Re: [asterisk-users] Dual WAN with load balancing
> I am not sure about the problem but note that it may be related to incorrect > IP being used. Sometimes, WAN 1 and sometimes WAN 2 Most likely. Get a provider that uses IP authentication rather than registrations, and enable access from both of your WAN IPs. All set. Luki -- _ -- 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 create a coredump for Asterisk
> Unfortunately, if I kill all asterisk-processes with "kill -9 ...", a > coredump never is writen to "/tmp", I also looked in other dirs. Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me. Luki -- _ -- 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] Channels In Use
> Are there any CLI commands to free this up or any other ways without having > to restart asterisk. Did you try soft hangup ? Or set an RTP timeout to avoid abandoned channels? Luki -- _ -- 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] OT: NAT in SPA922
>> However, when I connect a PC to that port, SPA922 works as bridge. Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike the SPA2102, etc). I think the 5.1 series is the latest firmware for the 922; the the 942, there is 6.1.5a. Luki -- _ -- 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] AGI <==> DeadAGI
> We run tens of thousands of call every day too. Call is controlled > by AGI , and the asterisk version is 1.2.24. I find memory leak in > asterisk. After serveral weeks, the memory used by asterisk will reach > 1.2 GB or higher. Each time I have to restart to asterisk, and the > memory leak will repeat. > > I wonder if you have the memory leak problem? Which version asterisk you > use? Thanks for reply. No, no memory leak here. Memory usage: 58 MB after: System uptime: 25 weeks, 5 days, 12 hours, 39 minutes, 52 seconds. Asterisk version 1.4.23.1 (with about 25 custom, in-house patches). This particular box only handles signaling from a dozen static peers. No registration, no media (directrtpsetup=yes), no NAT, no transcoding, no MOH... but it does use realtime for SIP and IAX, and AGI and DeadAGI for routing. Luki -- _ -- 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] AGI <==> DeadAGI
> It is irrelevant who hangs up, you want to just use DeadAGI in the h > extension I wish that would be the case, but at least on 1.4 I see: [Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new stack [Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI The good news is, we run tens of thousands of calls every day through this box and about half of them spit out this warning, but it never caused any problems for over a year. Thus this warning is probably safe to ignore. Luki -- _ -- 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 can I record the conversations in a conference call?
> "Please note: A Zaptel timer must be present for conferencing to work!", but > if the user does not have "ZAP/DAHDI hardware", he can use ZAP/DAHDI DUMMY Actually, my understanding is that this is incorrect. The conference must contain ZAP/DAHDI callers. A dummy won't do. The reason is that the ZAP/DAHDI driver mixes the audio in the driver and when this is not available it falls back to mixing within MeetMe. But in such case, you can neither record the conference nor run an AGI in the background. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe. This is from 2004, maybe it changed by now. Luki -- _ -- 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 can I record the conversations in a conference call?
> I use the option 'r' on 1.4, to record the meetme application. > Asterisk leaves these records at /var/lib/asterisk/sounds/meetmeXX. That option only works for conferences using ZAP/DAHDI hardware. You can, however, start to Monitor() the channel prior to entering the conference, but you should only do it for the first caller. MeetMeCount() will help. The caveat is that if the first caller disconnects, the remainder of the conference will not be recorded. If anyone has a better solution, please tell us :). Luki -- _ -- 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] Ring Two Extensions Simultaneously with different caller ID values?
Lincoln,
> Is there any way to feed different caller ID information to both sets while
> keeping them ringing simultaneously? My idea is to prefix the called
> extension to the name field (so as not to break redial/callback features on
> the phones)
you can do this with a local channel, like:
Set(__TARGET=${EXTEN})
Dial(SIP/phone1&Local/pho...@common_area)
[common_area]
exten => _phone.,1,Set(CALLERID(name)=${TARGET}: ${CALLERID(name)})
exten => _phone.,n,Dial(SIP/${EXTEN})
Something like that. I hope I got all the () and {} right, I don't do
that much dial-plan coding anymore...
Luki
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Re: [asterisk-users] Cache sound files for faster processing
> Are there any way of configuring of Asterisk so it'll cache sound files in > memory, > and when Asterisk receive a call, instead of loading sound files from the disk Not directly, but it's not really needed. A long as the machine has enough RAM, the files will be served from RAM by the operating system. Sure there is the overhead of opening/closing files and reading them, but on modern OS this overhead is negligible if the files are cached (asterisk may even use mmap, but I'm not sure). You can also make a ram disk (say via tmpfs), copy the sounds there and symlink the sound directory to that location. However, I don't think you will gain much. Luki -- _ -- 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] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk
> When this bug occurs, it freezes I/O to all devices and the only way to > recover is to reboot the system. Are you running asterisk with realtime priority (-p)? I once managed to take town a box with a dial plan loop; asterisk was taking to 100% CPU and because it had highest priority, nothing else would run. Kernel would respond to pings, but that's it. We no longer use realtime priority for that reson :). Luki ___ -- 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] Cisco 7961 - can't place calls
> Thanks for the reply. I am not getting any output from the Asterisk CLI when > I place the call. The phone give busy signal as soon as I push the first > digit of the extension #. Sounds like the same problems I am having with the 7971G (see my message on this list couple days ago). In my case it's an authentication mismatch between the matched peer and the peer name in the SIP message. Try turning sip debug on and see if the packets give you some hints. Incoming calls also always work for me. Luki ___ -- 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] Setting up Nokia e71: registration problem
> In SIP setting on the e71 I set the public user name as > 1...@10.10.11.180. There is a sip.conf context [1995] I can confirm that the Nokia E71 works perfectly fine with Asterisk. It looks like you have a space between sip: and your username in your SIP Profile on the phone. If in doubt, remove the profile and recreate it. Luki ___ -- 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] Cisco 7971 behind NAT
Darryl, OK, that could work but it makes the use of these phones behind consumer routers rather impossible. How many of those will inspect and transform SIP packets? Oh why does Cisco have to do things differently from everyone else... Luki 2009/11/16 Darryl Dunkin : > You need to enable SIP transformations on the firewall, the packets will > have to be dynamically re-written to handle multiple Cisco phones of > these models. Be sure 'nat=no' is set in sip.conf for the phones as > well, or Asterisk will reply to the incorrect ports (source instead of > the mangled contact header). > > In this case, you'll need to compile in the SIP connection tracking/NAT > bits in the kernel, they should be able to mangle the packets > appropriately. I have never tested this, as all my deployments have > hardware firewalls with SIP support built-in. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7971 behind NAT
Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at that location, unless I'm mistaken). Incoming calls work fine on both accounts, but outgoing calls work only from the most recently registered account (the order is random due to timing) since both appear to asterisk as IP:5060. An outgoing call from the other account is rejected with an authentication mismatch, which makes sense. Asterisk matches the most recently registered peer by IP/port and if the user name differs, it complains, even if the password is the same for both accounts. So, is this the worst SIP implementation ever in those Cisco 7971's or am I doing something very wrong here? Technically even without NAT this confusion would occur as both accounts use IP:5060 so Asterisk cannot tell them apart during the initial peer matching stage. Of course the source port the Cisco selects is different with every dialog, so that doesn't help either. Any input would be appreciated before I throw that phone out of the window. Thanks, Luki ___ -- 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] E65 fails registration, soft phone works
