[Asterisk-Users] Pattern-matching in the dial-plan
Hey all, I'm trying to add some logic to a dial-plan to allow the caller to terminate a number with a #, but also accept it without this terminator. While trying this, I noticed that, for example, extension _[*0-9]XXX.# always seems to match, whether the last digit dialled is a # or not. It's as if the parser assumes everything after the . will match and doesn't look any further. Is this expected behaviour? If so, what would be the best solution to my problem? I currently solved it by avoiding the . and matching every possible number-length seperately, both with and without the #-terminator. It works, but seems like it should be doable with just 2 matches. The box I'm trying this on is running the CVS HEAD of about a week ago. Thanks in advance for any suggestions. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFAX/spandsp
Hey Steve, On Mon, Mar 22, 2004 at 22:13:37 +0800, Steve Underwood wrote: Hi all, If you have had trouble with multiple concurrent channels running app_rxfax or ap_txfax, where was a silly bug. Updated versions are available at ftp://ftp.opencall.org/pub/spandsp The latest spandsp-0.0.1f seems to working for quite a lot of people. I guess there will still be plenty more compatibility issues to deal with, though. If you try this software and find reliability problems try checking for frame slips. A single frame slip in the audio stream will kill a fax transfer. This is true for any fax modem, and is not a spandsp issue. Just to let you know that this is the first version that has actually worked for me. Previous versions seemed to abort at the beginning of the transfer, no matter what I tried. This one worked out of the box, using the exact same test-setup the earlier version failed in. :-) Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Dutch PSTN-line on X100P
Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves, best described as talking in a bathroom. It very much sounds like very bad feedback, most of the time. The system concerned is a PIII-750 with an X100P and a TDM400P. The problem occurs when a call is bridged between the X100P and a port on the TDM400P. Calls from the TDM400P to a remote IAX2- connected box seem OK. rxgain / txgain are at their default (0) for all interfaces and I'm using the default echo-cancellation settings, with echocancelwhenbridged enabled. The PSTN-line concerned is a standard analog line from KPN, which also has ADSL coming in over it. The X100P is, ofcourse, connected behind the splitter. When I connect the X100P to an internal analog port of a small legacy home-switch, the problem doesn't seem to occur, either. I'm using opermode=1 for the X100P, but read on this list somewhere that the chip used on the X100P is actually not the international version, so this setting might not have any effect. In this case, it doesn't seem to matter if I use it or not. There's no notable difference. I'll probably investigate this problem further and play with the echo- canceller and rxgain / txgain a bit. Just curious if anyone else is experiencing something similar and might have found a solution already. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Dutch PSTN-line on X100P
Yo Eric, On Thu, Oct 02, 2003 at 11:56:44 -0500, Eric Wieling wrote: Check /proc/interrupts to make sure the cards are not shareing IRQs with anything. Sorry, forgot to mention it. All Zaptel-cards in that machine already have their own unique interrupts. I will try moving the cards to different slots, though. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD-gateway prefix
Hey Chris, On Thu, Aug 07, 2003 at 04:12:57 +0200, Chris Wetemans wrote: - Original Message - From: The Traveller [EMAIL PROTECTED] Hey all, As there seem to be some problems with DTMF-signalling between chan_sip and several clients, due to which many could not properly dial a number at the dial-tone of the XS4ALL-gateway at FWD-number 42442, I've now arranged for a prefix on FWD for this gateway. From FWD, you can now dial 1010-666, followed by the Dutch toll-free number or IAXTel-number you wish to reach, as you would have dialled it from the dial-tone at FWD-number 42442. I've tried dialing the following from my FWD-client (X-lite): 1010-666-800-0402 1010-666-0800-0402 1010-666-31800-0402 none of them worked. Am I doing something wrong? The correct way to dial Dutch toll-free numbers using the gateway-prefix is: 1010-666-0800-rest of number I haven't tried the *31(800)... they mention on their site yet, but I don't have special provisioning on the gateway for it and the first time I heared about it, was in their newsletter. They're either using a different gateway for it, or re-write the number to use the prefix internally. I'm currently only allowing 0800-0101 (KPN calling-card and collect-calls, IVR-system) of the short 0800-numbers, as some of them are service-numbers, tied to the phone-line calling them, on which you can change service, request accumulated charges, etc. for the line from which you're calling, without any other authentication. Not a good thing to allow anyone to access for our lines. :-) I might have a closer look at it in the future and allow the non-service short numbers only. All long numbers (0800-XXX) are allowed through the gateway. On the FWD website it says you can dial 31-800-0402 directly , but that doesn't work either 0800-0402 has never worked through the XS4ALL-gateway. Where on their site did you read that this number should work? Maybe it was on one of their forums and they didn't write it themselves? Otherwise, you should mail them about the error on their site. BTW: The PSTN access-number on their site is also incorrect. It should be +31 20 3987567. Although the old number they mention still works, it only has limited capacity and will probably be taken out of service at some point. www.gnophone.com also still mentions the old number. Furthermore, they mention you have to dial 170099number through it to reach FWD-numbers. At the moment, you can dial any FWD-number directly, which eliminates the extra hop in the call, as it won't have to go through IAXTel. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AgentCallbackLogin
Hey Jim, On Wed, Aug 06, 2003 at 15:12:50 -0500, Jim Friedeck wrote: I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: [...] extensions.conf: [c_in_1];internal lines (up to 48 agents and admins) exten = 400,1,AgentCallbackLogin(|c_in_1) [...] I don't understand where the default context comes from in the message 'No such extension/context [EMAIL PROTECTED]'. Where do I tell the queue app the proper context? Any ideas? Try adding an @ in front of the context in the argument to AgentCallbackLogin, like so: exten = 400,1,AgentCallbackLogin(|@c_in_1) Also make sure you're running the latest CVS-version, as the functionality has just been added yesterday or so. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 and reinvites
Hey Dan, On Mon, Jul 28, 2003 at 22:50:21 -0300, Dan Fernandez wrote: Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes. AFAIK, IAX already does that by default. It's called native bridging. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD-gateway prefix
