Re: [asterisk-users] Capture queue agent drop and put caller back in queue
Mitch Claborn mitch...@claborn.net writes: Shouldn't asterisk somehow know when the agent disappears? You are a bit out of luck since SIP session timers, the obvious solution, cannot be set lower than 90 seconds. rtptimeout set to e.g. 10 seconds may work, but you need to then set rtpholdtimeout higher and hope that voice activity detection or mute does not kill the calls... Does Asterisk consider RTCP packets for the purpose of rtptimeout? That could solve the problems of silence suppression and mute. /Benny -- _ -- 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] Capture queue agent drop and put caller back in queue
Christopher Harrington ch...@acsdi.com writes: Since nobody seems to have come up with an Asterisk-specific solution, it sounds like the real approach here is something more generic. You can set up Nagios to fire off an event if it detects endpoints or infrastructure are suddenly dead. In particular, Nagios could launch a program written for this purpose, passing the endpoints it detects are missing, and that program could then query Asterisk via AMI about the call IDs each endpoint is a participant in, then do a forced-transfer to a dedicated queue that announces the failure condition to the caller. This AMI could also conveniently remove the dead endpoints from the existing queues (including the failover queue). Can a Nagios-based solution provide quicker failover than the 90 seconds provided by sip timers or the 10-30 seconds provided by rtptimeout? /Benny -- _ -- 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] Capture queue agent drop and put caller back in queue
Un-topposted Eric Wieling ewiel...@nyigc.com writes: Using qualify=10 ? qualifyfreq=10 is fine, but Asterisk will not AFAIK do anything to a call just because the peer goes unreachable qualify-wise. You are still stuck with running a script that listens to qualify-unreachables and does the appropriate thing to the calls. It is doable and a valid solution, but not something I would be very happy with personally. I still think that rtptimeout is the appropriate solution. Combine it with the g option to Dial, and you can redirect the call wherever you want. The problems are still: can it be made safe against mute and silence suppression, and does HANGUPCAUSE or similar get set to a useful value when a call is hung up due to rtptimeout? /Benny -- _ -- 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] Top Posting
isr...@gmail.com writes: Just my pitch in to post From a blackberry you can only top post there is no way of bottom posting So if I would have to wait to get to a computer to bottom post I would just never answer Just delete the original post then. Not including context is perfectly fine, it is easy to go to the parent article as long as the post includes correct headers. Your post had the proper headers. /Benny -- _ -- 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] Top Posting
Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. It is often a quick way to see if it is worth responding to someone. If they top post, nothing of value is likely to come out of the conversation. So by all means, everybody who wants to, keep top posting. /Benny -- _ -- 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] Noise on phones while speaking...
Carlos Chavez cur...@telecomabmex.com writes: I have a new install and the customer is complaining that they hear noise on all calls, no matter if it is internal or external, desk phones or softphones. The noise is only present when the user is speaking, not the remote side. The remote side does not hear the noise, only the local user. If you record the call (with Monitor or Wirehark), does the noise show up on the recording? /Benny -- _ -- 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] USB FXS device
Jeff LaCoursiere j...@sunfone.com writes: Nifty! Love this Raspberry Pi. I keep thinking of new things I want to do with it. If I could only clone myself. I have a video doorbell project at the top of the list, if I don't find a USB FXS device :) The Raspberry Pi has some problems with USB cameras and other USB video capture devices (due to the USB host controller). Hopefully the problems will be solved soon, but until then it is probably not the ideal device for that purpose. /Benny -- _ -- 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] USB FXS device
Jeff LaCoursiere j...@sunfone.com writes: The basic question was has anyone made a USB FXS device work with asterisk. Now that I have additionally defended my architecture decisions, can anyone actually answer the question? The Open USB FXS project is exactly what you want. It seems to be discontinued. Depending on volume, it might be worth resurrecting the project -- it looks like the price could get reasonable if you need a few thousand... It does not seem like there is anything commercially available right now. /Benny -- _ -- 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] Bypass queue wrapup time
Mitch Claborn mitch...@claborn.net writes: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? Slightly hacky solution which only works for ringall: Designate a phone to be in the queue but never get answered. When you are ready for a call early, do a directed pickup of that phone. For a less hacky solution, see https://reviewboard.asterisk.org/r/1619/ /Benny -- _ -- 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] Bypass queue wrapup time
Olivier oza_4...@yahoo.fr writes: That's the point : to me, casual @pickupmark mechanism don't work with calls that entered into a queue : the extension rings but you can't pick the call up with a directed pickup. (For general pickup, that's another strory). (and I would be very pleased to be wrong) That seems to be fixed a long time ago, if I read the various issues correctly. I haven't actually tried it. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Streaming MeetMe Conference
Adam K. Dean a...@dmcip.com writes: Hi, I was wondering if anyone has any experience in streaming a MeetMe conference so that others might listen in to it? It would be nice if the audio format could be AAC, but at first any format will do. I did come across this: http://www.voip-info.org/wiki/index.php?page_id=991 Which looks interesting, but if anyone knows of a better way I would be interested! There's an example using Ices here: http://www.757.org/~joat/wiki/index.php?n=Main.HomebrewAsteriskConferenceManager Search for Streaming the conference. I'm not sure there is a better way that Ices; I think it's a pretty cool way. /Benny -- _ -- 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] One leg in a conference and adjusting stream volume of other leg