Martin, sounds like the hiccup my E71 had once. I think the symptoms were identical. Changing the transport type from Auto to UDP solved the problem for me. The Auto setting worked, but only sometimes. Maybe the E65 is similar... Luki 2009/9/12 martin f krafft : > Hey folks, > > I am trying to get an E65 to connect to asterisk, and I would really > appreciate a second set of eyes. The SIP dialog completes fine, but > the phone subsequently says "Registration failed". > > I am in a network that has what seems to be a SIP-capable NAT > gateway, but the asterisk is configured nat=yes anyway. Using > a softphone (twinkle), I can connect just fine, SIP and RTP work. > > But when the E65 tries to connect, it seems to complete the SIP > REGISTER dialog, but then it'll say "Registration failed": ___ -- 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] Seemingly easy question: NPA/NXX
> When number starts with 011, and as country code and city code is > identified, expect as many numbers as determined by country+city code > (once you know country and city code, you know how many local digits to > expect) ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. Luki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP carrier billing technicalities
> Do most carriers the carrier just use CallerID as an origination > number? As far as I am aware, the concept of a BTN is gone with > SIP I don't know about most carriers, but the couple "bigger" providers we're using use CallerID as the BTN for outgoing calls. They bill us by destination LATA/OCN and determine if it's an intrastate or interstate (inter/intra-LATA) call based on CallerID. Therefore calls from the same machine (no user/pass since they authenticate by IP) are billed differently simply due to CallerID. So yes, technically it's possible to be charged the lower interstate rate for an intrastate call if the CallerID is set out of state, but IMO that doesn't make a good impression and isn't worth the savings. YMMV. /Luki ___ -- 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] Multiple SIP phones behind a Linksys firewall
> I always assumed that you can have multiple SIP phones behind a Linksys > firewall/router (WRT54G) all using the same STUN server/port. I got 10-20 SPA942's behind a OpenWRT router (on WRT54G, WRTSL54GS, ...) at several sites, no STUN, no special configuration, no problems at all. Just as a precaution, I set the SIP port and RTP port range for each phone differently so that it's unique (i.e. Phone 1 SIP port 6001 and RTP 10100-10199, etc.) but that's really just a precaution to help the the Linux' conntrack on the OpenWRT a bit. It's not really needed as the router will resolve port conflicts by rewriting the ports transparently. Bottom line, a few phones behind a well-behaved NAT should work just fine. /Luki ___ -- 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] Reputable company for SIP/IAX2 trunking
Steve, if you want quality and reliability, then you need to get as close as you can to the actual big guys operating the equipment, such as Level3, GlobalCrossing, XO, CommPartners. But they won't be interested in doing business with you for just 1 DID and couple thousand minutes a month. So find yourself a good first-hand reseller of those big guys who is interested in doing business with you. There are many out there. We have been getting >90% of our west-coast DIDs from CommPartners directly, and over the last 3 years, I don't recall a single indecent when they let us down service. The actual VoIP service is excellent; billing and paperwork can be messy at times. Luki On Dec 15, 2007 4:25 PM, Steve Finkelstein <[EMAIL PROTECTED]> wrote: > Hi all, > > There's a myriad of options these days and I haven't been keeping up to date > with what's respectable any longer. > > I essentially need a provider that will provide me with one DID to start and > let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on > my end and have full control over the dial plan. This way I can branch out > my DID into extensions and have it dial individual peers according to an > extension. > > Looking for some feedback on what provider is quality these days. I don't > mind paying an extra dollar or two. > > Thanks, > > - sf ___ --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] RTP traffic not being forwarded
> When using 'rtp debug' on the asterisk console, it shows that it is > receiving traffic from one endpoint, but not the other. A wireshark trace > reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's "on the wire", i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
> We tend to find that on the rare occasions asterisk does decide to die, > it very often doesn't die completely. Agreed. Need to also watch out for SIP deadlocks (asterisk is up, you can connect to the CLI, but it does not respond to any SIP traffic, or sip reload, or unload chan_sip, or restart). Sipsak via an external monitoring script works for us. On a timeout it first tries to gracefully stop asterisk, then it force kills it with kill -9, and the restarts it. That has proven successful in minimizing downtime. On the same machine, same binary, "same" traffic volume, sometime asterisk stays up for months (current record, 51 weeks!) and sometimes it will decide to die after couple days (current record, 6 times on the same day). A core dump doesn't reveal anything suspicious. It's not load related as crashed can happen at 3am when it's pretty quiet. I suspect it's due to network issues (dropped packets, etc) because sometimes just before the crash the console is full with __sip_destroy: Trying to destroy ... not found in dialog list?!?! messages. No, I have not upgraded to 1.4 yet. Still, I find it ironic that all this effort goes into fixing the symptom rather than the cause. But I'm not complaining... just don't have a better idea how to fix it. Luki ___ --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] SER/OpenSER as registrar to Asterisk (1500 SIP users)
> Some caveats (which may be different for OpenSER, so someone else can > chime in): What about: 1) Message waiting notifications? Especially in a distributed system with multiple Asterisk servers? 2) Different codecs for different SIP users/accounts? DTMF modes? I know SER doesn't deal with the media at all, but if you let SER handle registrations and authentication, then I'd rather not keep track of codecs/DTMF on asterisk as well. Those two have been bugging me most. Luki ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
> Luki, thanks for writing to say it DOES work. I've have just now had > another look, found my mistakes (basically $MAC instead of $MA), and > it's working! I'm glad you got it sorted out. Yes, it works with XML or compiled files. To help with troubleshooting, specify a syslog server and set the debug level to 3 in the initial spaXXX.cfg, and the device will tell you what it tried, what worked and what failed (i.e. XML parse error, invalid parameter, URLs, etc.). That's just a note should someone get hang up on that in the future. --Luki ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
> I'd like to be able to templatize a server, add a bunch of new handsets > into sip.conf and extensions.conf, and then plug the phones into a > network and have some DHCP and/or TFTP "glue" logic that sees the DHCP > or TFTP request, and from it generates a boot file (an .XML file) and a > response parameter list for DHCP... populates a file into the /tftpboot/ > directory, etc. Here's how you do it. 1) In the DHCP server's config (dhcpd.conf) you specify the IP of the TFTP server: option tftp-server-name "66.55.44.33"; This can be a remote server, as long as it's accessible by the device. 2) The factory settings on the Sipura devices (ATAs and phones) have /spa$PSN.cfg in the Provisioning profile rule, so the device will connect the TFTP server you specify and will try to retrieve that file, i.e. ftfp://66.55.44.33/spa942.cfg for the SPA-942 in this example. 3) This file contains very minimal information, which tells the device where to download its final configuration from. This can be a remote http server so you can maintain the configs on one central server. Example: http://YOUR.HTTP.PROVISIONING.SERVER.HOST/$MA.bin 4) The device will then connect via HTTP and will try to retrieve for configuration for its MAC address. Since it's a HTTP request, you can generate the provisioning data on the fly (even from the a database), either in XML format or in compiled format if you have the Sipura compiler. The above works just fine and very reliably. We have disabled periodic resync as the Sipura phones seem to reboot sometimes for no good reason when they apply the "new" but unchanged profile. If there is a config change, we just push it on the phone with SIP NOTIFY option. --Luki ___ --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] Paging possible on an ATA?
> Is it possible to configure a PAP2 to > auto-answer for either paging or intercom? No. You cannot force the connected device (phone) to auto-answer. Imagine you have a plain old phone attached to it, who's going to lift the receiver? ___ --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] indications.c: Can't generate that much data!
