digit other than 1 is pressed after the 9. That isn't really necessary (unless 91 followed by 9 digits is actually a valid number for whatever you are doing with it). Also, you are using the brackets unecessarily, since you only have one digit within them. An equivalent dialplan that doesn't use the digit timeout feature would be: digitmap dialplan.digitmap=7xx|9[2-9]x|91xx The digit timeout feature is typically used for direct dial international calls and calling the operator. If you don't need either of those then you could do something like this: digitmap dialplan.digitmap='7xx|9[2-9]x|91xx|[79]x.T dialplan.digitmap.timeOut=15/ which would timeout and send whatever sequence you had pressed after 15 seconds if you hadn't already matched one of the other patterns. Note that asterisk may possibly respond with error code 484 if the sequence pressed isn't complete, which would make the phone continue to ask for more digits. So, the other part of the solution is to add: exten = _X.,1,Congestion() to extensions.conf in the context you are using for your polycom phone(s). That will match anything that doesn't match one of your valid extensions as long as it is two digits or more. So you still will get the behaviour you don't like if someone just presses 7 or 9 and nothing else. But it will give you most of what you want, assuming I understand what you are looking for in the first place (you could try x.T in the digitmap and _. in extensions.conf, but _. is likely to cause other problems). Note: When the Polycom gets the congestion response from Asterisk it plays the congestion tone for only about 3-4 seconds, and then hangs up, which is different behaviour from when you press an 8 for instance. If you want the behaviours to be similar you could do something like this: exten = _X.,1,Answer() exten = _X.,2,Playtones(congestion) exten = _X.,3,Wait(30) exten = _X.,4,Hangup() John ___ --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] What don't I get about SIP?
Mike wrote: Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? It's the opposite. Asterisk always uses the most specific match for an extension, i.e. anything that matches _1XXX will take precedence over _, but if it matches _12XX that will take precedence over _1XXX, etc. John ___ --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] What don't I get about SIP?
Mike wrote: It certainly makes sense, and I tried it...it works, you are right. So what do you make of this page : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Interesting. I got my information from Asterisk: The Future of Telephony (in the dialplan chapter). Perhaps the wiki page refers to 1.0 behaviour, and 1.2 behaviour is what is defined in Asterisk: TFOT? My experimentation so far has shown the Asterisk: TFOT information to be correct. I haven't played around with #include, which the wiki says can change the dialplan extension sorting. I may have to experiment with that to see if it has any effect. I hope that what I said is correct regardless, because it makes the most sense and is less likely to cause weird issues when changing the order of #includes, etc. John ___ --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] IP phone with 2 ethernet jacks
Matt Birmingham wrote: Can you supply any information/links as to how you got this working? Sure. I followed the instructions here: http://www.voip-info.org/wiki/view/Asterisk+presence However, there was one important fact missing. By default the presence feature is turned off in the Polycom's (at least that was the case with my Polycom 501's). It needs to be turned on before any of the other settings take effect. To turn it on you can edit sip.cfg. Look near the end for the feature section. Feature 1 is presence, changed feature.1.enabled from 0 to 1. This is also covered in section 4.6.1.23 of the Polycom admin manual. John ___ --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] IP phone with 2 ethernet jacks
Mario wrote: We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. This feature isn't missing from Polycom phones, although it may not work the same as what you want. But the phones do support the presense feature, and I have it working on my IP501's with Asterisk. John ___ --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] How to set externip in sip.conf automatically?
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for Asterisk
to access a Linux environment variable containing the IP information in
the form of ${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Actually, I don't think you read his response carefully enough. He was
giving you a method of doing it automatically.
But first, lets dismiss the environment variable solution. I haven't
played with using environment variables in Asterisk, so I can't help you
there. But I do know about environment variables in general, and you
cannot use them to solve your problem. Environment variables are not
global, i.e. if you change one it does not effect the value in all
currently running programs. The environment (all of the environment
variables and their values) is inherited from the parent process (it is
passed in by the kernel on the new processes stack when the process
first starts). After that, only the process itself can change its own
values in order to pass on a changed value to a child (but again, only
when that child process is started, i.e. a parent cannot affect the
environment of an already running child process). In summary, you can't
change the values of Asterisk's environment variables after Asterisk has
already started. The values that Asterisk sees are the values that it
inherited from its parent process, i.e. most likely the rc scripts that
started Asterisk when you first booted the machine.