Markus unive...@truemetal.org writes: thanks! I wasn't clear enough in my original mail. What I meant is: the volume of the stream that a user is listening to is adjusted, but the volume of the conference itself is not changed! That means, a conference is going on, and everyone is listening to the same music at the same time, but when the music becomes too loud or annoying, a user can individually adjust the volume of his music, while the volume of the speech of each user, basically the conference itself, remains the same. Your requirements are such that the only solution is to mix the audio for each participant individually. This is a rather expensive operation and not supported in either of the Asterisk conference applications, AFAIK. I can only think of one way of doing that: give each member their own conference and bridge one leg of that conference into the main conference. How to accomplish that is a bit beyond me though, but perhaps others can help. All the mixing is likely to cause a strain on your hardware, and the sound quality could suffer. /Benny -- _ -- 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] Suggestion of Server Specifications for Asterisk
Leandro Dardini ldard...@gmail.com writes: A single sata disk will be an unacceptable single point of failure. Get three disks and get in raid5 configuration. You'll gain in safety and speed. RAID-5 is slower than single disks when it comes to write IOPS (a commit is not done until the slowest disk has answered). Avoid it for write heavy workloads at all costs unless you are writing sequentially in one file with write caching enabled. /Benny -- _ -- 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] Suggestion of Server Specifications for Asterisk
Steve Edwards asterisk@sedwards.com writes: Won't 200 simultaneous calls result in a lot of 'head thrashing' that would be avoided by staging the recordings to some form of non-mechanical storage and then copying the the recording at the completion of the call? The extX family of file systems probably will not be clever enough to try to unfragment the writes; they will likely all end up in one long fragmented stream. This is exactly what you want if you do not expect to actually listen to more than a small fraction of the recordings. If you do expect to listen to most of the recordings, copying them after they have been written is a great idea. Especially if you have an off-peak time where the disk is idle anyway, or by placing them on RAM-disk if you can afford to lose some. Frankly I would not worry too much, just use reasonably modern hardware with support for write barriers, enable write caching, use a not-too-clever filesystem, and go RAID-1 on just two disks. You could make a ridiculously fast system with NILFS2 and SSD though... /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Kevin P. Fleming kpflem...@digium.com writes: I've just looked into this a bit, and I don't see how using connect() would actually solve the problem. If we receive a UDP datagram from a SIP endpoint, we could use socket() and connect() to create a socket specifically for sending to (and receiving from) that endpoint in the future, but we can't specify the source address to be used by that socket. The only way I know of to specify the source address for outbound packets is to use a raw socket and compose the IP header ourselves, which would be overkill. You just bind() to the source address you want to use for outgoing packets. I have just tested it, it works here at least. The tricky bit is knowing which source address you want to use. That you can get from IP_PKTINFO, somewhat portably. Once you have a socket with connect() and bind(), the full 5-tuplet of protocol, srcaddr, srcport, dstaddr, dstport is defined, and all further traffic related to that connection should be going to that socket. However, in between the time that the first packet arrived and that socket is up and running, more packets may have been queued on the original server socket. This cannot happen with TCP because accept() is atomic (the client cannot send more data before the three-way-handshake is done), but there is no such luck with UDP. /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Raj Mathur (राज माथुर) r...@linux-delhi.org writes: Precisely. In fact, if a packet from 192.168.2.n is received on /any/ interface, the response will always go out from the 192.168.2.X interface. (Barring some weird routing/iptables configuration, of course.) This is only the case for TCP, because TCP accept() fixes the whole five-tuple of protocol, srcaddr, srcport, dstaddr, dstport. UDP does not have an accept() equivalent and most applications just use sendto() which lets the OS pick a source address. /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Kevin P. Fleming kpflem...@digium.com writes: That's quite interesting; can you describe a scenario where this occurs? Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on. Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same socket for answering. That would tell the system IP stack that this is in fact a connection, and so the system would ensure that the reply source IP would be correct. Alas, few programmers are aware that you can even do connect() for UDP, and I believe it would be a rather large change to the Asterisk SIP stack to pass connected sockets around rather than just remembering IP addresses and port numbers. (Admittedly I haven't looked at that code in ages, so I could easily be wrong). The workaround is to explicitly bind to 192.168.1.1. Since Asterisk can bind to precisely one address, that kills off IPv6. /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Kevin P. Fleming kpflem...@digium.com writes: I must be missing something. If a phone sends a UDP packet to 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 interface on the Asterisk server? The easiest way is that the Asterisk server itself is the router. Phones on 10.0.2.0/24 have 10.0.2.1 as default gateway. Another option is that there are no real routers; the phones are disconnected from the Internet, they still have 10.0.2.1 as default gateway but they only use it to reach the Asterisk server. The only way I can imagine that happening is if a router in between the phone and the server has been told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems like a bizarre way to construct a network. I do not feel that the two scenarios above are particularly bizarre. Getting replies from Asterisk *back* to the phone would also require the IP stack on the Asterisk server to route those replies back over the 10.0.2.0/24 interface instead of the 192.168.1.0/24, which doesn't make any sense either. If the phone is on 10.0.2.0/24, the IP stack will route packets to it directly through the 10.0.2.1-interface by default. It actually takes quite serious contortions to make it send the packets elsewhere. Servers with multiple interfaces are a bit out of the ordinary. Right now Asterisk is difficult to work with on such servers, which is not a large problem for Asterisk in general, because they are so rare. You can always work around the problem either by creative routing or by explicitly binding to one address. chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. It would be greatly appreciated :) It is low priority for the Asterisk project, as there are always workarounds. Extra points for making Asterisk support IP addresses appearing and disappearing... That would make VRRP/HSRP failover work. (It works if you bind to 0.0.0.0, but it is difficult to get Asterisk to use the VRRP-address as the source address for outgoing packets). /Benny -- _ -- 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] Help choosing the right card
Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. /Benny -- _ -- 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] CDRs on multiple servers.