> I don't see what difference removing the "r" option has made from an > Asterisk perspective - in both cases Asterisk tries to emulate a > ringtone but fails for some reason when "r" is present. According to the > the "show application dial" help having no "r" present for Dial should > NOT generate a ringing tone yet here it does. Simple. When using "r", asterisk needs to generate the ringing tones. For some reason your indicactions.conf describe a tone which is longer in duration than what can be generated by asterisk, so the error is shown and no tone is generated. Probably the max buffer length is somewhere preset in the code. If you do NOT use the "r" flag, asterisk simply passes call progress indications from the source, without the need to generate any. Hence no error, and you hear ringing. Yes, it's a bug, but there no magic in the symptoms you observe. Luki ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
> Any good SIP providers out there? It really depends where you are. We're serving pretty much only Los Angeles and Seattle rather than the entire US, and thus by focusing our efforts on those limited markets we can achieve pretty good quality and reliability. Servers are <15 ms away, less potential for congestion, etc. Of course with the Internet being a best-effort network there are no guarantees, but by minimizing the potential for trouble you can achieve decent quality nevertheless. So, try to find a provider "near you" focusing on your market. Luki ___ --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] Hardware that can ring my phone?
> You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) > connection instead of a Digium card. The price is around $90-100. > > Almost any old PC will do if it can run Linux. There are also other > alternatives to a PC such as the Linksys WRT54GL. The OpenWRT (on whatever supported "router" hardware) + SPA-3102 is a pretty decent combo. You can reinvite the traffic between the FXO and FXS (g711 only) and get good quality without even taxing the router. FYI, a WRT54G had no problem running asterisk 1.2.x with 4 concurrent channels (g711, no transcoding, just RTP proxying). I'd look into something like that. And you can "expand" it fairly easily by adding another SPA for a second line. Luki ___ --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] SPA-2100 Distinctive Ring
> I did find out how to add the sip message for distinctive ring > i just dont know what variable needs to be passed in > order for it to work. Try: SetVar(_ALERT_INFO=Bellcore-r2); etc. ___ --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] kore dump
> just an idea, but maybe qmail, samba, and bind have a smaller memory > footprint than an in-use asterisk? No, probably not. Asterisk's is about 20-40 MB depending on the number of extensions, etc. Smbd's is similar, bind's is actually 90 MB (with about 600 zones). > can you take the hardware offline long enough for a memtest? The machine has been retired (routine upgrade cycle). But I hardly doubt that was the problem. My guess is it was somehow related to limited CPU power (thread switching, interrupts, or whatnot). The old hardware was single CPU and a lot slower. --Luki ___ --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] kore dump
> I am having a problem with Asterisk frequently crashing on me as > well. I just run it under supervise: But that's just a band-aid. If it crashes, it takes all calls with it. Hardly a good thing, unless you only have 1 call at a time -- then it's probably no the end of the world. I still don't know what's up with the crashes but here are two observations I made: 1) I moved the same installation from one hardware to another. On the former hardware it would crash every 2 weeks, on average. On the new hardware, it has not yet crashed and it has 9 weeks of uptime. Same call volume, same devices, same network. I'm running asterisk chroot'ed so all libraries, binaries, config files, etc. are identical. Only the hardware and kernel are different. 2) The same old hardware has been in service for 3 years and no other programs crash on it. Ever. It's no unusual seeing uptime for say qmail, samba or bind of 200+ days. I have therefore reasons to believe that the hardware is OK. So go figure. And BTW, the crashes (based on the core dumps) are always at a different place. There is no consistency. Right now I'm just glad it no longer core dump on me :). --Luki ___ --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] got-name
>> I don't know how to contact them, but I am having the same problem. > The domain is registered to Jed Stafford. If you want the domain contact > details you can do a whois. The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to increase call count
> So far, I've only been able to make about 15 concurrent calls before the cound > quality gets poor, and I really need to increase this. 15 calls isn't very many at all. > I've got QoS configured to prioritize IAX2 traffic above all and my connection > to the Internet is a PtoP 100Mb ethernet link. (255.255.255.252 subnet mask) > > The server is an AMD Athlon(tm) 64 Processor 3400+ with 512Mb of RAM. The Nic > isn't sharing an IRQ with anything else and the CPU never exceeds 15% > utilization. You shouldn't see call problems with 15 calls with this box. You say yourself that the CPU isn't taxed and 512 MB RAM should be plenty for 15 calls. Perhaps there is problems with the Internet connection. Can you check at the receiving end if the packets are arriving on time with no loss? You could also check if they are leaving your box as scheduled, but I'd imagine that's not the problem. How do pings and traceroute look like with 15 calls up. Could you try SIP instead of IAX? Sounds like the problem might be upsteam from you. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blocking 900 calls
Presently I have _all_ 900 calls blocked in Asterisk 1.25 but today I had to call a parts vendor at a 972 number. Blocking anything with 9XX isn't a good idea. There are lots of regular area codes in the 9XX block -- take a look: http://www.localcallingguide.com/lca_listnpa.php?section=9 I *think* only the exact 1-900 prefix is a premium rate call. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Codec
I'm wary of using g711 of public broadband networks. ... It'd be interesting to see some comparisons or comments from people using g726 as this does seem to be supported by quite a few hardware devices. We are using g711 pretty much exclusively for all residential customers in the US and it worked out well for us. For those with very slow DSL connections in rural areas (128 kbps up / 256 kbps down) we use 40 ms packets as 20 ms packets still used two ATM frames and hence the overhead was rather large. If that fails, we found g726-32 to be a good alternative. Voice quality is almost as good as g711 (a bit duller), music is acceptable. Transcoding overhead is low, many ATAs support it, and the bandwidth (with overhead) is about 40 kbit/sec. It's a good alternative, IMO. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
Average CPU utilization per call: 0.137% (~1735 MHz) Perhaps a naive question, but how does 0.137% CPU utilization per call equal 1735 MHz per call? If 1735 MHz / 0.137% = 1735 MHz / 0.00137 => 1266423 MHz at 100% utilization ??! Even with 4 CPUs, those would be 316 GHz CPUs. I think you meant: Average CPU utilization per call: 0.137% (~17 MHz) --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemails after X days
You should expect this to massively break voice mailboxes. Well, it won't massively break them, just a bit. We do this on some mailboxes and it works OK. The problem is that is you delete message 1 and leave 2, a new message will become 1, thus breaking the sequence. They will be played back as 1 (newer) followed by 2 (older) message. Then again, I'm not sure what happens if there is a break in sequence -- I think I patched my code to deal with that. It's ugly and inefficient. Still all of these solutions are a band aid at best. I don't like do it this way. I wish Asterisk could do it itself. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
How did you set it up ? With OpenPBX aka Callweaver. Is part of source, no patching needed. Which app_rxfax version did you try ? http://callweaver.org/browse/callweaver/trunk/apps/app_rxfax.c Does it offer T.38 termination (forwarding TDM faxes to T.38 gateways) ? Yes, via T38Gateway(). But it's rather beta. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
So you'd rather have the entire PBX crash in order to avoid creating sufficient iaxmodem instances to handle your fax call load? No, but so far this occurred only once in about a year of service. Not ideal, but "acceptable" considering Asterisk itself segfaults or deadlocks every now for no apparent reason. I had more trouble when trying to use T.38 with the newest app_rxfax so I abandoned it for now. And iaxmodem cannot do T.38 anyway... So you are saying a pool if iaxmodems and a loop through Dial() to find an open one is the way to go? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Forget them! Use Hylafax and iaxmodem instead. I wondering, how do you guys handle multiple calls? We frequently get many concurrent faxes, sometimes even to the same number. As far as I know, one instance of iaxmodem can only support one fax session at a time. So essentially you need a pool of iaxmodems running on different ports, and then Dial() them until you find one that accepts your call. Or did I get that wrong? That seems really like a drawback to me, that's why we're sticking to app_rxfax, which in the newer versions also supports error correction. With app_rxfax you are always "guaranteed" that that there is someone to answer the fax, given sufficient resources (CPU and memory). The biggest drawback with app_rxfax is that if it crashes for whatever reason (happens sometimes), it will take down the entire PBX and all sessions with it. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowing call every 15mins
GotoIfTime can help you here, but it'll be a little messy:
That should be a sign that GotoIfTime is not the right tool to use here.