Now, back to the solution proposed by Brad. He was in effect proposing
that you dynamically change sip.conf. However, parsing a rewriting
sip.conf automatically is kind of ugly, but luckily Asterisk supports
#include. So, he suggested that you can periodically generate a file
with a single line in it, i.e. externip=xx.xx.xx.xx and then use
#include in your sip.conf to include it (i.e. sip.conf doesn't have to
ever change). The final part of the solution is to make Asterisk reread
sip.conf (and the included dynamically created file at the same time).
You can do that with:
asterisk -rx sip reload
which you can put into the same cron job that you currently are using to
refresh /etc/myip.
John
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[asterisk-users] No retry after DNS failure
Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never recovered from that, i.e. it never retried so those providers were unavailable. The only provider that was still available was one that I had entered the IP address for, rather than the host name. Have any of you run into this issue, and if so, how have you solved it? It seems that since Asterisk periodically tries to reregister it should also retry the DNS lookup at the same time, rather than never trying again if the lookup fails. This would indicate that Asterisk would also fail if the voip provider changed the IP address of its server because Asterisk would never see the new IP address. Here are some workarounds I thought of, but none of them are particularly good: 1) Get a UPS so my machines won't reboot when the power fails. This actually might not solve anything, because I'm connected to a remote DSLAM in my neighborhood that I believe does not have backup power, so it won't work when the power is out. But perhaps Asterisk is more robust after it has booted (I'll have to test this). 2) Change all host names in sip.conf to IP addresses. This is kind of ugly and also will break when a voip provider changes their IP address. There is a reason for DNS! 3) Have a cron job send asterisk periodic sip reload commands. 4) Delay the start of asterisk until the internet connection has come up. This could cause me to be without any phones if there is any delay or failure in bringing up the network (I also have zap channels and PSTN lines). 5) A hybrid of ideas 3 and 4 above: Have a startup script that waits for the internet connection to come up, and then sends a sip reload command to Asterisk. Any other ideas? John ___ --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] No retry after DNS failure
James Harper wrote: Are you sure that it was Asterisk? Did you try an nslookup after the network came up while Asterisk wasn't working? How long did you leave it before taking matters into your own hands? Fairly sure. I didn't realize there was a problem until about 6 hours after the power outage. Local calls go out via the PSTN, so I didn't notice the problem until I tried to make a long distance call. It took a few minutes to diagnose the problem. It was when I did a sip show peers and noticed it was missing my outbound voip providers that I realized there was a problem. A sip reload immediately brought them back. DNS will do negative caching as well as positive. If sip.sometel.com failed to resolve (because it doesn't exist or because your network wasn't up), your caching dns server or resolver or both may remember this as a 'negative cache' entry, so that it remembers that it doesn't exist. I'm fairly sure there is a different error response between name doesn't exist as opposed to a failure to connect to an authoratative name server in the chain. For example if the name servers responsible for .com said that sometel.com didn't exist, or the nameservers for sometel.com said that sip.sometel.com didn't exist then that should be negatively cached. But if the name servers for .com said that the IP address for the name servers for sometel.com are xx.xx.xx.xx and then your dns server can't connect to xx.xx.xx.xx then a different error response is returned, and I don't believe that is or should be negatively cached. If your asterisk server is responsible for the network connection, then the ifup/pon script should take care of flushing the dns cache for you for exactly this reason. If you have another server or router that is responsible for the connection then the server running asterisk will have no idea that its negative cache is no longer valid. Well, you may have hit upon a reason why others may not see this problem. The asterisk server is not responsible for the connection, an external DSL modem/router is responsible. However, I run a local DNS server, so it is available to Asterisk. Perhaps Asterisk is more robust when it can't contact the DNS server, as opposed to it being able to contact the DNS server and the DNS server responds with an error. Perhaps Asterisk doesn't differentiate how it behaves depending on the type of error. John ___ --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] No retry after DNS failure
Rich Adamson wrote: If memory serves correctly, most of the above has been raised as issues in the past and the suggested work around has been to run a dns caching server on the asterisk box. That's exactly what I am doing, unless you mean that the dns caching server caches results over a reboot. FWIW, I always use IP addresses instead of dns names. But, I don't have to deal with dynamic ip changes of any device either. I don't have to deal with any dynamic IP changes either, other than the possibility of one of my voip providers changing the IP address of their servers, which is probably a very rare occurence. John ___ --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
