Re: [Asterisk-Users] Differ between "private" and "out of area"?
usecallerid=yes hidecallerid=no callerid=asreceived usecallingpres=yes callwaiting=no callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=10 context=pstn rxgain=8.15 txgain=2.0 signalling=fxs_ks channel => 1 - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> > Paste the section from zapata.conf that handles the x101p. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP
Which firmware do you need? I have couple of them, but I don't know is it legal to share them over Internet :-)) - Original Message - From: Stern, Craig To: asterisk-users@lists.digium.com I have been looking for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any help in locating would be much appreciated. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Differ between "private" and "out of area"?
I have CID identificator connected in parallel with X101P/Asterisk, and it displays "Private" for hidden calls, and "Out of area" for calls from rural areas (with old phone systems). But Asterisk always deliver CALLERIDNUM="", CALLERIDNAME="" in both cases. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Differ between "private" and "out of area"?
Yes, I know that, but, how to distinguish between them at incoming call?
- Original Message -
From: "Matt" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: pon 19. sep 2005 16:22
Subject: Re: [Asterisk-Users] Differ between "private" and "out of
area"?
A private call is a call that someone has specifically blocked. An
"out of area" or "unknown" call is simply a call that the caller-id
did not come through on correctly, for some reason.
On 9/18/05, Goran Dj. <[EMAIL PROTECTED]> wrote:
> Is there any method to make difference between Hidden ("Private") and
> unknown ("Out of area") incoming calls on ZAP/x101p? I want to block
any
> hidden call, and to allow unknow calls, but ZAP channel (X101P) always
> delivering empty CALLERID=""<> in both cases.
>
>
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Re: [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
No, no, it's more SIMPLE than that. Try this:
[incoming]
exten => s,1,setcallerid("NAME")
exten => s,2,dial()
That's all (after couple of hours of investigation).
If call origin from SIP, i see NUMBER on my phone, if call origin from
PSTN, i see NAME on my phone.
- Original Message -
From: "Shaun Ewing" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: pon 19. sep 2005 2:04
Subject: Re: [Asterisk-Users] sometimes CIDNUM shows, sometimes
CIDNAME??? Why,why, why, why?
On 9/19/05, Goran Dj. <[EMAIL PROTECTED]> wrote:
> Why Asterisk showing (on SCCP and H323 phones) different CID related
to
> type of Incoming channel:
> If incoming channel is SIP, on phone is displayed CALLERIDNUM
> If incoming channel is ZAP, on phone is displayes CALLERIDNAME
>
> It vas very frustrating! I lost couple hours of my time to find that
my
> dialplan is not faulty, but asterisk is!
Have you considered the possibility that your SIP provider may not be
sending you the caller id name?
CNAM looksup do cost money, and it's probably the exception rather
than the norm to find a VoIP provider that will deliver it.
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[Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Differ between "private" and "out of area"?
Is there any method to make difference between Hidden ("Private") and
unknown ("Out of area") incoming calls on ZAP/x101p? I want to block any
hidden call, and to allow unknow calls, but ZAP channel (X101P) always
delivering empty CALLERID=""<> in both cases.
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[Asterisk-Users] X101P ringing too long !
Pause between successive incoming rings on my phone line is 4 sec, so when x101p do not receive next ring signal after 4.5 sec, call should be consider as ended. But, if caller hang-up, call is ended (Hungup 'Zap/1-1) exactly 8 sec after last ring, and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change in zapata.h some lines: #define ZT_DEFAULT_RINGTIME 500 #define ZT_LOOPCODE_TIME 3000 #define ZT_RINGOFFTIME 2000 but with no effects. Call is still ended 8 sec after last ring. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to shorten ringing stop detection onX101Pclone?
This working only when zap answer call. But, if zap don't answer (ringing), and (outside) caller hangup, then there is no busy tone. By the way, do you know some voip provider in Paris with Direct Inward Dial numbers? Where can I found best information about prices of France Telecom (PRI od BRI ISDN/RNIS, tarrifs, etc...) - Original Message - From: <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: cet 1. sep 2005 7:13 Subject: RE : [Asterisk-Users] How to shorten ringing stop detection onX101Pclone? Hello Goran, Modify your /etc/asterisk/zapata.conf like this : busydetect=yes busycount=3 And, of course, you must have chosen your correct country for ringing mode in your /etc/zaptel.conf file : loadzone=fr defaultzone=fr I am in France :-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Goran Dj. Envoyé : jeudi 1 septembre 2005 02:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] How to shorten ringing stop detection on X101Pclone? When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to speed-up INCOMING-RINGING-ENDED detection on X101P/zapata?