Instead try:
exten => 1,1,Set(M=${TIMESTAMP:11:2})
exten => 1,n,GotoIf("${M}" = "00" | "${M}" = "15" | "${M}" = "30" |
"${M}" = "45"?good_timing)
exten => 1,n,VoiceMail([EMAIL PROTECTED])
exten => 1,n,Hangup
exten => 1,n(good_timing),Dial(SIP/techsupport)
You get the idea... but I agree, why on earth would you want to do
that? We only provide 6.7% tech support?!
--Luki
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Re: [asterisk-users] Delay in Dial()
Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. Not directly, but yes. Hint: Local channel + Wait. Something like this: Dial(SIP/phone&Local/[EMAIL PROTECTED]) [delayed] exten => XX,1,Wait(10) exten => XX,2,Dial(SIP/[EMAIL PROTECTED]) --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP peer disappearing
Hi all, I'm having this weird issue that I can't explain. Maybe someone can explain what is happening. This is a Asterisk install that has been in production for 6+ months. It's version 1.2.10. Couple weeks ago one SIP peer started disappearing randomly. And I mean it simply disappears. One second "sip show peers" shows it, and then it's gone. A simple "sip reload" fixes it: (all good...) [Mar 21 19:53:53] NOTICE[4481]: chan_sip.c:12049 handle_request_register: Registration from '' failed for 'IP' - Username/auth name mismatch lax*CLI> sip show peer luki1 Peer luki1 not found. lax*CLI> sip reload [Mar 21 19:54:10] Reloading SIP [Mar 21 19:54:10] == Parsing '/etc/asterisk/sip.conf': [Mar 21 19:54:10] Found [Mar 21 19:54:10] == Parsing '/etc/asterisk/shared/users.conf': [Mar 21 19:54:10] Found [Mar 21 19:54:10] == Parsing '/etc/asterisk/sip_notify.conf': [Mar 21 19:54:10] Found [Mar 21 19:54:22] -- Registered SIP 'luki1' at IP port 5060 expires 60 [Mar 21 19:54:22] -- Saved useragent "Linksys/SPA2102-3.3.5(a)" for peer luki1 It only affects one peer. Sometimes it disappears after a few hours, sometimes after a week. The box can be mostly idle or loaded. No difference. I did restart Asterisk completely, no help. Upgrading is an option, but so far there was not need. Any ideas what is happening?! This peer has been fine for months, and now this. Line 2 on the Sipura registers with another box (it's actually Asterisk 1.2.5) -- no problems there. And yes, I did reboot the Sipura. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys not Ringing
shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Something strange is going here, and my bet is on some kind of NAT screw-up. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packetization Rate
Obviously somewhere in the asterisk code 30ms must be coded... is it set in just one place, and if so can I set that to 20ms? The default is 20 ms for most (all?) codecs. It's in rtp.c, where ast_rtp_write() creates a new smoother. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number of SIP messages per minute
Just how many SIP packets do you think it takes to set up a call? Probably around 8 - 10 per call, excluding any ReINVITES, DTMF, etc. INVITE, Authentication Required, ACK INVITE w/AUTH INFO, TRYING, RINGING, OK BYE, OK --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with a Linksys SPA 2102 and asterisk
Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? They work fine with Asterisk; most likely it's your wireless link that's the cause of your problem. The jitter buffer will only affect received audio, i.e. on your side, and since that is fine, you probably don't need to adjust it. Instead try this: 1) Change packet size in increments of 20 ms (i.e. 0.02, 0.04 or perhaps 0.06). Your wireless link may not like too many small packets. 2) Turn off silence suppression if it's on. 3) Try a different codec -- g726-32 or even ulaw to see if it makes a difference. See if that helps. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR reports short call length
Before I go into higher detail, does anyone have any ideas about this? Yes, see the transfer option for IAX. Set it to transfer=mediaonly which will leave the signaling unchanged and the channel alive, and thus produce correct CDRs. See: http://www.asterisk.org/doxygen/1.4/Config_iax.html PS: Never tried it... --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinct call permissions for each user
someone please give me one example?
[locals]
exten => _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten => _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten => s,1,Dial(SIP/[EMAIL PROTECTED])
exten => s,2,Dial(Zap/.../${ARG1})
[fullaccess]
include => locals
include => longdistance
include => ...
[restricted]
include => locals
include => ...
Put user A into the restricted context, and user B into the fullaccess
context. You can include other extension (i.e. services) and implement
roll-over onto a backup trunks in macro-outcall.
You can of course also simply it and only have two contexts and no macro, etc.
--Luki
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Re: [asterisk-users] Long call setup times on SIP to zaptel calls
Jordan said "the SIP device sends the request almost instantly" so it's not the SIP phone's fault. The channel bank probably takes 1-2 seconds to pick up and wait/check for dial tone, 1-2 second dialing, and the telco takes 1-2 second to ring. So the complete PDD is ~5 seconds. You could try putting a Ringing(); before the dial statement to let the SIP phone know the call is being connected. I believe once progress comes from the Dial command, it will replace the Ringing. However, if your channel bank answers the call right away, this won't help. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Number of Calls Asterisk Can Handle
in theory, a gigabit interface can move 1048576Kbit/sec - now if i generously allocate 96Kbit/sec for every G.711 call, the network transport can handle, again in theory, 10922 simultaneous calls. would it be wrong to expect performance near this mark for the asterisk software? 10922 on any currently available PC architecture? Nope. It's closer to 160 kpbs per call (two legs, 80 kbps each) in either direction. With 20 ms packet size, for 10922 calls you'd be looking at 2184400 packets/sec processed by Asterisk... I don't think so. Plus with 10922 calls and an average of 2 mins/call, you're looking at about 90 call setups/tear downs a second. I don't think even without running the RTP through Asterisk this box could handle 10922 concurrent calls. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite problems
Stefan, When I have 2 SIP endpoints that both aren't configured with "canreinvite=no" then I get no sound. The Sipura 3102 definitely works fine with canreinvite=yes and I never really had a problem with any of the Sipura devices in this respect, especially when there is no NAT involved. However, the default "Auto NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255, so your 192.168.254.0/24 network would be considered a LAN address by the 3102 and hence the traffic would go out the LAN interface (not WAN). Change this setting by removing this range. It's on the Admin > Advanced > LAN Setup tab. If that doesn't help, then you need to check what traffic is being sent. Since all devices are on the same internal network I assume they can see each other. You need to look at the Invite (and ReInvite) messages sent and received and see if the IP addresses for RTP listed there make sense. Then I suggest you use tcpdump to see what traffic is sent by each device, and where. If you have a switched network environment this will be a bit tricky as your * box won't see this traffic, so you may want to use a hub for this test (just temporarily) or if available set up port mirroring to sniff the traffic. Good luck and keep us posted. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
No such host: 3213)
Look for an extra closing parenthesis in your Dial command:
Dial("SIP/3210-084eaa80", "SIP/3213)|30|to")
It should be SIP/3213 rather than SIP/3213).