Hey all, As there seem to be some problems with DTMF-signalling between chan_sip and several clients, due to which many could not properly dial a number at the dial-tone of the XS4ALL-gateway at FWD-number 42442, I've now arranged for a prefix on FWD for this gateway. From FWD, you can now dial 1010-666, followed by the Dutch toll-free number or IAXTel-number you wish to reach, as you would have dialled it from the dial-tone at FWD-number 42442. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get musiconhold to work
Hey AJ, On Fri, Jul 25, 2003 at 19:23:50 -0400, [EMAIL PROTECTED] wrote: I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash I get no music. The sample mp3 file is in the mohmp3 directory. Does anyone know what I might be doing wrong or how I might be able to correct it? Also I have tried assigning a extension with the MusicOnHold application and it still doesn't seem to work. I'm running RH9 here as well and came across the same problem. RH9 seems to use mpg321 by default and, when you run `up2date' or a similar tool, that will supercede mpg123 and replace it with mpg321. To add to the confusion, RH9's mpg321-package also symlinks mpg321 to mpg123. I solved it by putting the real mpg123 in /usr/local/bin/ and changing the path accordingly in apps/app_mp3.c and res/res_musiconhold.c. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with chan_sip on multi-homed hosts
Hey all, I'm experiencing a problem with chan_sip on a multi-homed machine. The machine has 1 interface to the rest of the world and 1 interface on a local network. The local network has public IP-addresses, though, and the IP-addresses of both interfaces are reachable from the outside world, but by default, outgoing traffic from that machine to the outside world will have the IP-address of the interface the default-route points to as it's source, which is the one with the outside world behind it, obviously. I've set the bind-address in sip.conf to the IP-address of the interface on the local network, because I want to force it to only use that interface's IP-address. This works great for binding only to that IP-address and I can even make outgoing calls, but when registering to a remote SIP-provider, chan_sip seems to use the IP-address of the wrong interface (the one the default-route points to) as the source of it's registration-requests, so as soon as a call comes in from that SIP-provider, it's sent to that IP-address and fails, as chan_sip isn't listening on it. BTW: The remote SIP-provider is FWD in this case. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger(4.7) Setup
Hey Neel, On Fri, Jul 25, 2003 at 10:40:55 -0500, Neel Datta wrote: Thanks Roy, I this worked! Only one thing I can't seem to do- If I have a password set in my sip.conf as in the 'secret' key, I can't get the msn client to authenticate properly. (And yes, I'm typing the exact same word I have in secret =) Try appending @asterisk to your sign-in name in Messenger's configuration. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio in Messenger
Yo, I'm trying to get Asterisk working with Messenger 4.7. After skimming through the list-archives, I've got it to register to my Asterisk-box and can make calls. Unfortunately, there's no audio from the Messenger- side of the call to the other caller. I can hear the caller in Messenger, though. It doesn't appear to be a Messenger or network-problem, as I can talk to FWD from it just fine. This same problem also shows up on the PSTN to FWD-gateway I just set up. If the other end of the call uses Messenger, there won't be audio from it. It works fine with X Lite, tried from the same machine. I've tried everything from using different CODECs to canreinvite=no, nat=yes, insecure=yes, etc, without much luck. Does anyone have an idea or got it working? I'm using the latest CVS for all Asterisk-stuff. Thanks in advance! Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] XS4ALL Gateway now also does FWD
Yo all, I just added FWD (http://fwd.pulver.com/) to the XS4ALL PSTN-gateway. Here's a quick update on how it works: VoIP: From IAXTel, dial 31800rest of number for Dutch toll-free numbers. FWD is not (yet) directly reachable from IAXTel. I'll talk to Mark to see if he's intrested in setting this up. From FWD, dial 42442, wait for the dial-tone and dial your number. This can be any IAXTel-number or Dutch toll-free PSTN-number (starting with 0800). PSTN: From the PSTN, dial +31 20 3987567, wait for the dial-tone and dial your number. This can be any IAXTel-number or 5-digit FWD-number. Note that I'll probably include the other numbers and prefixes on FWD shortly. For now, it only does the standard 5-digit numbers. Let me know if you have any questions or comments. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail instructions
Hey Florian, On Wed, Jul 16, 2003 at 13:56:45 +0200, Florian Overkamp wrote: Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user creates their own message which includes 'please leave your message after the tone' or similar, so the generic message is undesirable (or should be override-able). Is there something in the apps I've missed that allows this already ? That's what the s is for. Use it together with the b or u to suppress the recording-instructions. This only works with Voicemail2, BTW, as the original Voicemail-app doesn't allow it to be used together with the other options. - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Mark talked about adding an extra menu some time ago and this was one of the features discussed, if I'm not mistaken. I'm not sure what the status is at the moment. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
Hey Jan, On Wed, Jul 16, 2003 at 11:45:13 -0700, Jan Rychter wrote: Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? I have the exact same problem over here, and it seems some others on the list and on IRC have it as well. I'm still using IAX1 because of it. My setup is very similar to yours, except that I also use SIP and H.323- phones locally, which doesn't seem to make any difference. Both Asterisk-boxes have Zaptel-hardware in them (which seems to make a difference, as IAX2 uses it's timing, if available), I use GSM and trunking is off (turning it on didn't solve this problem, though). Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog commands
Hey Jay, On Tue, Jul 15, 2003 at 18:41:12 +1000, Jay Tyndall wrote: Hi, When I use the analog phone connected to Zap/1 how do I transfer hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? I had the same problem with the standard flash and pulse-timings in Zaptel. Some phone-systems use a flash with a length of about 150ms, which is way too short for the standard Zaptel-config, which assumes your flash is at least 750ms long, and sees the short flashes as a pulse dialled 1 (which is what gets sent in DTMF to your caller). The following changes to the Zaptel-sources fixed it for me: Index: zaptel.h === RCS file: /usr/cvsroot/zaptel/zaptel.h,v retrieving revision 1.11 diff -r1.11 zaptel.h 775c775 #define ZT_DEFAULT_FLASHTIME750 /* 750 ms default flash time */ --- #define ZT_DEFAULT_FLASHTIME150 /* 150 ms default flash time */ 789c789 #define ZT_MAXPULSETIME (150 * 8) /* 150 ms maximum */ --- #define ZT_MAXPULSETIME (80 * 8)/* 80 ms maximum */ If possible, you could ofcourse also program your phone to generate the longer flashes and leave Zaptel at it's defaults. Good luck! Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does callerid= in sip.conf do?