Owais Ahmad millennium@gmail.com writes: Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. cdr_adaptive_odbc handles multiple servers. Just define several with [foo] and [bar] and it Just Works. /Benny -- _ -- 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] using Wifi smartphones as SIP clients
Bart Coninckx bart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- 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] using Wifi smartphones as SIP clients
man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I don't know that particular phone, but dedicated wifi phones definitely CAN work for professional use. E.g. ASCOM phones work absolutely great, they are just expensive. /Benny -- _ -- 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] Set SIP peer state busy
Jonas Kellens jonas.kell...@telenet.be writes: I know you can do this by pressing DND on your IP-phone, but can this state also be set in the dialplan ? You cannot actually achieve this by pressing DND on your IP-phone. All that will accomplish is that the phone answers all calls with busy, but Asterisk will keep bombarding it with calls if it happens to be member of a queue. This can be detrimental to the health of the Asterisk-server. /Benny -- _ -- 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] Rate sheet normalization
Markus unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
Jason W. Parks jason.w.pa...@gmail.com writes: Thanks for the info. As we move forward, we'll be testing and making a phone selections. No doubt we'll run into this. Are you saying if the phone is stated to be a 10/100 phone, it still may not work at 10? I must admit it isn't something I have looked at particularly deeply. I hav experienced that Snom 3xx phones have not linked up at 10Mbps a few times and I have never seen them successfully achieve link at 10Mbps. This could be a firmware issue or a general issue with the quality of those 10Mbps networks. Or Snom simply does not allow their phones to connect to a 10Mbps network. In each case the network installations and switches were improved so 100Mbps worked without any problems. /Benny -- _ -- 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] Cordless SIP phone
Carlos Alvarez car...@televolve.com writes: Perhaps you meant one that is not wi-fi? I would agree with that, I have a pile of totally useless wi-fi phones, they are all garbage. Ascom has some fantastic Wi-fi phones. They are expensive, but they are the only Wi-fi phones I have tried which actually work as phones. They manage to get a signal in places where other devices give up, and if they lose signal they are quick to regain it when you get closer to the access point. The cost is a challenge in most installations; DECT often comes out cheaper even if you include SIP DECT base stations. /Benny -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Olivier oza_4...@yahoo.fr writes: I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). I see the CURL function in Asterisk 1.8.7.1, found in the res_curl module. In Fedora it is available in a separate package called asterisk-curl. If you do not get res_curl, it is likely because a prerequisite library was not installed when you built Asterisk. Try looking at MENUSELECT_DEPSFAILED in menuselect.makeopts. /Benny -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Olivier oza_4...@yahoo.fr writes: 1. But, on your own 1.8.7 system, do you have something related to CURL when typing core show functions (or core show applications) ? I'm asking because func_CURL is missing from https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions (asterisk 1.8 version) which is misleading. == 8 == ursa*CLI core show version Asterisk 1.8.7.1 built by mockbuild @ x86-02.phx2.fedoraproject.org on a x86_64 running Linux on 2011-10-17 21:15:10 UTC ursa*CLI core show function CURL -= Info about function 'CURL' =- [Synopsis] Retrieves the contents of a URL == 8 == The Wiki documentation is sadly not perfect yet. /Benny -- _ -- 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] Best non polycom SIP conference room phone
Luke Hamburg l...@solvent-llc.com writes: Carlos- Sorry if this is too much of a digression but this piqued my interest as I've been pretty happy with Polycom in my limited experience (haven't used the SPAs much, just Yealink Polycom, and an occasional Snom here and there). If the config files were not the issue for you, then what _were_ the problems? A button has been pressed. Polycom must reboot for the change to take effect. Reboot now (Y/N)?. Yes it's a recycled Windows joke, but it applies much better to Polycom than it did to Windows. It is IMHO a bit mean to use Polycom's in the Asterisk exam; the difficulty of passing the exam is quite high if you haven't worked with them before. Pretty much anything else is quicker to get to basic working state. Of course, once you get provisioning working they are excellent phones. /Benny -- _ -- 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] Anyone have a reliable T.38 Solution
David Backeberg dbackeb...@gmail.com writes: Thanks for clearing that up. I was getting all excited that I could flash the PAP2T; I've always used regular voice tones over SIP with the PAP2Ts. SPA-2102 supports T.38. If you ignore the WAN-port, it is practically a PAP2T. The only time you cannot ignore the WAN-port is when doing provisioning. /Benny -- _ -- 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] DTMF fun
Tom Browning ttbrown...@gmail.com writes: My question is this: Is Asterisk simply relaying the client's DTMF signalling untouched or do I need to look at Asterisk more closely and turn some knobs. I would recommend that you grab some wireshark traces before and after the DTMF traverses Asterisk. It should be fairly easy to verify whether Asterisk changes the length or number of DTMF messages. /Benny -- _ -- 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] Binding asterisk to two static IPs
ge...@riseup.net writes: Any idea how to solve this? You can control src address selection with with ip route command. E.g. if you know that you want to reach 192.168.0.0/24 with a source address 192.168.0.50, you can do: ip route change 192.168.0.0/24 src 192.168.0.50 scope link dev eth0 (You need to change the IP addresses and device name of course) This may enable you to use bindaddr=0.0.0.0 /Benny -- _ -- 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] Trouble with *8 Pickup
Paul Hayes p...@provu.co.uk writes: If you time the *8 just right so it is being handled during the end of the Dial then I got: [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL Does this happen when using the Pickup() application as well, or is it specific to *8? /Benny -- _ -- 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] Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)
Benoit Panizzon benoit.paniz...@imp.ch writes:
Is there a way to get asterisk not to invent a CALLERID(name) if there is
none?
Id did try to set ${CALLERID(name)=} but that resulted in From: sip...
and the displaying of this empty string on the subscribers phone.
I believe you have hit issue 17451,
https://issues.asterisk.org/jira/browse/17451
There is a patch for 1.6.2.15 attached to the issue.
