Re: [Asterisk-Users] Hangup's not detected correctly
Well first of all if you're outside of US or callprogress-supported zones then you can use only busydetect. And that will only work if after the remote hangup your telco gives the fast-busy or any type of busy. You can tweak the duration of tone/pause and increase the count and it *will* work properly. regards Martin On Wed, 7 Jul 2004, Gelson Dias Santos wrote: Steven Critchfield wrote: On Tue, 2004-07-06 at 17:52, Ruben Fagundo wrote: I have an easy question. I setup Asterisk with a TDM400 w/ 4FXO ports and I have the following problem. Yep, so easy it seems to be covered almost weekly here because no one looks up any of the information already provided to them. Not quite easy. I agree its asked about once a week, but they get no solution. Callprogress does not work at all outside US, because it´s just a hack. Busydetect sometimes work, sometimes doesn´t and sometimes drops calls in the middle. I have busydetect=yes and busycount=15 and I still have dropping calls and no hangup detections on a daily basis. I also played with BUSYDETECT_MARTIN and/or BUSYDETECT_TONEONLY and it makes no difference. I also tried editing dsp.c and adjusting BUSY_MIN and BUSY_MAX, but nothing fixes these problems. Gelson A call comes in correctly. The callers dials extension 100 (grandstream SIP phone). The caller then hangup, before the call goes to voice mail, however, the phone continues to ring, then goes to voicemail, and leaves an empty vmail message, long after the caller has hung up. Is there a way I can correct for this, ie, have the system detect hangups correctly ? On analog... callprogress and/or busydetect. Better yet, get disconnect supervision if offered, ___ 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
Re: [Asterisk-Users] Digium cards supporting EM signaling
all T1/E1 boards do regards Martin On Wed, 30 Jun 2004, Gonzalo Mateos wrote: Hi there, I'm quite new to asterisk and digium hardware. I needed to know which of the digium cards supports EM signaling?. thnaks, Gonzalo --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 30-Jun-04 ___ 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] notransfer=yes but still tryin to bridged
notransfer might be still a [global] only keyword for IAX2. regards Martin On Tue, 20 Apr 2004, Hans-Henrik Andresen wrote: Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA ___ 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
Re: [Asterisk-Users] tor2 driver panics with 2 sticks of memory
it looks like some other usb module tries to get loaded and that's what causing it. try to insmod the zaptel tor2 run ztcfg -vv instead. or rmmod all the uhci modules... regards Martin On Fri, 16 Apr 2004, Jim Gottlieb wrote: We use dual Athlon machines with up to three T400P 4-span T1 cards. If I have more than one stick of memory (2 1GB modules or 2 512K modules, each identical), I'm getting a panic soon after I modprobe the tor2 driver. I just loaded the latest from CVS and I'm still getting the panics, which look in part like: Apr 16 14:42:28 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:44 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 000 0 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: Call Trace: [f894d3a0] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [f894d3a0] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [f894ac60] rh_int_timer_do [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: CPU 0:f6a2bea4 c023f901 0001 fff f c010a362 c023f916 Apr 16 14:42:47 test71 kernel: f79ce6a4 f6a2bef8 c017f574 04 00 0005 04bf 8a31 Apr 16 14:42:47 test71 kernel:7f1c0300 01000415 1a131100 170f1200 00 00 f6a2a000 f782d978 f782d978 Apr 16 14:42:47 test71 kernel: Call Trace: [c010a362] __global_cli [kernel] 0x e2 Apr 16 14:42:47 test71 kernel: [c017f574] change_termios [kernel] 0x24 Apr 16 14:42:47 test71 kernel: [c017f844] set_termios [kernel] 0x164 Apr 16 14:42:47 test71 kernel: [c017c6e2] tty_ioctl [kernel] 0x352 Apr 16 14:42:47 test71 kernel: [c0151887] sys_ioctl [kernel] 0x257 Apr 16 14:42:47 test71 kernel: [c0108c5b] system_call [kernel] 0x33 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 last message repeated 2 times Apr 16 14:42:47 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:47 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 42029098 000 [...] Any ideas? Thanks... ___ 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
RE: [Asterisk-Users] question about CPU usage
try to do ps -auxm to list all the threads of the asterisk. Then connect with gdb to the thread that takes 99% of CPU and find out what it's doing. Martin On Mon, 22 Mar 2004, Bill Hamlin wrote: Nope same problem. I just started it and did a couple of ps aux's and got this output: [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.6 1.3 115880 6676 ? R15:43 1:10 asterisk -vgcd root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.3 1.3 115880 6676 ? R15:43 1:13 asterisk -vgcd root 20223 0.0 0.1 3568 624 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.7 1.3 115880 6676 ? R15:43 1:16 asterisk -vgcd root 20225 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.4 1.3 115880 6676 ? R15:43 1:18 asterisk -vgcd root 20227 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.6 1.3 115880 6676 ? R15:43 1:20 asterisk -vgcd root 20229 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Monday, March 22, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage I think Steve is referring to the following line: export LD_ASSUME_KERNEL=2.4.1 If you put this in your command line before starting asterisk, you will get around the RH9 problem of leaving zombies when AGI processes quit. Other than that, I don't think it influences CPU load. Note that the line is not necessary for Fedora Core 1 regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 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 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] Round-robin chan_zap groups...