Yo BK, On Sat, Jul 12, 2003 at 11:52:42 +0900, BK [address only for mailing lists] wrote: Hi since callerid= in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? Works perfectly over here. My guess is that the calls you make with that phone aren't coming in over the user / peer-entry you're expecting, for some reason. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
Hi bk,
On Wed, Jul 09, 2003 at 17:16:55 +0900, BK [address only for mailing lists] wrote:
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat = 9
exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten = _9[123456789]XXX,2,Congestion
this is properly included in the handsets' context but the dial tone
disappears after pressing 9.
am I missing something?
I had the same problem here and discovered that ignorepat only works if
it's placed in the actual incoming context of your channels and not if
it's included from another context. Not sure if this is a bug or a feature.
So, try placing the ignorepat in your handset-contexts instead.
Grtz,
Oliver
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[Asterisk-Users] PRI with variable length numbers
Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same digits. In that case, dialling will be signalled as complete as soon as the shortest of the numbers is dialed. An example: exten = 31801,1, ... exten = 3180,1, ... In this case, dialling 3180 will immediately start ringing, while on similar setups, with TDM40B's and analog phones, Asterisk will wait for the duration of digittimeout for more digits, if it can't be absolutely certain that the dialled number is complete. If I remove the 3180 in the above example, ringing will only start after 31801 is fully dialled. Relevant configs: /etc/zaptel.conf: bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf: switchtype=euroisdn signalling=pri_cpe immediate=no Any ideas? Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI with variable length numbers
Hey Martin, I'm not receiving fixed-length numbers on that PRI and it really seems to be the Asterisk end which decides when dialling is complete. I've arranged for a block of numbers, starting with 3, to be routed from the Meridian to Asterisk, over this PRI. As long as the numbers I set up in my dialplan don't start with the same digits, I can put in numbers with varying lengths and it works OK, dialling them from an extension on the Meridian. Don't ask me how, because I'm not really into the low-level signalling-protocols on PRI's yet. I could send you pri intense debug-output of some calls off-list, if you want to see for yourself. For example, I have set up several extensions in the range of 3000 to 3099 for IP-phones, and those can all be dialled directly. I also included a context in the incoming PRI-context which allowed dialling IAXTel-numbers, with a prefix of 3, and you can now indeed call 31700rest of number from any extension on the Meridian for an IAXTel-number. The Meridian has no special programming to differentiate between dialled extensions in the 3-range, so my conclusion is that Asterisk signals when dialling is complete. As Asterisk seems to be the one determining when the number is finished, it seems possible to use the same extension matching-logic as on analog ports. Grtz, Oliver On Wed, Jul 09, 2003 at 13:25:33 -0500, Martin Pycko wrote: Why do you see the problem with that ? How would you use that functionality that analog channels have ? With PRI you're going to always receive the full number that was dialed. So since you have a limited number of DID's do the matching one for one and it'll work. regards Martin On Wed, 9 Jul 2003, The Traveller wrote: Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same digits. In that case, dialling will be signalled as complete as soon as the shortest of the numbers is dialed. An example: exten = 31801,1, ... exten = 3180,1, ... In this case, dialling 3180 will immediately start ringing, while on similar setups, with TDM40B's and analog phones, Asterisk will wait for the duration of digittimeout for more digits, if it can't be absolutely certain that the dialled number is complete. If I remove the 3180 in the above example, ringing will only start after 31801 is fully dialled. Relevant configs: /etc/zaptel.conf: bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf: switchtype=euroisdn signalling=pri_cpe immediate=no Any ideas? Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TDM40P
Hey Adam, On Tue, Jul 08, 2003 at 00:58:08 +1000, Adam Goryachev wrote: Without quite just saying me too, see below... [...] My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones I have an AMD XP 1800 with 256MB RAM, and a single IDE HDD. I have a X100P, a TDM40P and a ISDN BRI card using ISDN4Linux driver. [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 0:1929531 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci, usb-uhci 8: 1 XT-PIC rtc 10: 18895682 XT-PIC eth0, wcfxo 11: 23378 XT-PIC HiSax 12: 17057246 XT-PIC wcfxs 14: 45267 XT-PIC ide0 15: 13 XT-PIC ide1 NMI: 0 ERR: 2 I actually also have an ISDN BRI-card in there, but it was not in use during these tests. My problems with this setup are as follows: - I hear a soft fading hiss in the background on the TDM40P-connected phones. It's also present when I call something like voicemail or a conference on my local box, so it doesn't seem to come from the X100P. If I uncomment the NO_CALIBRATION-option in the Zaptel Makefile, the hiss decreases in volume, but is still audible. I have the same problem, but it seems to only be on one phone. Originally I had all 4 extensions passed across a single CAT5 cable, now I have 4 separate CAT5 cables with RJ45 connectors on them and it seems better, but it is still there. I also get a funny 'ring' tone if I am in Currently, some of the lines run over the same CAT5-wire for me as well, but I also have some phones which are connected directly to the TDM40P, with at most 2 meters of wire. They have the same problems, though. voicemail/conference/phone call/whatever and another extension rings (sort of like a crossed line perhaps, but it isn't the sound of the ring, perhaps the actual ring voltage being applied/leaked to my line). I've noticed this leak as well. Besides that, I can also clearly hear the caller-ID and MWI-status beeing signalled to other extensions in the background. It doesn't seem to be my wiring, as I also hear it with 2 phones connected directly to the TDM40P, over seperate cables. - Some of my TDM40P-connected phones go dead frequently. By dead, I mean that the line-voltage goes away, so there's nothing there when they're picked up. When this happens, the remedy is to call that extension and let it ring once, or restart Asterisk, after which the line-voltage re-appears. I have this problem with several different phones and know at least 1 other person with similar problems. I have the same problem, usually all 4 lines will go at the same time, but while I was re-cabling (lots of pickup, listen for dialtone, hangup cycles on each line to see if my cabling was working) and I got this a number of times. ie, one or more extensions not working. My fix was a stop asterisk, remove modules, insert modules, start asterisk. Yups, I certainly noticed that lots of unplugging / reconnecting of phones also seems to trigger it. I've also had a number of problems where all TDM400 extensions are dead but the X100P and ISDN still work fine. When I stop asterisk, and remove the modules, all is OK, but as soon as I insert the modules, the machine locks up, and a power off/on is the only thing that works. I don't really know how to track this one down, or how to diagnose it to better assist other people to fix the problem That's intresting. I'm seeing that one as well. About 1 in 10 to 20 times I unload and load the Zaptel-drivers, wcfxs prints some sort of register- dump and informs me that the Freshmaker register-test failed, after which the machine is locked up solid. The register-dump contains mostly ff's in such a case and there doesn't seem to be a kernel-panic. I suspect the wcfxs driver-init of the TDM40P does some pretty nasty stuff, as even during a regular and succesfull init, the machine becomes completely unresponsive for the duration of the init. I suspect this frozen state doesn't get undone properly in some failure-situations, which leaves you with a frozen machine if that happens. - Making data-calls from a line on the TDM40P, through the X100P, to an external number is almost impossible. Even 2400bps is unreliable, I've not tried data calls, but I think there is a d parameter to the Dial application (see show application dial) which assists with data calls. Are you using that? Yups, I've added the d, but it doesn't appear to make any difference. Thanks for your information. Hopefully, Mark will be able to shed some light on it as well. I'm also available for supplying more debug-info, if needed. Grtz, Oliver ___
Re: [Asterisk-Users] Direct entry to your own voice mailbox
Hey Dan,
On Mon, Jul 07, 2003 at 18:47:07 +0300, Dan wrote:
Hi,
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
Sure. Pass the mailbox as an arg to the VoiceMailMain or
VoiceMailMain2-app. See show application voicemailmain on the
Asterisk-console for more info.