/Benny
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Re: [asterisk-users] Asterisk binaries on CentOS version 6
A J Stiles asterisk_l...@earthshod.co.uk writes: Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even take long anymore (on any target system with the grunt to run Asterisk). The only thing to beware of is, if configure complains that you need a package that you already have, then you need the corresponding -devel package. There are advantages to packaging systems though. E.g. you never end up with an outdated module causing trouble in a newer version. You can probably make your own RPM's for CentOS 6 based on either the Digium RPM's for CentOS 5 or the Fedora RPM's for Fedora 15. Just download the source RPM and rebuild it; if you don't get any errors you are generally golden. For extra points, install mock and let that do the rebuild. /Benny -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
Kevin P. Fleming kpflem...@digium.com writes: OT: Take a look at 'systemd'; this is exactly what's happening there, and Fedora is likely to incorporate it into Fedora 16, and it will make its way into other distros after that. It was incorporated into Fedora 14, and it is the default in Fedora 15... /Benny (And yes it meant I couldn't boot after upgrading to Fedora 15. It couldn't handle that I had the cgroup file system mounted on /cgroup in fstab.) -- _ -- 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] benefits of asterisk 1.8
Paul Belanger pabelan...@digium.com writes: Sounds like asterisk was not told to generate a coredump, add the following, then you can generate a backtrace[1]: asterisk.conf [options] dumpcore = yes The challenge with Asterisk and core dumps is that the Asterisk user often does not have permissions to write to the directory it has as current directory. By default, that is where the kernel writes the core dump. You can change the directory by changing the kernel.core_pattern sysctl, but make sure that you pick something which does not present a security threat. It would be very convenient if Asterisk could be told to keep a specific directory as current directory. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. /Benny -- _ -- 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] receive faxes
A J Stiles asterisk_l...@earthshod.co.uk writes: (For my part, I'm actually surprised that nobody came up with a proper protocol for encapsulating the stream of zeros and ones that make up a fax transmission but rely on the precise timing inherent with a circuit-switched network, into something more suitable for sending over a packet-switched network. That would have fixed it good and proper.) It is called T.37. However, asynchronous protocols like T.37 are a bad match for the expectations people have of faxes. /Benny -- _ -- 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] MWI not working on most ATAs in Asterisk 1.6.2.17
maill...@lightspeed.ca writes: We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. My testing of 1.6.2.17 as well as the svn branch of 1.6.2 a few weeks ago indicated that MWI was fairly broken. I managed to get it working somewhat reasonably on Snom phones with a combination of subscribemwi=no (despite the fact that the Snom phones subscribe for MWI!?), pollmailboxes=yes and pollfreq=30 (despite the fact that we have nothing but Asterisk touching the voicemail files). In the testing, I managed to get Asterisk to flood the phones with tens or maybe hundreds of MWI's by simply leaving one voice mail. No, I have not had time to file bugs, we simply did a fall back to 1.6.0.28 + a lot of patches. It was not the most serious bug anyway; the larger problem is that Asterisk deadlocks. /Benny -- _ -- 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] From 1.4 to 1.8: stdexten issue
Mathieu Chouquet-Stringer math...@csetco.com writes:
I've googled and pretty much tried all forms of the syntax but I've yet
to make it work. For instance I tried not including stdexten and
calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
either...
stdexten in the default extensions.conf seems to only handle extensions
with at least 2 digits...
/Benny
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Re: [asterisk-users] Multiple SIP endpoint registrations
Kevin P. Fleming kpflem...@digium.com writes: Why do you need a Local channel to do this? If extension 234 exists in some context, the Dial() statement in that extension can dial SIP/234-foo and SIP/234-bar itself. Good point. It can be a bit of fun keeping track of the phones when they are added to or removed from queues, and the owner expects both of them to be added/removed at the same time. It is still doable without Local channels. Once you need to do manipulation of calls before passing them on to the phone (change callerid individually, handle tT options etc.), Local is unavoidable, but at that point multiple registrations would not work either. /Benny -- _ -- 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] Multiple SIP endpoint registrations
--[ UxBoD ]-- ux...@splatnix.net writes: Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? No. Our solution is to give each phone an account and make a Local/234@somecontext which dials SIP/234-fooSIP/234-bar. There are some challenges with Local channels, and we are working with Olle E. Johansson to get some of them resolved. One of them is that in some cases Asterisk cannot do reinvite if Local is done without /n. /Benny -- _ -- 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] Testing from where number is...
Piotr Górski pi...@prnet.pl writes: So how to bill customers? Number portability makes it pretty impossible... In the US, you pay the same to call a cell phone as you pay to call any other phone. The callee pays for the airtime. This is a sensible arrangement, as it allows for number portability and price competition. Alas, Europe chose to pass the costs onto the caller, without even making it reasonably possible for the caller to know whether he is calling a cell phone or not! The Danish number plan in particular is completely insane. /Benny -- _ -- 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] DTMF and Snom
Jonas Kellens jonas.kell...@telenet.be writes: Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. Which phone model? If 870, you may want to look at this thread: http://forum.snom.com/index.php?showtopic=4084 You may want to experiment with a different firmware version. It is a bit surprising that they do not allow you to set the DTMF duration and volume. /Benny -- _ -- 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] Hide the plain text password
ken...@gnat.com (Richard Kenner) writes:
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains just a string.
- A CLI command core encrypt string is added to Asterisk. It takes the
provided string, encrypts it using the string in master_key, and outputs
a string of the form {enc:encrypted_version_of_string}.
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert strings in password fields -- otherwise you risk false
positives in e.g. the dial plan.
I can recommend contracting with one of the indepedent Asterisk
developers to get this done. You will likely find them on the
Asterisk-biz-list.