--Luki
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Re: [asterisk-users] Logging to /dev/ttyS0
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
I think you might be better off with a System() call in your dial plan such as:
System(echo ${CALLERIDNUM} > /dev/ttyS0)
That will send the callerID number followed by a new line. You can of
course change the format to your desire. Make sure /dev/ttyS0 is
writable by the asterisk user, and is also properly set up (baud rate,
bits, ...).
--Luki
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Re: [asterisk-users] Dimensioning a 50 sip phone installation
I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. Should be plenty if not an overkill. One of our setups: 20 phones, 8 outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel. Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per call. Quite reliable (hence not upgraded). This is a g711 only setup with no transcoding. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P I do. Exclusively. I personally don't like the g729 compression (audio quality and license issues) any my customers definitely notice the difference right away and wonder why the quality "degraded". I guess I spoiled them with ulaw. So no g729 here. g726-32 on the other hand was acceptable, although the difference is still noticeable. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
So, my "peak" would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Sounds about right. Or the other way around (if you need to know the peak bandwidth usage): For audio: 1,000,000 minutes/month = 33,000 minutes/day 10% daily usage in 1 hour = 3,300 minutes used 3,300 minutes used in 60 minutes = 55 concurrent calls 80 kbps / 1 call direction * 55 calls = 4.4 Mbps per direction Assuming full duplex audio, you need 4.4 Mbps in + 4.4 Mbps out per call leg. If you route the call so each packet comes in and goes out the network (2 call legs), then double the bandwidth. I guess adding 0.1 Mbps for call setup and tear down is safe. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI History
thats prety smart... think hard.. wot was the command u gave to exit the CLI?? OK, come on everyone. This is getting ridiculous. That's the entire point that "stop now" was NOT the last command on the CLI, yet it shows up at the most recent upon recall with the Up key. I have the same, except in my case it's stuck on "show channels" (which is rather convenient so I didn't complain). And yes, it doesn't matter if I exit the CLI with Ctrl+C or exit. In my case it's probably a permission issue since I run * non-root and chroot'ed. Either way, I don't see why the history could not be save upon exit with Ctrl+C -- the mySQL client does it. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN As SIP Tunneling?
If my IP Phone set QoS and the VoIP Termination provider's * PBX sets QoS. And we now connected via a VPN tunnel. We should be able to guarantee Quality due to the Tunnel. Nope. You only control the QOS within your tunnel (i.e. among other traffic flowing through the tunnel). But what QOS guarantee does your tunnel traffic have? None, if it goes through the public Internet. You don't gain anything QOS-wise by going through a tunnel, except hiding your traffic in case your ISP purposefully assigns lower priority to VoIP traffic and doesn't do it to OpenVPN/GRE/ traffic. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does voicemail authentication take place?
Luki, thanks for the response. Could you give me an
example of the use of vmauthenticate in a very short
dialplan?
Thanks
Jez
*CLI>
-= Info about application 'VMAuthenticate' =-
[Synopsis]
Authenticate with Voicemail passwords
[Description]
VMAuthenticate([EMAIL PROTECTED]|options]): This application behaves the
same way as the Authenticate application, but the passwords are taken from
voicemail.conf.
If the mailbox is specified, only that mailbox's password will be considered
valid. If the mailbox is not specified, the channel variable AUTH_MAILBOX will
be set with the authenticated mailbox.
... and ...
*CLI>
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
Authenticate(password[|options]): This application asks the caller to enter a
given password in order to continue dialplan execution. If the password begins
with the '/' character, it is interpreted as a file which contains a list of
valid passwords, listed 1 password per line in the file.
When using a database key, the value associated with the key can be anything.
Users have three attempts to authenticate before the channel is hung up. If the
passsword is invalid, the 'j' option is specified, and priority n+101 exists,
dialplan execution will continnue at this location.
... so something like that (never tried it):
exten => s,1,VMAuthenticate
exten => s,2,NoOp(Authenticated as {$AUTH_MAILBOX})
--Luki
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Re: [asterisk-users] When does voicemail authentication take place?
res |= ast_register_application(app4, vmauthenticate, synopsis_vmauthenticate, descrip_vmauthenticate); You need to look more closely at the code. This snippet registers the dial plan application VMAuthenticate so vmauthenticate is called wherever you use that function in your dial plan. static char *app4 = "VMAuthenticate"; static char *synopsis_vmauthenticate = "Authenticate with Voicemail passwords"; static char *descrip_vmauthenticate = " VMAuthenticate([EMAIL PROTECTED]|options]): This application behaves the\n" "same way as the Authenticate application, but the passwords are taken from\n" "voicemail.conf.\n" " If the mailbox is specified, only that mailbox's password will be considered\n" "valid. If the mailbox is not specified, the channel variable AUTH_MAILBOX will\n" "be set with the authenticated mailbox.\n\n" " Options:\n" "s - Skip playing the initial prompts.\n"; --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is 1.2.12.1 production ready
I suspect that for every problem you hear about on the list there are probably 100 other happy asterisk administrators. Not to downplay legitimate issues, but many times, instabilities can easily be attributed to the OS, hardware or a million other things not caused by asterisk. I agree. However, there seems to be some randomness involved as well. For example, one of my production machines now has an Asterisk uptime of almost 8 weeks on 1.2.10. Before that it had a week full of crashes, typically 2-3 times a day. Same version, same configuration. I was planning on fixing it, but then suddenly it started behaving without my intervention, reinstall or anything. Why, is beyond me. I'm keeping my fingers crossed. So far I've had excellent luck with 1.2.5 -- 8 months of uptime till reboot. No crashes ever. Still, an 8 week uptime isn't great IMO. The same machine has no issues running thttpd or qmail for 12+ months without a hiccup. So it's hard to blame it on the hardware because no other program seems to crash randomly on the same machine. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sequential Dial() commands
exten => context,1,Dial( SIP/[EMAIL PROTECTED]) exten => context,2,Dial(SIP/[EMAIL PROTECTED]) Currently, if the first number doesn't answer, the session is closed. Specify a time out. Without it * will not continue to priority 2 if [EMAIL PROTECTED] is reachable but does not answer. exten => context,1,Dial(SIP/[EMAIL PROTECTED],20) exten => context,2,Dial(SIP/[EMAIL PROTECTED],20) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax over ip
Christopher,
Also, how am i supposed to get my fax machine onto my ethernet network? i
assume it needs some kind of Aanalog Telephone Adapter, just like with VOIP.
You need a T.38 capable ATA. There are a few but not too many. I
believe the Grandstream ATAs have T.38 support or will have it
("Support transparent Fax pass-through and in the future T.38
(pending)"). The Linksys/Sipura 2100 can definitely do it -- it works
well for me :).
Then you need a T.38 capable ITSP. Some are listed on
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38 but I
don't have experience with any other them.
T.38 pass-through support in Asterisk is available as a patch on the
bug tracker for 1.2. Not sure if it made it into the 1.4 beta version
or not, but on 1.2 it works OK for me.