I want to be the same extension for all phones, not a specific one for each
of them.
I think you could do that by passing ${CALLERIDNUM} as the arg.
It is possible to have a time stamp in the recorded message? I want to know
when the message has been recorded.
I think someone here was working on a patch for that, which was waiting
for prompts to be recorded. Not sure of the current status.
Grtz,
Oliver
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Re: [Asterisk-Users] Direct entry to your own voice mailbox
Hey Dan, On Mon, Jul 07, 2003 at 20:42:16 +0300, Dan wrote: It is possible to have a time stamp in the recorded message? I want to know when the message has been recorded. I think someone here was working on a patch for that, which was waiting for prompts to be recorded. Not sure of the current status. Why other prompts? There is an application available (DateTime) to say current date and time. It cannot be integrated in the Voicemail appication? I guess it could. It's just that the bigger patch this enhancement was part of, required some new prompts for an extra menu one could enter while playing a message. If I remember correctly, Mark was thinking about adding this functionality, although it was some time a go and he seems very busy at the moment. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Hi Jim, You're probably not receiving disconnect-supervision on your analog lines, or have Zaptel configured incorrectly to recognize it. Check the list-archives (available from www.asterisk.org). You could try the busydetect-statement in zapata.conf. Also check Asterisk's main Makefile for some options related to busydetect. I strongly recommend the improved busydetect-routines (BUSYDETECT_MARTIN), which are not the default yet. Make sure that your voice-menu's always have a timeout, so it's impossible for your system to get stuck in one in the first place, should a caller not hang up or this fact not be detected by Asterisk. Grtz, Oliver On Wed, Jul 02, 2003 at 14:34:56 -0400, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk
Yo all,
As there has been some intrest, here's my updated version:
I post it to -dev as well as -users, as it may be of intrest to
both.
Inspired by the example in the tips tricks-section of
http://www.junghanns.net/asterisk/;, I built a more elaborate
set of features. Currently, my implementation supports call-
forward unconditional, on no answer and on busy. It furthermore
provides each extension with a set of user-definable speed-dials.
It validates if the number entered is actually valid for the current context
and additionally saves this context into the DB and always uses it to
originate the divert from, as you can't expect the forward destination-
number to be available from all caller's contexts. Only the
speed-dials use the extension's current context, but do save the
context a speed-dial was entered from into the DB anyways.
Some things you need to know:
- No check for forwarding-loops yet. I haven't tried what happens
when you create one, either. :-)
- When a call gets forwarded to another local extension and goes to
voicemail from there, voicemail will be left for the extension that
originally forwarded the call, instead of the extension the call was
forwarded to. This might be seen as a feature.
- The contexts to set the features are to be included into the
context of each extension you want to allow to use them and you
should replace your macro-stdexten with my version (which accepts
some more args than the original version, to allow for passing most
arguments to the Dial-application. This means you have to change
either your extension-entries or the macro, if you're already using
macro-stdexten.
- If you where using my first version, you should remove all related
entries from your ast-DB, as I modified the format somewhat.
- As you can see, it currently uses Festival for it's prompts,
which should obviously be installed and working with Asterisk.
Without it, everything will probably still work, but no prompts,
except the standard invalid-recording, which is used when entering
invalid numbers, will be played.
- I'll probably add more features when I have time, and post an
update. Also, this stdexten-macro, like the original one,
assumes everyone has voicemail. Making a second copy without
voicemail or adding an extra argument to enable / disable VM-
processing should be trivial.
Usage:
921 + number - Set unconditional forwarding.
921 - Cancel unconditional forwarding.
9921 - Check unconditional forwarding.
961 + number - Set forwarding on no answer.
961 - Cancel forwarding on no answer.
9961 - Check forwarding on no answer.
967 + number - Set forwarding on busy.
967 - Cancel forwarding on busy.
9967 - Check forwarding on busy.
970 - 999 + number - Set a personal speed-dial.
970 - 999 + 0 - Clear a personal speed-dial.
970 - 999- Call a personal speed-dial, if set.
9970 - - Check a personal speed-dial.
Note that I choose 9 instead of * because a lot of IP-phones
don't allow the user to dial a number containing the * or #.
Nothing stops you from changing it as needed.
Also note that this is almost my first go at it, and I haven't tested it
very heavily. Comments, suggestions and additions are welcome. Hope
it's useful to some of you.