/Benny
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Re: [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Tilghman Lesher tilgh...@meg.abyt.es writes: Correct; and Asterisk needs to be started as root, even if it will drop privileges after startup. Do this, and there should be no problems. Starting as root + dropping privileges is fine. Running configure as root is not so fine; that basically makes building RPMS impossible. /Benny -- _ -- 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] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Sorry for resurrecting an old thread... Tilghman Lesher writes: Out of curiosity, what platform are you running on? On most platforms that are able to run Asterisk, with the possible exception of Solaris, increasing the maximum file descriptor for use with select(2) is possible. I am not entirely sure yet, but it looks like Asterisk 1.8.x fails to increase the maximum file descriptor when running on Linux, if configure is not run as root. If configure is run as root, everything works as expected. /Benny -- _ -- 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] ReceiveFAX issue.
David Backeberg dbackeb...@gmail.com writes: So you're saying if you turn off t38 in sip.conf, you receive faxes successfully? Problem solved. Don't use T.38 in your particular environment. That is not particularly useful advice. Fax over VoIP without T.38 is inherently unreliable except in very controlled environments. That a few faxes happen to work does not make T.38 a bad choice. It is in the interest of the Asterisk community to fix whichever bug/incompatibility Bryant Zimmerman is hitting. Of course in reality no one with the right skills may have the time to do so. Again, that does not make the problem solved. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Michelle Dupuis mdup...@ocg.ca writes: Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com (also on the voip wiki) You get the cluster/heartbeat replication without needing to add openSER or full HAlinux. A simpler approach - easier to config and manage How do you handle replicating voice mails? /Benny -- _ -- 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] Need to pick your brain for recommendation on using 2.5 or 3.5 HDDs for Asterisk server...
bruce bruce bruceb...@gmail.com writes: Other than the price difference (2.5 is more expensive and can't find many of the 1TB or so) is there any preference, advantage, or disadvatage of chosing 2.5 HDD or 3.5 when it comes to the server operations or Asterisk operation? There is no difference. Pick the server which offers the disk bandwidth and I/O's per second which you need. Do you really need 1TB disks? If you do, be careful what you place on those disks. Reading e.g. a voice mail or a speak off a large slow platter which is busy writing CDR's does not sound good at all. /Benny -- _ -- 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] rtp problem with 1.8.0-rdc1
cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /Benny -- _ -- 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] rtp problem with 1.8.0-rdc1
cov...@ccs.covici.com writes: But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being unable to handle that other devices do. If you switch to 1.6.2.x and enable internal-timing, you should have a shot at getting it working. /Benny -- _ -- 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] Purpose of qualify=yes
Chris Owen ow...@hubris.net writes: So I guess my question is what is the real purpose of the qualify setting in a non-NAT situation and can one safely set the qualification as something higher. I'd think something like 15 seconds would be more than enough for BLFs and the like. The purpose is simply to see if the phone is available. For your particular use it is likely best to simply turn it off completely. If a phone disappears, its registration will eventually time out anyway. /Benny -- _ -- 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] Force ip disconnect after register?
Bryant Zimmerman brya...@zktech.com writes: Is there a way to force the connection to drop and reconnect after let's say 50 attempts. Most firewalls have tools for removing specific connections from the connection table. Alternatively a switch to SIP/TCP might help, but I've never tried SIP/TCP with Asterisk so I don't really know what state it is in. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with new AEX800 card dying because of interrupt problems
Christian Weeks c...@weeksfamily.ca writes: Hello I purchased an AEX800 card to replace the ageing cheap channel bank/T1 card solution a few months ago, assuming that it would be a more robust solution for my small scale phone system. However, it appears to be anything but that. Originally implemented as a XEN dom-u virtual machine on a large server class machine, using PCI passthrough to pass the AEX800 and a small older TDM400, then recently migrated to the dom-0, the aex800 has continued to experience interrupt errors: wctdm24xxp :04:08.0: Missed interrupt. Increasing latency to 8 ms in order to compensate. wctdm24xxp :04:08.0: ERROR: Unable to service card within 25 ms and unable to further increase latency. Can you do a at /proc/interrupts? /Benny -- _ -- 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] A way to check against a list of numbers?
Hose hose+aster...@bluemaggottowel.com writes: The most straightforward way would be to just define explicit patterns. Obviously that works, but doesn't seem scalable in terms of maintenance. I don't think that maintaining the list in the dial plan is all that bad, actually. Dump it in its own context and file... If that isn't convenient enough I'd go for the Asterisk database next. Also on the option list is private e164/enum or an SQL database. /Benny -- _ -- 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] PostgreSQL is asterisk friendly with it?
Bryant Zimmerman brya...@zktech.com writes: As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Yes. The ODBC drivers don't really care which database you access. Is there any down side to PostgreSQL over mySQL or will it be a big win? The only issue we have with Postgres is the dump/reload cycle when upgrading database version. This is being fixed in the latest versions though. Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? It works fine from Windows as well. /Benny -- _ -- 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] CDR on Transfer...
Carlos Chavez cur...@telecomabmex.com writes: I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensión the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11). You can set a TRANSFERCONTEXT. In that context you can try to use ForkCDR and its companions to get the records right. If you come up with a setup which acts perfectly no matter the scenario I would be happy to hear about it. Note that TRANSFERCONTEXT is not invoked when the phone does a SIP redirect before the call is answered, AFAIK. Notice that it's been a long time since I battled with this part of Asterisk, and I didn't check that I remembered correctly. This will all be a lot more sane with Channel Event Logging in 1.8.x, but at that point you need to run mediation before you get CDR's you can use for billing. /Benny -- _ -- 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] CDR on Transfer...