> Pause betwen incoming rings on my phone line is 4s, so when x101p clone > (wcfxo driver) do not receive next ring signal after 4.5 sec, call > should be consider as ended. > > What should I change to set that time (4.5 sec) for incoming ring end > detection? I measured, event "-- Hungup 'Zap/1-1'" is shown exactly 8 sec after last detected ring (on X101P), and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change in zapata.h some lines: #define ZT_DEFAULT_RINGTIME 500 #define ZT_LOOPCODE_TIME 3000 #define ZT_RINGOFFTIME 2000 but with no effects. "Hungup" is still shown 8 sec after last ring. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why ZAPATA inserting pause before last digit, during dialing? GRRRRR....
> I want to speed-up dialing on X101P clone (Ambient modem). I probably > must change wcfxo.c, but what line to change? I found what to change: digits.h line 23 from #define DEFAULT_DTMF_LENGTH 100 * 8 to #define DEFAULT_DTMF_LENGTH 50 * 8 and my dialling is now much faster. But, I have new question: Before last digit, there is always inserted pause (500ms) or maybe two (1000ms). I don't use pause anywhere in my dial-plan, so, why is inserted and dialled? To test is that really a pause or something else, I changed line 25 from #define PAUSE_LENGTH 500 * 8 to #define PAUSE_LENGTH 2000 * 8 and, guess what, now I have 4sec pause before last digit is played. How to get rid of it? I can maybe "#define PAUSE_LENGTH 0 * 8" but that this is very dirty solution. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to shorten ringing stop detection on X101P clone?
When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to speed-up dialnig with X101P clone modem?
I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (AT&W) to be permanent) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect Dialtone
Dialtone detection should be an option in .conf for zap channel, i agree with that. > Are you trying to play with the case where you have an analog phone > bridged on your fxo line, and detect the lack of dialtone when > someone is using that analog phone? Belive or not, but at some places on the world are still in use some old (non-digital) ATC-es which do now provide dial-tone instantly. For example, when ATC ARF-102 is very congested with outgoing calls, you must wait some (unknown) time to get dialtone (10sec, 1min, 5min...) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi, modprobe: Can't locate module capifs, ERROR: fopen(/etc/capi.conf, r)
> But, now I cannot start chan_capi.so: > WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI > disabled! > > from tty: > capiinit > ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or > directory (2) > > capiinfo > capi not installed - No such file or directory (2) > > capiinit show > ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) > ERROR: no cards configured in /etc/capi.conf I resolved missing /dev/capi20 with shell script "makedev-capi.sh" but, now, when starting capiinit: modprobe: Can't locate module capifs modprobe: Can't locate module capifs WARNING: filesystem capifs not available ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi, cannot open /dev/capi20, no cards configured in /etc/capi.conf
> wget > ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-08-21.tar.bz2 > > tar xvjf isdn4k-utils-CVS-2005-08-21.tar.bz2 > cd isdn4k* > cd capi20 > ./configure > make > make install > > that's all > > Sergio Ok. Thanks. It's working, and I compiled successfully chan_capi-0.5.3 (because 0.5.4 producing some error). But, now I cannot start chan_capi.so: WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI disabled! from tty: capiinit ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or directory (2) capiinfo capi not installed - No such file or directory (2) capiinit show ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) ERROR: no cards configured in /etc/capi.conf So, whats happening? What is responsible for making /dev/capi20, and how to make /etc/capi.conf? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?