Grtz,
Oliver
;
; Macros
;
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Timeout
; ${ARG4} - Other options to app_dial
;
exten = s,1,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFU)
exten = s,102,Goto(s|4)
exten = s,2,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFUC)
exten = s,3,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,4,Dial(${ARG2},${ARG3},${ARG4})
exten = s,105,Goto(s|205)
exten = s,5,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFNA)
exten = s,106,Goto(s|8)
exten = s,6,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFNAC)
exten = s,7,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,8,Answer
exten = s,9,Voicemail2(su${ARG1})
exten = s,10,Hangup
exten = s,205,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFB)
exten = s,306,Goto(s|208)
exten = s,206,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFBC)
exten = s,207,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,208,Answer
exten = s,209,Voicemail2(sb${ARG1})
exten = s,210,Hangup
;
; Special features, Call Forwarding, unconditional.
;
[feature-cfu]
exten = _921X.,1,Answer
exten = _921X.,2,ChanIsAvail(Local/${EXTEN:[EMAIL PROTECTED])
exten = _921X.,103,Playback(invalid)
exten = _921X.,104,Hangup
exten = _921X.,3,DBput(FEAT/${CALLERIDNUM}/CFWD/CFU=${EXTEN:3})
exten = _921X.,4,DBput(FEAT/${CALLERIDNUM}/CFWD/CFUC=${CONTEXT})
exten = _921X.,5,Festival(Call-Forward Unconditional: Has been set too: ${EXTEN:3}.)
exten = _921X.,6,Hangup
exten = 921,1,Answer
exten = 921,2,DBdel(FEAT/${CALLERIDNUM}/CFWD/CFU)
exten = 921,3,DBdel(FEAT/${CALLERIDNUM}/CFWD/CFUC)
exten =
Re: [Asterisk-Users] fixed point mec3
Hey Mark, I wouldn't make MEC3 the default just yet. I was just testing with MeetMe, which was one of the things where MEC3 went wrong for me in the past. After around 8 channels from an E100P-connected PRI joined the conference, everything became one big chaos of noise. When enough channels leave, the audio returns back to normal again. There's definately an improvement, as the previous MEC3 failed in about the same way with just 1 channel in the conference, as far as I remember, but that test was a while back and many other things have changed in the meantime. This is on a dual Xeon 2.4Ghz with plenty of RAM, running RH9. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP only with no soundcard?
Hey Matt, On Sun, Jun 29, 2003 at 14:14:18 -1000, Matt Darnell wrote: Aloha Oliver, That was it! Thanks, I am going to download the eStara softphone and try to talk from one phone to another! Nice that it worked. An alternative method might be putting the line noload = chan_zap.so into the modules-section of Asterisk's modules.conf-file. I just realized that some of the Good Stuff in Asterisk, like conferencing and music on hold, needs Zaptel as a timing- source and you might wish to install it again, using ztdummy to get your timing from the USB-controller instead of a Zaptel-device. Ohwell, let's first get your 2 softphones to call eachother. There's more than enough other Good Stuff to explore in Asterisk until you get to that part. :-) Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail issue
Yo Dan, Try adding the s to the arguments you give to VoiceMail2, so, for example, Voicemail2(sb1000) for the busy-message of ext. 1000. Note that only Voicemail2 allows the s to be used together with b and u. Grtz, Oliver On Fri, Jun 27, 2003 at 15:04:44 +0300, Dan wrote: Voicemail It is not a bug.. Just do not want to record a separate message like Hi, I'm unavailable to answer your call...blah blah blah and another one for Please leave your message after the toneblah blah blah because this one (the last one) is common for all the mailboxes, so a single voice must be used for all users, which sometimes is unacceptable. There is any practical reason to break this message in two pieces? BR, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 27, 2003 2:52 PM Subject: Re: [Asterisk-Users] Voicemail issue Don't know.. Are you using voicemail or voicemail2?? Maybe you have found a bug.. Hi, Nope. I have recorded my own busy and unavailable message from the '0' menu of my voice mailbox. When someone is redirected to the mailbix, it hears both of them... first my recorded message, second the default one. I check that on two separate Asterisk boxes. I have the latest version from the CVS. What else can be done? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 27, 2003 1:46 PM Subject: Re: [Asterisk-Users] Voicemail issue It should do that already.. when you record your busy and unavailable message it should overwrite the default ones.. Hi,. How can I make that Voicemail app to play only my own recorded message without the default one? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
Heya all,
I ran some more tests with different kernel-options and my preliminary
conclusion is that the problem goes away when you disable SMP in your
kernel. I even put the Eicon-card, which I suspected was causing the
problem, back into the machine and loaded it's drivers, making calls
through it during my stress-test, but the machine is still stable.
Hopefully, it stays that way. These panics sometimes take a bit
longer to show up.
Be sure to keep the CPU-option to use the local APIC checked in your
kernel-configuration, or you'll only have the 16 lower IRQ's available,
with a high probability that your devices end up sharing IRQ's. In my
case, both SCSI host-adapters, the ethernet, the Eicon-card and the
Zaptel-drivers suddenly all decided they wanted IRQ 5, even while there
where others available. :-)
PS: I had uncommented the #define for SMP in the Zaptel Makefile while
testing with SMP and have now commented it again, while testing with a
non-SMP kernel, in case anyone wonders.
Alex: Could you try a non-SMP kernel on your machine as well and report
if it fixes your problem? I'm using the standard GCC shipped with RH9,
which identifies itself as
gcc version 3.2.2 20030222 (Red Hat Linux 3.2.2-5), for all my tests.
The kernel I'm currently using is the v2.4.21 I started with, but with
SMP disabled.
Grtz,
Oliver
On Thu, Jun 26, 2003 at 12:27:12 -0500, Matthias Granberry wrote:
Also, make sure that the kernel and all the modules are compiled with
the same gcc version. I had to manually change some hardcoded zaptel
makefile targets to use gcc-2.95 instead of ${CC} or gcc hardcoded in.
The entire asterisk build system is somewhat weak, but it seems to
work if you tweak it all just right. It's obvious what things the
developers are interested in, though. The boring parts are all
half-done, and the interesting parts are all fairly high-quality.
Matthias
The Traveller [EMAIL PROTECTED] writes:
Hi Alex,
The problem is most likely to occur with high volumes of call-setups and
disconnects. This could be reproduced by putting 2 of your T-1 ports
back to back and then using the auto-dialer to generate a large amount of
very short calls between the ports.