Please don't top-post. Geraint Lee gera...@gmail.com writes: to get accurate cdr's i just use a border server to send every call through that logs cdr... doesn't matter how many times it gets transferred internally the border server still gets accurate records of the whole call. That is what we do too, but customers are requesting CDR's which include information about e.g. which specific phone answered the call. This information is unknown to the border servers. We provide customers with access to the CDR's generated on their particular virtual Asterisk, but we receive complaints about the deficiencies of the 1.6.x CDR's. It is particularly troublesome that dial plan changes often change CDR's. With Channel Event Logging we should be able to provide all the information which customers ask for and at the same time insulate them from dial plan changes by logging only the information we want in precisely the format we wnat. I look forward to that, even though it means a bit of work mediating the logs before presenting them to customers. /Benny -- _ -- 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] install asterisk
Alan Lord (News) alansli...@gmail.com writes: Fedora is *not* a server operating system and not one I would choose to run asterisk on. Fedora is an excellent server operating system. I manage more than a thousand installs if you count virtual ones. I would recommend using either CentOS or a Debian/Ubuntu Server build without X11 and all the other cruft that comes with a Desktop OS - which is what Fedora is. No one forces you to install X11 on Fedora. /Benny -- _ -- 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] spam blacklist
SIP s...@arcdiv.com writes: Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Don't ever bounce spam! You WILL get blacklisted for doing so. Either reject it in the transaction or drop it on the floor. /Benny -- _ -- 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] How to record and playback at the same time
Please do not top post. Sherwood McGowan sherwood.mcgo...@gmail.com writes: I'm going to go ahead and say that while I'm not one of the developers, I think it's safe to say that you cannot record to a file and play it back at the same time. Probably something like file locking (for the record, locks it from access by other processes, etc)... There is nothing in Unix/Linux which prevents the playback of a file while it is being recorded. File locks in Linux are purely advisory; it is up to the applications whether they choose to respect them. The only challenge is whether the header has been written correctly, and you should be able to do without that in a pinch. What happens if you copy the half-written file to a computer with speakers and try to play it through the speakers? /Benny -- _ -- 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] Random DTMF Tones Only on heard on ATA
Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Amazon Web Services
Randy R randulo2...@gmail.com writes: I'd think twice about trying this, taking into account the recent spate of attacks to so many of us coming from Amazon EC2 and particularly their answer to complaints, which was something like Deal with it. Indeed, my personal threshold for dealing with EC2 traffic has become if in doubt, ban it. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
Ira i...@extrasensory.com writes: running on Centos 5 with yum update showing it's all up to date. I think it's 5.2 or 5.4, I just don't know how to get it. cat /etc/system-release. Or redhat-release, if system-release doesn't work on CentOS yet. It should be 5.5 if you're all up to date. /Benny -- _ -- 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] Redial dtmf tones randomly...asterisk 1.4.21.2
I would appreciate it if you didn't top-post. das sandesh sandesh...@gmail.com writes: Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... How do you interface with the PSTN? A Digium card? Either way you may want relaxdtmf=no in dahdi.conf if you don't have that already. You can see the DTMF happening on the Asterisk console if you set verbosity high enough. /Benny -- _ -- 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] play alaw file with .wav extension
Quy Pham Sy qu...@vega.com.vn writes: they've just named as xxx.wav so I guess there is no problems with copying or linking solutions. You're simply lucky that the header is short enough to not sound too bad. /Benny -- _ -- 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] Redial dtmf tones randomly...asterisk 1.4.21.2
das sandesh sandesh...@gmail.com writes: In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Which end hears the DTMF, the SIP phones or the phones on the PSTN? When you say sent from the server side, is the server side the Asterisk or the Cisco? /Benny -- _ -- 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] MAC Address prefixes of Voip equipment
Frank Church voi...@googlemail.com writes: Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports? With Snom you can actually find the specific phone model from the MAC address. Unfortunately this information isn't published anywhere. Perhaps there would be community interest in maintaining a database for the various vendors? /Benny -- _ -- 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] Y-cords - What are they ?
Zeeshan Zakaria zisha...@gmail.com writes: making use of the fact that both Cat5 networks and BRI ports don't use all the 8 pins, so why not use extra wires in the cable for something useful instead of wasting them. For Ethernet, this is only true for 10Mbps and 100Mbps. Gigabit and up uses all four pairs. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Delay with remote stations?
William Stillwell (Lists) writes: I have several remote phones that experience a slight call delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? This is actually a somewhat common problem in SIP. One end sends media before the other end is ready to receive it, or a gateway receives media on one leg of the call but media isn't yet ready on the other leg... In your case I would guess that it is caused by firewalls/NAT reacting only to RTP traffic in one direction, thereby blocking traffic in the other until the first packet. Luckily it's IP, so you can use tcpdump or wireshark or phone-specific dump tools to capture the traffic and see where the problem hides. /Benny -- _ -- 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] Using Local in queues a good idea? (or at least not a very bad idea?)
Håkon Nessjøen haa...@avelia.no writes: But for a few years ago, I did some testing with Local/ channels, and they seemed somewhat unstable in large quantity. Are they more safe now? Is it safe to use local channels with the /n modifier as queue members? (i need the n modifier to be able to count continous calls using the h extension). They are pretty good in late 1.6.0.x at least. There are some issues though. E.g. Queue sometimes has trouble determining whether a given extension is busy or not, and therefore sends calls to a busy extension. When you return Busy in your Local channel and the queue is set to ringall, Queue just sends the call right back into the Local channel, causing load problems. Workaround: Detect whether the call comes from a queue with ringall strategy and do a Wait(1000) instead of returning Busy. /Benny -- _ -- 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] Silence suppression and internal timing
Is it possible to disable silence suppression by adding silenceSupp:off to the SDP Asterisk transmits even when Asterisk is using internal timing? As far as I can tell Asterisk stops sending silenceSupp:off when internal timing is on, which does make sense, but I would like to avoice silenceSupp for debugging purposes. The Asterisk version is 1.6.0.26 in case this matters. /Benny -- _ -- 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] Feature Request: GotoIfTimeWithOffset
Steve Edwards asterisk@sedwards.com writes: Wouldn't a set time function be more usefull? I really like that idea. Enough that I could try to lobby internally for funding, if you know someone who is willing to do the work... /Benny ___ -- 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] Random DTMF tones generated from speech in conversations
hbk fo...@online.no writes: Where to look for forgotten DTMF detection settings? Try relaxdtmf=no. sip show settings to check that it worked. /Benny ___ -- 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] iphone client app
Gavin Spurgeon gspurg...@dageek.co.uk writes: iSip (£2.39) http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8 I have been very impressed by the audio quality from iSip, at least from the other end so to speak. It shares the basic flaw of not being able to run in the background with every other iPhone app. They try to mitigate that problem with their Push service, if you give their server your passwords and allow them to access your Asterisk... /Benny ___ -- 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] Please some enlightment on ENUM !!