--Luki
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Re: [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?
I'm interested if anyone else in the Asterisk list can get through to +1-907-747-8633 via voip Sure, no problem. A nice friendly female voice tells you the time and temp, indeed. The thing is that the call never connects -- that info is sent via call progress, so a misconfigured server (i.e. one that uses the "r" option in dial() or equivalent) would just give you ringing and ringing... [Sep 21 17:49:45] -- Called [EMAIL PROTECTED] [Sep 21 17:49:45] -- SIP/trunks-094da090 is making progress passing it to SIP/1001-b7a030f8 [Sep 21 17:49:48] -- Ringing [Sep 21 17:49:48] -- Progress [Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345 etc. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax and T.38
Hi all -- Perhaps I haven't been looking in the right place, but is there a T.38 capable version of app_rxfax? I got T.38 working in passthru mode in Asterisk (thanks Steve!) with a Sipura ATA and the PSTN switch, and so far so good. I got app_rxfax working with the ulaw codec (which works most of the time) but having it receive faxes with T.38 would be ideal. Can this be done already? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3000 dialplan coding...
Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. I believe that would be just: [EMAIL PROTECTED];10 Where -16 is the volume (i.e. -16 db). The other parameter means to play the dial tone for 10 seconds, then go to a busy (see separate tone definition). That's at least my interpretation. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible to To Have Different Outgoing VM Messages, but One Mailbox?
I would like to have callers that call different DID numbers receive different outgoing messages (based on the number called), but have all of the incoming messages in one box. Any way to do this that comes to mind? Yes, symlink the INBOX and Old directories to the "master" mailbox. This "shares" the new and old messages folders, but not the custom folders (you could symlink those too). # ls -l mailbox2 lrwxrwxrwx 1 root root 14 Nov 5 2005 INBOX -> ../mailbox1/INBOX/ lrwxrwxrwx 1 root root 12 Nov 5 2005 Old -> ../mailbox1/Old/ drwxr-x--- 2 asterisk asterisk 4096 Aug 7 17:44 tmp -rw-r- 1 asterisk asterisk 264826 Apr 9 15:53 unavail.wav --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message waiting question...
Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. There is this old patch that does remote MWI over IAX (among other things). I used it on earlier versions and it worked quite nicely. This was before 1.2 so it may no longer work at all. At the very least it will likely required some updating. Doable, just depends how much time you want to put into it :). See: http://bugs.digium.com/view.php?id=4371 --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change current working directory to /tmp
Patrick, I run asterisk in a chroot'ed environment and within it I cd into /tmp just before starting asterisk. The kernel happily dumps the core files into that /tmp directory. As far as I can tell, this behavior has not changed recently and it definitely worked for 1.2.7.1. You can also force a directory where core files should be dumped with: mkdir /corefiles echo /corefiles/core > /proc/sys/kernel/core_pattern The kernel will then dump all core files for any process into the /corefiles directory. --Luki On 7/25/06, Patrick Cervicek <[EMAIL PROTECTED]> wrote: To get a core file, I started Asterisk with cd /tmp /usr/sbin/asterisk -g -p -U asterisk Unfortunately, asterisk always changes the cwd (current working directory) to '/' I checked that in /proc/.../cwd and with strace. I start asterisk as User 'asterisk', therefor it is not possible to write core dumps in /. How can I force asterisk to use /tmp as cwd? I have Debian Sarge with Asterisk 1.2.7.1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Germany VOIP provider
Thameem, 0180's are special. Some are billed per connection, some per minute. Typically the higher the next digit the more expensive it is. 0180 1 is same a local call from anywhere in Germany. See: http://www.elektronik-kompendium.de/sites/kom/0312221.htm --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!
Not sure this is exactly the problem we have, our call gets rejected by the device for some odd reason. I see. Here the call goes through just does not ring. Asterisk 1.2.7.1 and 1.2.10, no difference. If I pick up, the call connects just fine and is crystal clear. Just no rings :(. Maybe it's DOA. But more than one? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!
We have 6 or 7 SPA-2000's which all work with other installs of Asterisk but can't get a single one to receive calls using Asterisk 1.2.4. Ha! You're right. I just got some too and didn't even think of testing the ringer. Outgoing calls work fine, but incoming calls say "Call 1 State: Ringing" on the web interface and the call details are displayed but the phone does not ring. It obviously gets the SIP message that it should ring but it does not. Asterisk CLI also confirms that device is ringing. Increasing the ring voltage did not help either. Needless to say the same phone works fine with SPA 1000, 1001 and Grandstream. Interesting... any ideas what the heck is up with that? This is software version 3.1.9(LSa). I can't upgrade the software because the unit thinks it's not idle and hence does not start the upgrade process. Kind of disappointing. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need a pointer about scripting asterisk
So the pointer i need is... what would be a good way to tell asterisk to make a phone call from outside of asterisk. I would welcome any links to any docs, tutorials, etc... The easiest way would be to generate a call file and the putting it into the spool/outgoing directory. Asterisk will take over from there and place the call. See: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Make sure you create the file elsewhere and move it into the directory, and that it is readable by asterisk. Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text priority labels not working for me
The last log line suggests I can't use labels, but according to
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities it
shouldn't be a problem.
Labels work fine (and have been for a while). The snippet you provided
looks correct to me too. Are there are warning/errors when loading
extensions.conf? Does "show dialplan macro-dosomething" show the
correct labels and all priorities?
I.e.:
CLI> show dialplan macro-setcfwd
[ Context 'macro-setcfwd' created by 'pbx_config' ]
's' =>1. Playback(${ARG2}) [pbx_config]
2. Read(R|pls-ent-num-transfer|11||1|10) [pbx_config]
3. GotoIf($["${R}" != ""]?set)[pbx_config]
4. DbDel(${ARG1}) [pbx_config]
5. Playback(call-fwd-cancelled) [pbx_config]
6. Hangup() [pbx_config]
[set] 7. DbPut(${ARG1}=${R})[pbx_config]
8. Playback(${ARG2}) [pbx_config]
9. Playback(has-been-set-to) [pbx_config]
10. SayDigits(${R}) [pbx_config]
-= 1 extension (10 priorities) in 1 context. =-
--Luki
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Re: [Asterisk-Users] reinvite, DISA, and switching codec's.
James, Am I right in saying that because Asterisk has Answer()'d the call and done DISA(...), I can't do a re-invite to bridge the call between the PAP2 and the VoIP provider? Yes, you can reinvite after Dial()'ing your provider, but you probably won't be able to switch codecs once the call is connected. I may be wrong so just try it :). The ATA must be able to talk directly to your provider in such a case (i.e. not NAT). --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRTG54GS Capacity
Daniel, > Does anyone know how many simultaneous calls can a WRTG54GS handle? > Assuming SIP phones are connected locally using G711.u codec and the > WRTG54GS connects to a remote Asterisk server using IAX2 trunking > using GSM codec. Here are some of my experiences with Asterisk (I think 1.0.7) on WRT54G (not GS). You can definitely handle four concurrent calls. Possibly more but we didn't try it as we only had 2 x 2 port ATAs connected to it. Without transcoding (ulaw only) the CPU load is fairly low, but you are running fairly quickly out of memory -- both the flash to store anything and the RAM to run things. I think each thread shows up as a separate process in top on the WRT, so typically you see 10+ asterisk processes. With transcoding (ulaw <-> g726 in our case) the CPU gets fairly loaded with just couple calls, so I wouldn't plan for more than two concurrent calls with transcoding. Preferably none. These tests were done SIP to SIP, sorry not IAX. However, I don't think the protocol change would make a huge impact. We have finally dropped WRT as the platform because Asterisk crashed fairly frequently on it. Once a week pretty much guaranteed, sometimes every day with a call volume of about 200 calls/day. Canreinvite=yes crashed it every time reproducibly. The router itself stayed up for days if not months, so the safe_asterisk script restarted asterisk in a few seconds, but that's still less than ideal. This crashing is probably due to the old Asterisk version than anything else, but I didn't compile it; it was a binary that came from OpenWRT and I didn't have time to investigate this further. Couple weeks ago someone provided a package of a newer version for the WRT and I may try that and report. I still have the router in service (which currently is used only a router) so if you need more info let me know. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No reinvite - reason?