I'm currently attempting to figure out what's causing the problem,
by trying different kernels with different options. Trying a different
version of GCC is a good idea. Didn't think of that yet.
So far, I had limited success. The panics popped up in all the kernels
I tested with, although some things, like some other hardware / drivers, seem
to make them more likely to appear. See the other thread I started about
this problem.
Grtz,
Oliver
On Tue, Jun 24, 2003 at 19:10:08 -0500, Alex Zarubin wrote:
Mark Oliver,
It is too early to say, but the picture is different now. Our dual CPU,
dual T400P box is up for 4 days, under the load of 10 - 100 simultaneous
PRI - SIP calls. We installed 2.4.21 #2 SMP (it was still freezing after
that) and, what I think made the difference, recompiled
zaptel-libpri-asterisk
with gcc 3.3.
The problem, on the way, was that asterisk wouldn't start after that. It was
crashing while loading mp3 and lpc10 codecs. We put 'noload' for these two
into modules.conf - temporary solution, of course.
There are problems, still, with multiple connections at the same time.
Windows
to the box get frozen for a sec, D-channel error messages. The following
messages are dumped into /var/log/messages. What do you think?
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: wait_on_irq, CPU 1:
Jun 24 18:23:25 mspgate03 kernel: irq: 1 [ 0 0 1 0 ]
Jun 24 18:23:25 mspgate03 kernel: bh: 0 [ 0 0 0 0 ]
Jun 24 18:23:25 mspgate03 kernel: Stack dumps:
Jun 24 18:23:25 mspgate03 kernel: CPU 0:0200 036f 00e14603
1802 0310 6647 008e0200 4803
Jun 24 18:23:25 mspgate03 kernel:0078 001ffa02 5b490300
0600 01c7 074e0308 1afe 01c74d03
Jun 24 18:23:25 mspgate03 kernel:2302 d708 e101
0900 01d7 f5030001 0423 09300207
Jun 24 18:23:25 mspgate03 kernel: Call Trace:[f89bd281]
[f89bb132] [f89bbb47] [f89bd281] [f89bd281]
Jun 24 18:23:25 mspgate03 kernel: [f89bb132] [f89bd281]
[f89bd281] [f89bb132] [f89bbb47] [f89e7737]
Jun 24 18:23:25 mspgate03 kernel: [f89aa80a] [f89aa80a]
[c01feee4] [f89e7737] [c01f4eae] [c010a98e]
Jun 24 18:23:25 mspgate03 kernel: [c020d122] [c010abe3]
[c020d122] [c020d550] [c010a98e] [c020d550]
Jun 24 18:23:25 mspgate03 kernel: [c010abfe] [c01f0919]
[c01f0919] [c022a1ef] [c022a1ef] [c022a5f5]
Jun 24 18:23:25 mspgate03 kernel: [f89bd281] [f89bd281]
[f89bd281] [f89bb132] [f89bd510] [f89e7737]
Jun 24 18:23:25 mspgate03 kernel: [c022a5f5] [c01f0ffd]
[c01f112e] [c01f53c2] [c012005b] [c010abfe]
Jun 24 18:23:25 mspgate03 kernel: [c015147a] [c01509dc]
[c0147460
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
Hi Alex,
The problem is most likely to occur with high volumes of call-setups and
disconnects. This could be reproduced by putting 2 of your T-1 ports
back to back and then using the auto-dialer to generate a large amount of
very short calls between the ports.
I'm currently attempting to figure out what's causing the problem,
by trying different kernels with different options. Trying a different
version of GCC is a good idea. Didn't think of that yet.
So far, I had limited success. The panics popped up in all the kernels
I tested with, although some things, like some other hardware / drivers, seem
to make them more likely to appear. See the other thread I started about
this problem.
Grtz,
Oliver
On Tue, Jun 24, 2003 at 19:10:08 -0500, Alex Zarubin wrote:
Mark Oliver,
It is too early to say, but the picture is different now. Our dual CPU,
dual T400P box is up for 4 days, under the load of 10 - 100 simultaneous
PRI - SIP calls. We installed 2.4.21 #2 SMP (it was still freezing after
that) and, what I think made the difference, recompiled
zaptel-libpri-asterisk
with gcc 3.3.
The problem, on the way, was that asterisk wouldn't start after that. It was
crashing while loading mp3 and lpc10 codecs. We put 'noload' for these two
into modules.conf - temporary solution, of course.
There are problems, still, with multiple connections at the same time.
Windows
to the box get frozen for a sec, D-channel error messages. The following
messages are dumped into /var/log/messages. What do you think?
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: wait_on_irq, CPU 1:
Jun 24 18:23:25 mspgate03 kernel: irq: 1 [ 0 0 1 0 ]
Jun 24 18:23:25 mspgate03 kernel: bh: 0 [ 0 0 0 0 ]
Jun 24 18:23:25 mspgate03 kernel: Stack dumps:
Jun 24 18:23:25 mspgate03 kernel: CPU 0:0200 036f 00e14603
1802 0310 6647 008e0200 4803
Jun 24 18:23:25 mspgate03 kernel:0078 001ffa02 5b490300
0600 01c7 074e0308 1afe 01c74d03
Jun 24 18:23:25 mspgate03 kernel:2302 d708 e101
0900 01d7 f5030001 0423 09300207
Jun 24 18:23:25 mspgate03 kernel: Call Trace:[f89bd281]
[f89bb132] [f89bbb47] [f89bd281] [f89bd281]
Jun 24 18:23:25 mspgate03 kernel: [f89bb132] [f89bd281]
[f89bd281] [f89bb132] [f89bbb47] [f89e7737]
Jun 24 18:23:25 mspgate03 kernel: [f89aa80a] [f89aa80a]
[c01feee4] [f89e7737] [c01f4eae] [c010a98e]
Jun 24 18:23:25 mspgate03 kernel: [c020d122] [c010abe3]
[c020d122] [c020d550] [c010a98e] [c020d550]
Jun 24 18:23:25 mspgate03 kernel: [c010abfe] [c01f0919]
[c01f0919] [c022a1ef] [c022a1ef] [c022a5f5]
Jun 24 18:23:25 mspgate03 kernel: [f89bd281] [f89bd281]
[f89bd281] [f89bb132] [f89bd510] [f89e7737]
Jun 24 18:23:25 mspgate03 kernel: [c022a5f5] [c01f0ffd]
[c01f112e] [c01f53c2] [c012005b] [c010abfe]
Jun 24 18:23:25 mspgate03 kernel: [c015147a] [c01509dc]
[c0147460] [c0147fb8] [f89e7737] [f89e7737]
Jun 24 18:23:25 mspgate03 kernel: [c01f0998] [c01f0fac]
[c01f112e] [c01f53c2] [c0117fce] [c0117ef0]
Jun 24 18:23:25 mspgate03 kernel: [c0144a64] [c01246db]
[c0109023]
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 2:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 3:0070 cce30002 0cd8
08fa 6953 656c706d 6c616e41 73697379
Jun 24 18:23:25 mspgate03 kernel:0009a700 46534c00 65746e69
6c6f7072 32657461 6e655f61 0a810063 6953
Jun 24 18:23:25 mspgate03 kernel:656c706d 65746e49 6c6f7072
4c657461 39004653 530b 6c706d69 66736c65
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 1:e14d5eac c025c896 0001
0001 0001 c010a7c2 c025c8ab
Jun 24 18:23:25 mspgate03 kernel: f2d92124 e14d5f00
c0191104 0500 1805 00bf 8a01
Jun 24 18:23:25 mspgate03 kernel:7f1c0300 01000415 1a131100
170f1200 e14d4000
Jun 24 18:23:25 mspgate03 kernel: Call Trace:[c010a7c2]
[c0191104] [c01913d4] [c018e1e2] [c014c2c7]
Jun 24 18:23:25 mspgate03 kernel: [c0109023]
Jun 24 18:23:25 mspgate03 kernel:
Thank you.