SIP s...@arcdiv.com writes: It may work in Austria, and may even be valid in Austria. But if that's the case, it's because Austrian dialing is a complete hack -- NOT because that's the way it's intended OR designed. Err no? It's perfectly sane, and it was intended and designed that way. You are providing no justification at all for your opinion that it is a hack. It is quite apparent where the hack is in this thread. /Benny ___ -- 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] Patch for app_dial.c: exit when just one ext is busy.
Leif Neland le...@neland.dk writes: Because I might have more phones than mouths :-) If I'm busy with one conversation, I don't want to hear another phone ring. I might have a desktop and a portable phone. This use case is indeed very difficult to implement in Asterisk dial plans today. Especially if you add the phones to queues. The only way is through Local(), but Local() in a queue has nasty side effects which I have expounded on in other threads... I hope you have luck in getting the patch fixed and accepted. With that done there are only a few more things to fix before we can do away with Local() in Queue(). /Benny ___ -- 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] Prevent Dial if any extension is busy
Leif Neland le...@neland.dk writes: I think a modification should be done around here to return busy if just one channel was busy (only enabled if an option on dial is set) in asterisk-1.6.0.15/apps/app_dial.c, line 610 That is doable, but it can result in a bad experience for the caller. The Dial() is likely to indicate progress to the caller, which means that the caller will hear the familiar dialing tone (By the way, is there a dictionary of the names for the various telecoms tunes?). Right afterwards they will hear the busy tone, as if the callee rejected the call. It is best not to send a busy tone once you have indicated that the call is on the way to being connected -- unless you're trying to get rid of a telemarketer. /Benny ___ -- 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] Restricting transfers between SIP phones
C. Chad Wallace cwall...@lodgingcompany.com writes: So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? This is impossible. At that point the phone has done this: 1) Put the original caller on hold 2) Made a new outgoing call At some future point the phone might drop the second outgoing call and go back to the first, or it might bridge the two in a transfer. You can't know in advance. The only way to achieve what you want is to never allow a call to a different department when the same phone already has a call on hold. This will however stop the (in some places quite common) practice of calling the other department to ask a quick question, then returning to the original caller. It could be somewhat tricky to implement as well, but it should be doable with call-groups. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crosstalk - Is there a debug option for logging this?
JT djklut...@gmail.com writes: I'm struggling with an intermittent crosstalk issue resulting in a caller's audio being broadcasted to other calls (only one way as they are unable to hear the others listening in). This may be a long shot... I have experienced this when two SIP phones had the same IP address (a bug by itself of course). Now, obviously the SIP phone should not just play any random audio that someone throws at it, but apparently life is not so simple. /Benny ___ -- 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] Experience with LLDP
Warren Selby wcse...@selbytech.com writes: I believe I spoke with Aastra and Snom at the Astricon tradeshow and they said they support it on their newer models as well. For Snom the enhancement request is SCPP-227, but I don't believe it has been implemented. I can't find it in any release notes at least. The general public can't track SCPP's, which is a bit inconvenient. /Benny ___ -- 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 audio leaking between calls
Tilghman Lesher tles...@digium.com writes: Many consumer-grade switches effectively turn into hubs when more than 1023 MAC addresses are seen on a network. This may be done intentionally by somebody attempting to eavesdrop on all network connections sent through the switch. A reboot of the switch might (temporarily) remedy the problem, but you'd be better off getting an enterprise-grade switch that does not exhibit such misbehavior. Even so, all network cards automatically drop all unicast traffic not destined to their mac address (or addresses). This is turned off when the nic is in promiscuous mode, but that shouldn't happen on hardphones. Also, it is highly unlikely that IP stack wouldn't drop the traffic itself. This is simply too basic to get wrong. The only way I can see that a low-layer network problem could cause crosstalk is if two phones somehow acquired the same MAC address. They would likely end up with the same IP as well, and that could certainly cause problems. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POTS 4K linear codec
Cary Fitch ca...@usawide.net writes: Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. You can't get a guaranteed bit-for-bit identical stream through SIP/RTP or IAX. You can pick the same codecs as the PSTN uses (Alaw or ulaw, depending on country), but jitter and packet loss still makes things like DTMF or fax/modem unreliable. For DTMF it is better to signal that in RTP or SIP, for fax you want T.38, and for modems you need incense and strange rites at midnight. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
Steve Edwards asterisk@sedwards.com writes: atftpd can do PCRE substitutions to transform a requested file name into something else. I've not used this facility, but I'm guessing you could transform: SIPDefault.cnf - cisco/SIPDefault.cnf sip.cfg - polycom/sip.cfg spa841.cfg - sipura/spa841.cfg Cute, but all that accomplishes is renaming. I want to run a script which returns a different configuration based on the file name (and possibly the client IP address). Unfortunately there is also no UserAgent-header in TFTP... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
Olivier oza-4...@myamail.com writes: Most (if not all) IP phones support provisioning through DHCP/TFTP. The trouble is some phones seem to require to store their config files in TFTP root directory. A lot of IP phones support HTTP instead of TFTP. This helps, because it is fairly easy to write a script which dynamically generates the configuration. Someone really ought to write a TFTP daemon with the same feature... Or a TFTP plugin for apache perhaps. /Benny ___ -- 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] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes:
OK, I decided to write it up in AEL. It's incomplete and untested, but
it probably gets the idea across a little better.
context agentcalls {
_2XX = {
Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
if (${EPOCH}${DB(AgentPaused/${AGENT})}) {
// Let the call through to the cell phone
Dial(...);
if (cell call was rejected) {
// Flag agent as paused for the next 30 seconds.
Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
};
}
else {
// Agent still paused.
};
};
};
I was going in the same direction at the end of my first mail, but I
hadn't written any code. There is a problem though: The Queue
application will keep sending calls to the Local channel, which have to
be rejected, over and over.
Would it perhaps work to simply Wait(30) if the call is rejected by the
phone? If the Queue assumes that the phone is busy for those 30 seconds,
I have accomplished my goal. It's worth a shot.
/Benny
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Re: [asterisk-users] Queues with unavailable members
Benny Amorsen benny+use...@amorsen.dk writes: Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. This works! Actually I tried out Wait(1000), but that worked fine. After 30 seconds (the timeout in the queue) the Local channel was closed, and a short while later a new call attempt was made. Just as I was hoping. It would still be neat to have a min_dial_interval option, so that Queue never overwhelms the server with failing dial attempts. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes: It would only be trying one agent at a time for each waiting queue member... Would it? Almost all our queues are on a ringall strategy. I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Not that expensive, but still a bit of a waste when done every couple of seconds. Especially if multiple agents are unavailable. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. This bit is solved by the ringall strategy. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. I'm fairly happy with the Wait(1000) solution for now. We'll see if testing shows any problems with it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html That looks very interesting, thank you! First of all though I need to avoid having them autopause just because they don't answer their phone. It should only happen if the call to their phone fails completely. I guess that could be done by not doing autopause but instead pausing manually in the context that the Local call passes through. That would also solve my second problem, which is that I need to pause it in all queues, not just one queue. The last challenge is to somehow unpause them after a while. In traditional programming that would be something like keeping a list of timeout,queuemember ordered by timeout, and then when every call comes in unpause and remove the ones where timeout expired... I'm not sure that I can make an ordered list in the dialplan though. I may have to resort to AGI, but I still need somewhere to actually store the list. Tricky. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Lenz Emilitri lenz.lo...@gmail.com writes: You could configure them as agents and have them log off automatically after a while they're not responding. Agents have to log in and wait for calls though, don't they? There used to be AgentCallbackLogin, but that has been replaced by dialplan code and chan_local. Otherwise a nice idea though. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: That shouldn't be too hard to accomplish. If you've got the addons (and mysql) installed you could store them in a MySQL table (timestamp, device) and have a cron job set to run at X frequency that un-pauses the queue members via AMI. Don't want to go to MySQL? Use system() to 'touch' files named after devices. Then have your cron script go through the files by creation date. Either way gets you there. This seems like a very heavyweight solution. Having a cron job running every minute isn't particularly attractive, and making a daemon do the job isn't my cup of tea either. Perhaps the problem could be restated in a different way: After a queue member rejects a call (instead of just not answering), the queue should wait X amount of time before sending the next call. Queues.conf has a million settings, but I can't find one which does this. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
Gordon Henderson gordon+aster...@drogon.net writes: I use Draytek Vigor 2820's these days. Mostly (when not having something more corporate or dealing with geeks who want a Linux based one) Built in hardware assist VPN too. They do have a SIP ALG, but it's turned off by default (the earlier ones had it turned on) Port forwarding works as you'd expect it to, and the traffic shaping is better than no traffic shaping. Well hopefully Draytek's are now better, but when I tried to use them 2 years ago the software was complete crap. VPN was unreliable (and supported only one tunnel per gateway), DHCP relay broken, DHCP server broken and lacking features, dynamic DNS implementation got them blacklisted by OpenDNS... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues with unavailable members
We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play audio file within an active call
Vieri rentor...@yahoo.com writes: Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped automatically (softhangup). What's the simplest approach to playing a sound file within an active channel? I think you should be able to do this with ChanSpy and the whisper option. However, Asterisk already has a facility for this. This is from core show application Dial L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Danny Nicholas da...@debsinc.com writes: Since Digium's contribution to Asterisk (hardware-wise) is Analog DAHDI cards, this makes sense (to me). They make quite a few digital DAHDI cards too (PRI and BRI). Analog is a bit 80's. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Jared Smith jsm...@digium.com writes: In a nutshell, you can pass the test without having any experience on Polycom IP phones and Digium cards, as long as you know how to use Asterisk itself. You certainly can, but I think it's worth it to invest ~30 minutes beforehand so you know where you put IP addresses and accounts in Polycom phones, and so you can get basic DAHDI working. It isn't hard, it takes about 30 minutes to learn, and it doesn't even really require that you have the hardware in front of you. I don't think it's unreasonable at all that it is in the test -- if you can't connect SOME kind of phone to Asterisk, you don't deserve certification. They have to pick one brand because it's infeasible to bring 5 different phones for each test taker. So, to all you people who complain that the dCAP is too hardware specific: It isn't. Really, the only tricky thing to know is that you probably want the Address and the Auth User ID fields on a Polycom phone to contain the same value (often the phone extension, if you don't want to be fancy). The Address field should NOT contain an IP address. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
Örn Arnarson o...@arnarson.net writes: I'm seeing the same behavior in 1.6.1.6. Any info on this? It would be helpful if you copied the exact error message involving the username field. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Jared Smith jsm...@digium.com writes: Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. I believe I can reveal this much without causing any problems for Digium: Be sure you have tried to configure a Polycom phone and an analog DAHDI card. Wasting 30 minutes on those two things makes passing the exam slightly more challenging... /Benny (whose only experience with analog DAHDI so far has been that dCAP exam) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