> > capi6208 0 > > kernelcapi 30496 1 [capi] > > capiutil 22272 0 [kernelcapi] > > uhci 2 0 (unused) > > usbcore59308 1 [uhci] > > hisax 448240 0 (unused) > > isdn 116684 0 [hisax] > > slhc4976 0 [isdn] > > wcfxo 8384 2 > > zaptel176992 8 [wcfxo] > > ide-scsi9328 0 > > ne 6672 1 > > 83906000 0 [ne] > > crc32 2880 0 [8390] > > isa-pnp30736 0 [hisax ne] > > > > So, where is a problem? Should I compile kernel with capi as a part of a > > kernel, not as a module? How to do that? > > It's okay to use it as modules. But the cards supported by HiSax do not > provide CAPI interface. I don't know the status of mISDN, but that would > be the driver supporting CAPI. Hmmm? I don't know what hisax doing here (and even what is that). My ISDN card (winbond w6692cf chip) isn't in computer, I will put it there when I successfully complile chan_capi. What modules do I need? Only capi(&kernelcapi&caputil) and chan_capi? > > You don't have libcapi20 (or the development package of it) installed. Yes, but I dont have it on my Slackware10 CD'es. I don't have libcapi, or isdn4... or anything with isdn or capi in their name. Where to find libcapi20 (od devel...) for slackware? Goran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?
I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch of errors. (By the way, can I use chan_capi for ISDN card with winbond w6692cf chipset?) I'm not a linux expert, still :-) Before compiling, when I type "modprobe capi" to load capi module, and then "lsmod", i get list of modules: capi6208 0 kernelcapi 30496 1 [capi] capiutil 22272 0 [kernelcapi] uhci 2 0 (unused) usbcore59308 1 [uhci] hisax 448240 0 (unused) isdn 116684 0 [hisax] slhc4976 0 [isdn] wcfxo 8384 2 zaptel176992 8 [wcfxo] ide-scsi9328 0 ne 6672 1 83906000 0 [ne] crc32 2880 0 [8390] isa-pnp30736 0 [hisax ne] So, where is a problem? Should I compile kernel with capi as a part of a kernel, not as a module? How to do that? Errors when I try to compile chan_capi: [EMAIL PROTECTED]:#make ./create_config.sh "/usr/include" Checking Asterisk version... * no 'struct ast_channel_tech', using old pvt * ast_dsp_process() without 'needlock' * no 'struct ast_callerid' * found 'struct timeval delivery' * no 'transfercapability' * no 'ast_config_load' config.h complete. gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/i nclude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DASTERISKVERSION=\"\ " -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:49:20: capi20.h: No such file or directory In file included from chan_capi.c:52: chan_capi_app.h:28: error: parse error before "get_ast_capi_MessageNumber" chan_capi_app.h:28: warning: type defaults to `int' in declaration of `get_ast_capi_MessageNumber' chan_capi_app.h:28: warning: data definition has no type or storage class chan_capi_app.h:34: error: parse error before "_capi_put_cmsg" chan_capi_app.h:34: error: parse error before '*' token chan_capi_app.h:34: warning: type defaults to `int' in declaration of `_capi_put_cmsg' chan_capi_app.h:34: warning: data definition has no type or storage class In file included from chan_capi.c:53: chan_capi_pvt.h:133: error: parse error before "_cword" . . . chan_capi.c:2834: error: invalid lvalue in assignment chan_capi.c:2835: error: invalid lvalue in assignment chan_capi.c:2837: error: invalid lvalue in assignment chan_capi.c:2844: error: `error' undeclared (first use in this function) chan_capi.c:2845: error: `CMSG2' undeclared (first use in this function) chan_capi.c:2847: warning: implicit declaration of function `IS_FACILITY_CONF' chan_capi.c:2875: error: subscripted value is neither array nor pointer chan_capi.c:2877: error: subscripted value is neither array nor pointer chan_capi.c:2882: error: subscripted value is neither array nor pointer chan_capi.c:2886: error: subscripted value is neither array nor pointer chan_capi.c:2890: error: subscripted value is neither array nor pointer chan_capi.c:2894: error: subscripted value is neither array nor pointer chan_capi.c:2898: error: subscripted value is neither array nor pointer chan_capi.c:2902: error: subscripted value is neither array nor pointer chan_capi.c:2906: error: subscripted value is neither array nor pointer chan_capi.c:2910: error: subscripted value is neither array nor pointer chan_capi.c:2914: error: subscripted value is neither array nor pointer chan_capi.c:2918: error: subscripted value is neither array nor pointer chan_capi.c:2922: error: subscripted value is neither array nor pointer chan_capi.c: In function `load_module': chan_capi.c:3088: warning: implicit declaration of function `capi20_isinstalled' chan_capi.c:3094: warning: implicit declaration of function `capi20_register' chan_capi.c:3104: warning: implicit declaration of function `capi20_get_profile' chan_capi.c: In function `unload_module': chan_capi.c:3301: warning: implicit declaration of function `capi20_release' make: *** [chan_capi.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ast.1.0.9 (only) strange problem with IAX and DDNS