Alex Zarubin
-Original Message-
From: The Traveller [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 17, 2003 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues
On Tue, Jun 17, 2003 at 20:54:39 +0200, The Traveller wrote:
BTW: As I reported in my previous mail to the list, I've now installed
kernel
2.4.21-rc2 with ACPI-patch on the box
[Asterisk-Users] Important: PSTN access-number for Dutch gateway changed
Yo all, The PSTN access-number for the Dutch IAXTel - PSTN-gateway has changed. The new number is: +31 20 3987 567. Calling from IAXTel to Dutch toll-free PSTN-numbers is still done in the same way, by calling 31800rest of number. Mark: Could you please update your web-sites to reflect this change? The old number is mentioned on http://www.gnophone.com/;, not sure about other places. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: PSTN access-number for Dutch gateway changed
Hey Andy, Hm, I see. That's a bit odd, as in the sample extensions.conf from a current CVS-version, the pattern to match IAXTel-numbers is 1700NXX, where the N matches digits 2-9 and X matches digits 0-9. This is what I'm using at the gateway as well. Maybe Mark forgot to fix this in the example-config before starting to hand out those numbers. Can you mail me your full IAXTel-number, so I can verify this? I changed the N into an X for now, so the gateway should now work for 17001-numbers. Grtz, Oliver On Thu, Jun 26, 2003 at 22:50:08 +0200, Andy Powell wrote: Oliver, can you clarify how the gateways is supposed to be used, I've tried calling the number from a PSTN line, the call is answered and i get dialtone, I then try to dial my iaxtel number and just get told that it's an invalid extension.. the 'error' occurs after dialing 17001 of my iaxtel number... Thanks Andy *** REPLY SEPARATOR *** On 26/06/2003 at 21:55 The Traveller wrote: Yo all, The PSTN access-number for the Dutch IAXTel - PSTN-gateway has changed. The new number is: +31 20 3987 567. Calling from IAXTel to Dutch toll-free PSTN-numbers is still done in the same way, by calling 31800rest of number. Mark: Could you please update your web-sites to reflect this change? The old number is mentioned on http://www.gnophone.com/;, not sure about other places. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible solution to Zaptel panics
Heya Mark (and others), Here's an update on my adventures while trying to debug the Zaptel-related panics, as discussed on this list a while back. While debugging the problem, I completely swapped the machine for an entirely different model (Supermicro dual Xeon 2.4GHz with 2Gb of RAM), put the cards in, recompiled * + Zaptel and administered my stress-test. The machine stayed up in excess of 15 minutes, so I assumed a hardware or timing-problem had caused the panics on the old box and continued installing the other components (Kernel v2.4.21, with Eicon Diva Server- drivers, for the Diva Server 4BRI PCI v2.0 in that box, amongst others). After I had it up and running like the old server, I gave the stress-test another go, just to be sure, and sure enough, after just 1 or 2 minutes, it croaked again, with the familiar panic. I was quick to diagnose that the most probable difference that could cause this was that the Eicon Diva-drivers where now loaded. And indeed, after I unloaded them, started * without chan_capi and administered the stress-test again, it didn't panic. In fact, it has been running under the load for around 20 minutes now. I just loaded the Eicon-drivers and it crashed again, in under a minute. So, it seems like the Eicon Diva Server 4BRI PCI and Zaptel (at least with the E100P-hardware that I'm using) don't like eachother very much. I checked /proc/interrupts and these devices aren't sharing any IRQ's: CPU0 CPU1 CPU2 CPU3 0: 151846 0 0 0IO-APIC-edge timer 1: 3 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 4: 46 0 0 0IO-APIC-edge serial 8: 1 0 0 0IO-APIC-edge rtc 9: 0 0 0 0IO-APIC-edge acpi 15: 2 0 0 0IO-APIC-edge ide1 16: 0 0 0 0 IO-APIC-level usb-uhci 18: 0 0 0 0 IO-APIC-level usb-uhci 19: 0 0 0 0 IO-APIC-level usb-uhci 24:1433575 0 0 0 IO-APIC-level t1xxp 28: 7387 0 0 0 IO-APIC-level aic7xxx 29: 15 0 0 0 IO-APIC-level aic7xxx 31: 4810 0 0 0 IO-APIC-level eth0 48: 65 0 0 0 IO-APIC-level DIVA 4BRI 6261 NMI: 0 0 0 0 LOC: 151337 151329 151330 151353 ERR: 0 MIS: 0 I'm using a stock v2.4.21-kernel, with the Eicon-drivers from http://www.melware.de/; with the RH8.0 Diva Server software from http://www.eicon.com/; (a bit tweaked to use the aforementioned drivers, instead of the ones shipped with it). The kernel currently also has FreeS/WAN v2.0 in it, but it doesn't seem to be related to my problems and it's module wasn't loaded during any of my tests. The box is running an up2date Redhat 9. The Eicon card and E100P are currently the only cards in the system. I'm going to try the ACPI-patch (http://sourceforge.net/projects/acpi/) again and see if that changes anything and play a bit with the BIOS, but it might be a good idea for someone with more knowledge of the internals of kernel and drivers to have a good look at this problem. I'm willing to assist in producing the right debugging-info, as I can reliably reproduce the problem. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i4l - summary of patches?