I put reinvite=yes in my sip.conf. For starters, it's canreinvite=yes. Then do a "sip show peer" on the peer and make sure it says that it can reinvite. Reasons why no reintives are even attempted include the transfer flag in the dial application and if the channel is monitor-ed (for obvious reasons). --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registered SIP:
Who is the file who listen when a softphone is run from a remote pc? -- Registered SIP '651' at 192.168.251.10 port 2209 expires 900 chan_sip.c --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Customer's voice not compatible with service?
That's awesome, can she make free long distance calls from payphones by dialing the number and chirping into the receiver to emulate the payment tones? That's great :)! Seriously, I found that if you use INBAND for DTMF and let Asterisk do the DTMF recognition you get less false DTMFs. The detector in the Sipura is too sensitive but you can avoid it by not using INFO or INFO+INBAND... just pure INBAND. Seems to help in the 1000, 1001 and 2000. Can't comment on the 2002. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questions from a working doctors' office installation
Michael, Is memory leak still as much an issue with 1.2.7 versus 1.2.5? In other words, is it worth it to upgrade a working, memory-leaking 1.2.5 to 1.2.7 or 1.2.8 just to potentially encounter other bugs in the new versions? Have other people been satisfied with the new versions so far? I have Polycom 501s and 301s. Call transfers are prone to crashing the system, getting sent to the wrong phone, etc. Huh... interesting... I had (and actually still do) have 1.2.5 version perfectly; it's been >60 days since the last restart so I figure I would have noticed memory leaks until now. This system is in a small real estate office with 15 extensions but with hundreds of calls a day, plenty of transfers. However, it's SIP only, no hardware, no IAX. Perhaps the memory leaks are specific to certain hardware or protocol or activity. Anyway, I'm not going to argue there are no memory leaks -- if you have them, try an upgrade :). Is there some sort of rollback function? I'm considering having a second PBX box for the upgraded version, then keeping the working production system as a backup. Yes. Here's what I do. I symlink the executable asterisk -> asterisk-1.2.5 and directory modules -> modules-1.2.5. When I want to switch versions, I change the symlinks for those two keeping everything else the same. No problems going back and forth, at least not between 1.2.x versions. When you build asterisk, don't do a "make install" but simply copy the executable to asterisk-VERSION and all .so files from the build directory to modules-VERSION -- i.e. cp -a `find -name '*.so'` /usr/lib/asterisk/modules-VERSION/. I run this in a chrooted environment, but you don't have to. My PSTN providers are voipjet (out) and Axvoice (in). Sometimes we have dropped calls incoming, or busy lines outgoing. Anyone else using good service providers they can recommend? That's something to the -biz list, probably but you may contact me off list if you need suggestions. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Way to disable codec in dialingplan
can we enable or force a codec on specified npa.. Depends on the channel. On SIP you can set SIP_CODEC to force a codec, but I don't think you can disallow one in the dialplan. See: http://voip-info.org/tiki-pagehistory.php?page=Asterisk+variables --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Trunk
Senario: If a call is initiated from Server 1 to Server 2, a trunk is established. While that call is progress another call is established from Server 2 to Server 1. Is a new trunk created, or is the same one used? I had exactly the same question and looked into this. If I remember correctly, a new trunk would be created in this case. Only case in the same direction are trunked together (i.e. if the second call would be from server 1 -> 2 it would use the existing trunk). You can verify yourself by watching the network traffic with tcpdump though. The packet size should give you the answer. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a WRT54G?
Why don't you put it up somewhere, if you need space I can put it on tel.net ? Yes, putting it up for download somewhere would be nice. I'd be interested too and I certainly can provide space for it too ... although that doesn't seem to be an issue, I see. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-1001 behind NAT -> Internet Asterisk box -- BOUNTY!
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. Looks like it wasn't a NAT of configuration problem after all... the SPA devices are quite nice, IMO. If there's a need, I guess Eric can explain it further... --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth via my Asterisk PBX
Am I correct in assuming that all calls from each organization would route through our Asterisk server & be passed off to the service provider That depends on your setup, on the provider and on the organization. If all support ReInvites and have them enabled, then it will work and the RTP traffic will flow between the organization and the provider. But each organization needs to be reachable via a public IP (i.e. not NAT) and the provider you use must support it too. I believe most(?) do, but you should check. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Running Asterisk as non-root
> If you run Asterisk as non root, you may have problems installing G729 > licenses. The digium registration utility has certain hard coded stuff, and > doesn't behave well when things aren't installed in the standard location. Good point. However, in the chrooted environment there is no need to make any changes to any paths or recompile asterisk. Asterisk thinks it's dealing with /var/lib/whatever while in reality it's accessing /usr/local/asterisk/var/lib/whatever. While I do not use g929, I don't think you would have a problem with the license install as long as you run the installation in the chrooted environment as well: chroot /usr/local/asterisk license-installation-script I don't have time to write up the steps to chroot asterisk, but if anyone is interested then I will tonight. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk as non-root
I saw where one should not run Asterisk as root: How important is this? It's probably not very important at the moment, however, it's not that hard to do either. I run Asterisk non-root and in a chrooted environment -- it keeps all necessary files nicely separated (easily portable, easy to switch versions), doesn't clog up common directories. Just make a new directory like /usr/local/asterisk and use that as the root for the chrooted environment. Chown all /var and /etc/asterisk files in there to the asterisk user and you're good to go. The tough part is to get all the shared libraries copies over -- ldd is your friend. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for input on which way to go withsmallbusiness setup
Terrelle, I've implemented a similar setup about a year ago. Here are couple observations worth sharing. YMMV, but these are my experiences: 1) A small LAN (~40 devices: PC, printers, phones) does not need QOS. Even when a workstation floods it with 100 Mbps traffic there is no quality problems one can hear (pings remain <1 ms anyway, no packet loss). 2) Get GOOD IP phones. The last thing you want is a phone crashing on you several times a day or during a phone call, or loosing connectivity, or having bad sound quality (accoustic feedback, hiss, etc). Saving here isn't worth it. 3) As people said, avoid FXO adapters. Go digital instead. 4) For 15 extension, you don't need a fancy machine. We used a PIII-800 with 512 MB RAM, it handles 10 calls at the same time just fine (load ~ 0.10); it's also the gateway for Internet traffic shaping, Windows logon server (samba), CUPS server and IMAP server. Using a new 2.6 kernel is key for scheduling and nice-ing processes accordingly. 5) Just like in your case, money was a concern. We decided to scratch the T1 or POST lines and use pure VoIP. No "phone lines" so your concurrent call limit depends on your available bandwidth. Why pay for 12 lines if one month you only use 4 but occasionally need 14 calls which you can't get with a fixed line setup? Initially we had DSL (2M down/768k up) but then went to Cable -- lower latency (~15 ms RTT to our PSTN gateway) and 10Mbps/1Mbps speed. So far (almost a year) everything runs great. Occasionally (half hour every couple months or so) connectivity isn't great (packet loss, latency) but this is acceptable to use given the cost savings compared to a T1. Bottom line in this case is that your Internet connection must be solid. This is a big variable though that required most time to get working right (including trying 40 ms packets to reduce the number of packets/sec; our modem was choking with >500 outgoing packets/sec), etc. It can be done. It all depends how much risk you are willing to take, and how important setup and operating costs are to you. Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extreme delay before * processes call files