You can also do R1 to do descending round-robin. Same with G1 and g1. Martin On Wed, 18 Feb 2004, Steve Creel wrote: I've not seen it documented anywhere, but scrolled past it the other day in chan_zap.c. Apparently you can specify a zap group with an 'r' instead of a 'g' to use the group in round-robin. I looked but didn't find anything in the archives on this, so I figured I'd mention it. Steve ___ Steve Creel[EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Pingtel SIPxchange IP PBX goes Open Source...
I wonder if that'll work only with Pingtel phones *smile*. Martin On Wed, 18 Feb 2004, Lenny Tropiano / asterisk.org Mailing list wrote: I just read that Pingtel (www.pingtel.com) will be releasing it's IP PBX (which runs under Linux) to open source (similar model to Redhat Linux, charging for support, etc.). Read more about it at... http://www.pingtel.com/a_opensource.jsp and http://www.tmcnet.com/usubmit/2004/Feb/1024036.htm I love Asterisk, I've migrated my entire company over to it ... maybe we can gleam some technology from this new Open Source Project. I have no idea how SIPxchange ranks up with other IP PBX products. ___ 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
Re: [Asterisk-Users] zaptel compile erro!(asterisk last version0.7.1)
you don't have libm (m for math) library ? Martin On Thu, 15 Jan 2004, [gb2312] Âí÷ë wrote: erro cocent:cc -shared -Wl,-soname,libtonezone.so.1 -lm -o libtonezone.so.1.0 zonedata.lo tonezone.lo /sbin/ldconfig -n . ln -sf libtonezone.so.1 libtonezone.so cc -o ztcfg ztcfg.o -lm -L. -ltonezone ./libtonezone.so: undefined reference to `cos' ./libtonezone.so: undefined reference to `sin' ./libtonezone.so: undefined reference to `pow' collect2: ld returned 1 exit status make: *** [ztcfg] Error 1 ___ 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
RE: [Asterisk-Users] 100% of cpu in an out of the box *
are you running safe_asterisk ? If so try to modify safe_asterisk ... CONSOLE=yes to CONSOLE=no or if not list all the asteirsk threads 'ps -axum | grep asterisk' find the thread that takes the most CPU and connect with gdb gdb /usr/sbin/asterisk pid and do 'bt' and post the last few lines back ... Martin On Thu, 15 Jan 2004, Craig Waddington wrote: Me too :( 100% CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa Sent: 14 January 2004 20:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 100% of cpu in an out of the box * Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa ___ 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 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] max queue time; newbie question
sure, use the 'n' option of the queue and put voicemail app as the next priority Martin On Fri, 9 Jan 2004, Ken Alker wrote: I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ 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
Re: [Asterisk-Users] hangup detection
busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc. I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ 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] ZTMonitor - /dev/dsp problem
try ztmonitor 1 -v Martin On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ 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] E100P connected to Cisco
You need to have HDLC generic support compiled into your kernel ... I think it's not good to have it compiled in modules ... just embedded in kernel. Martin On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig isdn voice-call-failure 0 controller E1 2 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial2:15 no ip address isdn switch-type primary-qsig isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn T310 4 isdn send-alerting no cdp enable voice-port 2:D cptone BR I configured my /etc/zapata.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 Any clue? Thanks in advance, Daniel ___ 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] sip show peers - disappearing
The registry expires after sime time. You can set the default expirey and max in sip.conf. It's up to your phone/sip device to reregister after the registration expires. Martin On Mon, 22 Dec 2003, Jonathan Tew wrote: We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in sip show peers with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ 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] Where is D channel in a PRI link?