Yo Iain, On Tue, Jun 17, 2003 at 21:48:34 +0100, Iain McWilliams wrote: Hi, I'm trying to get asterisk running on kernel 2.4.20 however trawling through the archives I've found a few references to patches to remove i4l's dtmf detection, but have been unable to find the patch itself (I think it is isdn_audio.c). Can anyone point me in the right direction? The problem I'm seeing is connecting a SIP softphone (tried a few) to an external number via an Hisax type 35 isdn card causes 1-2 sec silences in the audio on the SIP phone. Any pointers? I recently did this myself. Here's my patch. Works for me. Your mileage may vary. :-) Grtz, Oliver Index: channels/chan_modem_i4l.c === RCS file: /usr/cvsroot/asterisk/channels/chan_modem_i4l.c,v retrieving revision 1.2 diff -u -r1.2 chan_modem_i4l.c --- channels/chan_modem_i4l.c 27 Apr 2003 21:34:27 - 1.2 +++ channels/chan_modem_i4l.c 17 Jun 2003 21:52:04 - @@ -265,7 +265,7 @@ if (option_debug) ast_log(LOG_DEBUG, Ignoring Escaped character '%c' (%d)\n, esc, esc); return p-fr; - case '0': + /* case '0': case '1': case '2': case '3': @@ -280,7 +280,7 @@ ast_log(LOG_DEBUG, DTMF: '%c' (%d)\n, esc, esc); p-fr.frametype=AST_FRAME_DTMF; p-fr.subclass=esc; - return p-fr; + return p-fr; */ case 0: /* Pseudo signal */ return p-fr; default: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New busydetect routines for analog channels (FXO mostly)
Yo Martin, On Tue, Jun 17, 2003 at 17:03:15 -0500, Martin Pycko wrote: Hello, I've commited the new busydetect routine to CVS. You need to cvs update asterisk of course and then choose it in asterisk/Makefile and recompile asterisk. [...] It fails to compile here (Redhat 9, gcc version 3.2.2 20030222 (Red Hat Linux 3.2.2-5)): dsp.c:1055:15: missing terminating ' character make: *** [dsp.o] Error 1 Seems to be something simple, like the need to escape the '. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
Yo, I've seen very similar Zaptel-related freezes on a wide variety of mainboards (SMP as well as non-SMP), with X100P's as well as with an E100P. At some point, almost always at the moment a call through one of those cards connects or disconnects, the machine completely stops responding and needs a reset to come back to life. A very nice way to trigger it with the E100P seems to be to put around 10-20 channels of it into a meetme-conference and then issue the stop now-command on the Asterisk-console. A high volume of connects / disconnects seems to trigger the freezes. I'm still investigating the issue and am going to try different kernels and some custom kernel-patches. One of my boxes (dual PIII-750, Intel L440GX+-board) with an X100P and a TDM40P in it hasn't frozen since I installed kernel 2.4.21-rc2 with the ACPI-patch (http://sourceforge.net/projects/acpi/). I'll probably try that on the box with the E100P first. Be sure enable Power Management support in your kernel-config, disable APM, enable ACPI and check all ACPI-options, except for CPU Enumeration Only. Note that this ACPI- patch also handles IRQ-routing and might help in cases where the BIOS assigns the same IRQ to some devices (or, as was the case for me, none at all). Grtz, Oliver On Mon, Jun 16, 2003 at 13:03:20 -0500, Alex Zarubin wrote: Mark, As far as pings - we have cases when we could ping the box on both interfaces and there are cases when we could not (we tried 3-4 sets of NICs and drivers). All telnets, X, ssh etc. are definitely dead. No coredumps (asterisk was started with -g option), no kernel panics. Black console, Alt-SysRq combinations don't work. Pretty much no options but rebooting the box. As far as SMP and single T400P - we'll try and report the results but the idea was to go with as high density as possible ... What do you think of using hyperthreading - should we enable or disable it for the box running asterisk? What about -DCONFIG_ZAPTEL_WATCHDOG ? Can it help and how to use it? Thank you. Alex Zarubin -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Saturday, June 14, 2003 10:23 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues When you say stops responding do you mean no more pings, telnet dead, etc? Or do you mean asterisk stops responding? Is there a segfault or kernel panic, or any other failure diagnostic? Mark On Thu, 12 Jun 2003, Alex Zarubin wrote: Zaptel was compiled with -D__SMP__ We've installed irqbalance and the picture improved a lot (thanks to Jared Smith). Do you still see problems in our /proc/interrupts? The big issue for us now is that after 24+ hours of the test load PRI-SIP our Dell PE2650, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, 2.4.20-18.7smp #1 SMP stops responding to anything. So the questions are: - are there known issues with PE2650 and ways to fix them? - can someone recommend the 'stable' 2.4 SMP kernel for this kind of load? - any expertise in this area will be appreciated CPU0 CPU1 CPU2 CPU3 0: 230710 30030 50050 0IO-APIC-edge timer 1: 5 0 0233IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 14: 27 0 2 0IO-APIC-edge ide0 20:2085442 400221 0 230232 IO-APIC-level tor2 24: 2938481841658 10010 570568 IO-APIC-level tor2 28: 5 25643 0 0 IO-APIC-level eth0 29: 5 05165040 0 IO-APIC-level eth1 30: 43720 35467 1291 3296 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 310618 310616 310616 310616 ERR: 0 MIS: 0 Thank you. Alex Zarubin -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2003 9:48 AM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088IO-APIC-edge timer 1: 1 0 1 1IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 15: 1 0 0 1IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0