> > But then the call file just keeps sitting in the > > /var/spool/asterisk/outgoing directory and it seems that * is doing > > nothing with it?? Only after 10-30 seconds sometimes even much longer > > the call file is picked up. Check if the system times are in sync; if you copy a file with samba, the timestamp on the file is set by the samba client (Windows?). If the client's time is in the future (compared to the * server time) the call file won't be run until the server time catches up. You can just have the server touch the call file before moving it to the spool directory; this will update the timestamp to the current time. Give it a shot. Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: update - 512 Simultaneous Calls with Digital Recording
> Has anyone seen these solid state "Drives" from gigabyte yet? - > http://www.pcper.com/article.php?aid=224&type=expert&pid=3 Interesting device. Looks like the burst throughput is right on par with good drives, but you have better sustained throughput and obviously near zero latency. But what truly is the advantage compared to having 4 GB (dedicated) RAM in the machine and making a RAM disk with it? You need the RAM either way and that ought to be at least as fast as this card on a 33 MHz PCI bus. You loose the "non-volatile" advantage but that's about it, no? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Vonage
> Something I've been curious about is if it is possible to stick their > ata on a extra ethernet port on an Asterisk server and have the Asterisk > server spoof the Vonage server. Then, do a man-in-the-middle type thing > to use the ata for authentication, but have Asterisk handle all the calls. That would work assuming you write a transparent enough proxy that would forward all SIP traffic to the ATA but intercept REGISTER and INVITE messages that contain authentication data. Not quite trivial, but doable over a weekend. The question is, is it really worth it? The deal you get with Vonage isn't all *that* great. You can find as reliable termination/origination elsewhere with open credentials for the same price (or cheaper) if you look around... assuming typical residential usage. > Perhaps another idea is to hammer an ata with authentication requests > and create a long list of nonces and hashes that you replay back to the > server as needed. Not a good idea (all legal and ethical implications aside). Given an 8 byte hex challenge (32 bit) you would need 64 GB of space to store the MD5 hashes for all nonces. Assuming you can attack the ATA with 100 requests a second you would need more than a year to collect all the responses... and who says the credentials do not changed periodically and the ATA fetches new config from Vonage? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open Asterisk database
Erick, > I now want to run asterisk with the -U asterisk and -G asterisk > credentials. When I do it I have the error (asterisk is a valid > user/group in the system with nologin as shell): > /var/log/asterisk/messages: > WARNING[5230] db.c: Unable to open Asterisk database > WARNING[5230] db.c: Database unavailable I believe this refers to the AstDB not the mySQL database. Make sure the astdb file is writeable by user asterisk. The file is usually in /var/lib/astdb. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to reinvite twice?
Gabe, > If my setup goes: Phone => asterisk => asterisk => PSTN termination provider > Can I define "canreinvite" on both asterisk boxes so the phone call will go > directly to the PSTN provider? Yes, you can reinvite multiple times. The media path will collapse as much as possible. It works reliably, unless the two servers are too close to each other. Don't ask my why. My server is 1.2 ms RTT away from my provider's server and on about 50% of the calls I end up with a "482 Loop Detected" response to my reinvite on incoming calls. I found that putting a Ringing entry in my dialplan and a 0.5 second delay before Answer fixes it. I tried tracking this down but didn't have much luck. It's fine on outgoing calls. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changing REINVITE status of the channel dynamically
> I'd like to know if it's possible to set the REINVITE on or off dynamically,
> based on the extension being dialed.
Define two peers in sip.conf, one with canreinvite=yes and the second
with canreinvite=no. Then you can route your calls with or without
reinvites depending on the dialed number. Like:
[provider-reinvite]
type=peer
host=external_sip_server.com
canreinvite=yes
...
[provider-noreinvite]
trype=peer
host=external_sip_server.com
canreinvite=no
...
exten => _1[0-4]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
exten => _1[5-9]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
--Luki
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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
> Actully ethereal OK... Try canreinvite=yes in the [general] section; this makes it the default setting for all peers unless specified otherwise. Do the same for nat=no in [general] to rule out all NAT'ing related issues. You don't have tT in your Dial() statement, that's good. You say you verified that no transcoding is needed (i.e. both ends use the same codec). Well, then it should work! Once you get it to work, you can individualize the accounts and no longer use a global setting. But that's down the line. > asterisk always creates a 'native bridge' and seems to hold on for dear > life so far as I have seen :-) It says "Attempting Native Bridge" but it doesn't tell if you if it succeeded or not; there was once a notice saying the the bridge could not be established (failed?) but it caused even more confusion. You could add some statements to the ast_rtp_bridge() code in rtp.c and give yourself some feedback -- succeeded, failed because X / Y / Z. Hope that helps... --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
Steve, > The mission is to actually get a reinvite to work on the lan. There isn't anything special to get this working... normally. I trust you verified the traffic flow with a network monitor tool (tcpdump?), correct? Does SIP debug give you any info (i.e., does it match the right peer) -- you don't show if you allow reinvites globally? What about the nat= setting? Couple pointers I can give you to get you excited: 1) Reinvites work quite reliably, I use them between the PTSN gateway and the end user's ATA, all the way across the Internet -- nicely reduces latency. 2) If you use RFC2833 for DTMF you can issue an reinvite and still use t/T for transfer. NOTE that you have to modify the source to make asterisk reinvite even when it needs to listen to DTMFs. I give no guarantees how well it will work for you but it does work. See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c. 3) Reinvites *can* work even if both ends are behind NAT. It really depends on the NATing router and the ATA. Sipura's and good NAT routers work, but I would not call it "reliable" -- it's really pushing it a bit... So if you really want to see why your Reinvites do not work, then you probably will have to make your hands dirty and analyze where ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it makes the situation a lot easier. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring detection using SIP - Broadvoice addon line detection
Last time I checked, Broadvoice sent the Alert-Info header in the INVITE message. The main line does not have this header, an add-on line does. On 1/22/06, Robert Mann <[EMAIL PROTECTED]> wrote: > Can * detect distinctive ringing on a SIP line? The reason I ask is I have > broadvoice with an add on line. It does not send any type of info that I > know of for the two separate lines so I can not determine which number is > ringing. Broadvoice can however send distinctive ring tones so if I could > intercept that I could tell which line was ringing. Or does anyone have any > other ideas to offer? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Fax part 2
Just a quick note, make sure /var/spool/asterisk-fax/ exists and is writable by whatever asterisk runs as. Assuming your connection is good enough, I had "LINE ERROR"s because app_rxfax aborted right after the handshake as it would not write the output file. I think there is a debug mode in app_rxfax that may shed some light. Otherwise see Alexander's reply about the connection quality. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