It doesn't matter for the zaptel (since you can set dchan=any_channel) but in chan_zap.c in asterisk dchannel for t1 cards is hardcoded to by on 24th channel. You can change that though. regards Martin On Thu, 18 Dec 2003, Michael Welter wrote: We have contracted with Eschelon to provide voice and data over a T1 link. The plan is to terminate this link at a T100P card in the * system. The vendor has said that they will provide the D channel contiguous to the voice channels (voice on channels 1-8 and D channel on 9). The data channels would be 20-24. Will the T100P be able to accept this configuration? Does the PRI specification mandate where the D-channel should be? Thanks for your help. Michael Welter ___ 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] modprobe -r ztd-eth locks up machine...
Did you ifdown the dynamic interfaces first ? Martin On Wed, 17 Dec 2003, Steven Critchfield wrote: On Wed, 2003-12-17 at 10:36, john wrote: Hi, I have just begun working with TDMoE running between 2 fiber nics the dynamic span works great. In my main asterisk box's startup file I just 'modprobe tor2', then start asterisk. The zaptel, ztdynamic ztd-eth modules all load by themselves when tor2 is loaded. If I stop asterisk then 'modprobe -r tor2' the tor2 module is removed but the other three remain. If I then 'modprobe -r ztd-eth' it causes a complete lock up on the machine. The remote machine does not have any zap hardware in it yet and doesn't have these difficulties. I know I can just restart the machine but it is in a production environment (soon to increase from a few to ~30 simultaneous calls) and it is nice to be able to make changes and cvs update installs without restarting. Has anyone experienced this or am I just missing a step or going in the wrong order? Unloading of modules was of such a concern that it almost didn't make it into newer kernels. So you should probably not unload them. A production machine should have specified service windows available. Also decent hardware should be able to reboot fairly fast. The machine I have as our local asterisk machine can go from reset button to accepting new calls in under 50 seconds. Our remote machine is around 90 secs. Depending on y our call volume, and system setup, you should be able to handle this. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] pridump
/dev/zap/1 Martin On Thu, 11 Dec 2003, Paulo Mannheimer wrote: Sorry to bother again, but what is the syntax of a dchannel? I'm trying 1, zap/1, ... without success -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: quarta-feira, 10 de dezembro de 2003 19:10 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] pridump the two dchannels. mark On Wed, 10 Dec 2003, Paulo Mannheimer wrote: Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridump
two d channels of two separate pris Martin On Wed, 10 Dec 2003, Paulo Mannheimer wrote: Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ 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] chan_sip.c update to 1.253
most propably the globalnat is nat= defined in the [general] section. Martin On Wed, 10 Dec 2003, Andrew Thompson wrote: Can someone tell me what this setting is supposed to be? peer-nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ 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] Re-routing of existing calls
check the manager interface ... you can transfer the active call to some other extension. (redirect). If these are zap channels there is zaptransfer command and zapdialoffhook via the manager. regards Martin On Wed, 3 Dec 2003, Alistair Cunningham wrote: Does Asterisk have the capability to re-route calls that have already been connected? By this, I mean: 1. A call comes in to Asterisk. 2. It is routed to an extension as normal. 3. This extension answers, and the conversation starts. 4. After a few minutes, a plugin that I am writing decides that it wants to transfer the call to somewhere else. 5. It signals this to the core of Asterisk (this is the part I am unsure how to do, if it can be done at all). 6. Asterisk hangs up on the extension. 7. (optional) Asterisk plays a 'please hold' message to the caller. 8. The call is routed to the new extension. Is this possible? Can anyone point me to documentation on how to do step 5? -- Alistair Cunningham, Email: [EMAIL PROTECTED] ___ 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] WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
Don't use dtmfmode=inband on GSM codec it'll only work on G711. Martin On Mon, 1 Dec 2003, Bartosz Jozwiak wrote: What does it mean ?? WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel offset between Asterisk and PBX
You might need to edit the code of chan_zap.c You need two things to fix: outgoing calls and incoming calls. Outgoing you should be able to find pri_call call and do chan-1 for chans16. For incoming calls you need to find the handling of PRI_EVENT_RING or something like that and do chan+1 for chans16. regards Martin On Fri, 28 Nov 2003, Roman Sidler wrote: Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to nirwana also no indication is hearable, calling a second internal subcribes bridges them to the first. The PBX sends a SETUP message with channel identification 30 and Asterisk bridges them to Zap-30, instead of Zap-31. The configuration - Digium TE410p card, set for E1 in zaptel.conf span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 in zapata.conf signalling = pri_cpe switchtype = euroisdn context = pri1-in pridialplan = unknown channel = 1-15 channel = 17-31 What's wrong? Thanks in advance Roman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring requested on channel 1 already in use...