Asterisk 1.0.9: IAX2 registration timeout! --- I have 2 locations with ADSL lines, both with dynamic IP (+ dynamic DNS). On location 1 => Asterisk 1.0.RC2 / Slackware 10 On location 2 => Asterisk 1.0.9 / Slackware 10 They are on private network and connected via IAX2 through NAT(win2000server), and registering to DDNS name of each other. I know that Asterisk is not very smart on handling DNS, so when remote ADSL change IP address, I must reload IAX on local Asterisk to (resolve new address and) continue registering itself to remote Asterisk. But, here start problem: Asterisk 1.0.9 (location 2), sometimes (very often) when LOCAL ip address is changed, can't anymore register himself to remote Asterisk 1.0.RC2 which by the way DIDN'T change it's IP address! Remote Asterisk (unchanged IP) also can't register himself to local Asterisk (changed IP) even when I do "reload" of IAX (on remote Asterisk). That problem cannot be resolved with unload/load IAX2, or stop/start Asterisk. Only reboot of local Slackware (location 2, unchanged IP, Asterisk 1.0.9) helping, and after reboot everything working well (till some of next IP address changing). There things gets interesting: Asterisk 1.0.RC2 (location 1) didn't had that problem. Then, 2 day ago, I upgraded to 1.0.9, and now I have same problem on BOTH location! Registration to other networks (FWD for example) working with no problems, only registration to each-other is impossible. --- here is configuration: LOCATION 1: [general] register => L1o:[EMAIL PROTECTED] [L2o] type=peer username=L1i auth=rsa outkey=L1 host=dynamic qualify=yes canreinvite=yes disallow=all allow=ilbc trunk=no [L2i] type=user username=L2i auth=rsa inkeys=L2 qualify=yes context=incoming canreinvite=yes disallow=gsm trunk=no LOCATION 2: [general] register => L2o:[EMAIL PROTECTED] [L1o] type=peer host=dynamic username=L2i auth=rsa outkey=L2 qualify=yes canreinvite=yes trunk=no [L1i] username=L1i type=user auth=rsa inkeys=L1 context=incoming qualify=yes canreinvite=yes trunk=no ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autostart Asterisk (crashing)!
I did, but asterisk won't start when user is not loged in !? rc.local: if [ -x /usr/sbin/asterisk ]; then /usr/sbin/asterisk echo "ASTERISK started" fi I get echo "ASTERISK started" when turn on computer, but asterisk is NOT started. When I login as root and type "ps -e" i get list: PID TTY TIME CMD 1 ?00:00:04 init 2 ?00:00:00 keventd 3 ?00:00:00 ksoftirqd_CPU0 4 ?00:00:00 kswapd 5 ?00:00:00 bdflush 6 ?00:00:00 kupdated 10 ?00:00:00 mdrecoveryd 58 ?00:00:00 syslogd 61 ?00:00:00 klogd 169 ?00:00:00 khubd 521 ?00:00:00 inetd 524 ?00:00:01 sshd 535 ?00:00:00 crond 537 ?00:00:00 atd 540 ?00:00:00 sendmail 543 ?00:00:00 sendmail 553 ?00:00:00 smbd 555 ?00:00:00 nmbd 557 ttyS000:00:00 gpm 564 tty1 00:00:00 agetty 565 tty2 00:00:00 agetty 566 tty3 00:00:00 agetty 567 tty4 00:00:00 agetty 568 tty5 00:00:00 agetty 571 tty6 00:00:00 agetty 576 ?00:00:06 mpg123 579 ?00:00:08 mpg123 583 ?00:00:08 mpg123 587 ?00:00:08 mpg123 622 ?00:00:00 smbd 624 ?00:00:00 sshd 626 pts/000:00:00 bash 639 pts/000:00:00 ps Interesting here is that mpg123 is started from Asterisk, but Asterisk isn't on this list. Seems to me that Asterisk crashed during starting. If I execute /etc/rc.d/rc.local from my root console, Asterisk starting normaly. Why crashing? - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: uto 15. feb 2005 17:01 Subject: Re: [Asterisk-Users] Autostart Asterisk on Slackware? > On February 15, 2005 10:49 am, Goran Dj. wrote: > > How to autostart Asterisk (daemon) on Slackware 10? I know that I sh ould > > put something in /etc/rc.d, but what? > > Something like > > /usr/sbin/asterisk -g > > in /etc/rc.d/rc.local would do it. You can craft up more complex things if > you like, wrap safe_asterisk or do whatver, but that'll get you started. > > -A. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autostart Asterisk on Slackware?
Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone
IP phone party can hear ISDN party, but ISDN (incoming) party canNOT
hear IP phone party
because RTP stream from Asterisk is sent to 127.0.0.1 instead to real IP
address of AS5350
Here is H.323 debug, for both situations:
1) ---
--- outgoing call (RTP is ok, both party can hear) --
-- Call token is ip$localhost/12862
-- Call reference is 12862
-- Sending SETUP message
Recieved Open Recieve Channel Ack
=*= In CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
=-= In OnAlerting for call 12862: sessionId=1
--- found logical channel. Connecting RTP
RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16862
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 14152
-- Ringing phone for "10.10.10.61"
-- Asked to indicate 'Remote end is ringing' condition on channel
Skinny/[EMAIL PROTECTED]
RFC3389: 1 bytes, level 4...
Dec 22 18:51:26 NOTICE[557082]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
=*= In CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
=-= In OnConnectionEstablished for call 12862
-- Connection Established with "10.10.10.61"
-- Asked to indicate 'Stop tone' condition on channel
Skinny/[EMAIL PROTECTED]
=-= In OnReceivedAckPDU for call 12862
channelsOpen = 1
2) ---
---incoming call (RTP misplaced, incoming party don't hear)
Sending alerting
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16700
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
Recieved Open Recieve Channel Ack
answering call
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
RTP channel id 1 parameters:
-- remoteIpAddress: 127.0.0.1
-- remotePort: 2070
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
=-= In OnConnectionEstablished for call 5006
-- Connection Established with "10.10.10.61"
-- Received Facility message...
=-= In OnReceivedAckPDU for call 5006
-- Received Facility message...
channelsOpen = 1
___
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Re: [Asterisk-Users] sccp cisco 12sp HELP !!!
You don't need that file. My 12sp+ working even without tftp. If you using skinny, then just type right bootimage and mac address in skinnny.conf (press 3 times * on phone). example for skinny.conf: [goran] device=SEP00308062C777 ; to find this number, press 2 times * version=P00203010100;to find this number, press 3 times * context=telefoni callerid="Goran" line => 22 for sccp.conf you don't need version: [SEP00308062C777] type= 12 autologin = goran description = 12sp+ [goran] id = 22 ; Id is a number that is dialed to login to the line with. pin = 1234 ; number needed to log into the device. If missing anyone can log into it. context = telefoni callwaiting = 0 mailbox = 22 ; Check if this mailbox has any mail, and if so, show the MWI. callerid= "Goran", <22> ; CallerId to use on outgoing calls from this line. So, it must work for you if it works for me :-) Goran - Original Message - From: "Jason Price" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: sub 16. okt 2004 3:01 Subject: [Asterisk-Users] sccp cisco 12sp HELP !!! > ok guys, ive been trying to get this to work for 6 hrs now > ive got a cisco 12 sp and i am trying to get it to work with sccp. The > phone boots and is looking for the SEPDefault.cfg or the one below, > BUT i cant find anywere on the net what the content of this file is > im guessing that its the ip of the * box. im riping my hair out > on this one please help... > > > 20:54:47.793156 192.168.1.15.51216 > apollo.tftp: 28 RRQ > "SEP00D0BA848162.cnf" [tos 0x10] > > > > > Jason > ___ > 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 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] How big .CONF files can be?
I'm new to Asterisk. How big can be sip.conf (and other: iax.conf, extensions.conf...) Is there point when I must use DB (MySQL...) instead of pure .conf? ___ 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] chan_sccp.so: _use_ast_pthread_create_instead_
I tried to install chan_sccp (make; make install) but after that when asterisk starting: [chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: __use_ast_pthread_create_instead__ Sep 25 06:34:28 WARNING[16384]: loader.c:423 load_modules: Loading module chan_sccp.so failed! I tried to replace pthread_create() with ast_pthread_create() in chan_sccp.c, but same error... Help? ___ 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