Do you have up to date libpri and asterisk ? Also it'd be good if you could send pri debug span 1 (or 2) trace. regards Martin On Tue, 25 Nov 2003, Alastair Maw wrote: I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 2. Hanging up owner. Nov 25 10:49:25 WARNING[98311]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 3. Hanging up owner. Nov 25 10:49:25 WARNING[114696]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 4. Hanging up owner. A little while back I also had this in my logs: Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 11 already in use on span 4 Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 4 already in use on span 1 Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 12 already in use on span 4 Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 5 already in use on span 1 Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 3 already in use on span 1 Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 2 already in use on span 1 Nov 15 17:25:24 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 13 already in use on span 4 FWIW, my libpri/zaptel/asterisk installs are all about two months old. Might whatever causes this have been fixed by now? (I don't want to upgrade otherwise as this problem is quite intermittent). Anyone have any ideas? Alastair ___ 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] Strange code in rtp.c / disconnect - maybe reinvite problems
OK, that was obviously a 'typo' ... It's fixed.
Martin
On Tue, 25 Nov 2003, Detlef Wengorz wrote:
Daniel Chabrol wrote:
Hi List!
I get WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
524300 is not codec1 = 524300, can't do reinvite at my asterisk console.
The code there looks realy strange:
codec0 = pr0-get_codec(c0);
codec1 = pr1-get_codec(c1);
ast_log(LOG_WARNING, codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
/* Hey, we can't do reinvite if both parties speak diffrent codecs */
if (codec0 != codec1)
return -2;
I think the message should only occur *after* checking equality:
if (codec0 != codec1) {
ast_log(LOG_WARNING, codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
return -2;
}
I hoped this can't do reinvite would explain my disconnects from the
nikotel.com sip server after 60 seconds. But this little bug seems only
to be display-specific and not affect funtion. But maybe i oversight
That's correct :-(
but change the code like this
if (codec0 != codec1) {
ast_log(LOG_WARNING,
codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
ast_mutex_unlock(c0-lock); // unlock before return
ast_mutex_unlock(c1-lock); // unlock before return
return -2;
}
and try again.
maybe it helps.
something which still disables the reinvite even if i use
canreinvite=yes in my sip.conf:
[nikotel]
type=friend
username=USERID
fromuser=USERID
secret=PASSWORD
host=calamar0.nikotel.com
canreinvite=yes
context=internal
; no nat entry because im not using nat!
Is there someone which is able to use Nikotel.com with the current
CVS-Version (in my case CVS-11/24/03-19:24:22). BTW: 0.5.0 don't work
too in my case (at least not longer than 60 seconds). Pulver.com calls
and so on are no problem. Any suggestions?
Best regards,
Daniel
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Detlef Wengorz [EMAIL PROTECTED]
Detlef Wengorz [EMAIL PROTECTED]
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Re: [Asterisk-Users] PRI problems
check 'show dialplan nonauthenticated' regards Martin On Fri, 21 Nov 2003, James Sharp wrote: I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the following line in extensions.conf in [nonauthenticated] exten = 8005551212,1,AGI,ivr-main.pl ___ 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] echo cancellation
Did you place echocancel=yes before the definition of the channel with channel keyword in zapata.conf ? regards Martin On Wed, 19 Nov 2003, Elijah Chancey wrote: I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey ___ 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] fax extension isn't executed
Try to use Background application at s,1 Martin On Fri, 14 Nov 2003 [EMAIL PROTECTED] wrote: Hi I'm tring to use asterisk as IVR. But I have trouble when I recieve fax. When I recieve fax, asterisk show message to looks redirect incoming fax to fax extension. But scripts in fax extension never execute. Timeout happen later(I think this timeout is caused by DigitTimeout). Somebody have some suggestion? Asterisk, zapata and zaptel is new CVS(1 week ago) OS is RH8.0 FXO is X100P only (connect PSTN) no FXS asterisk messages is below... (asterisk -vvvdc) -- Set Digit Timeout to 10 -- Set Response Timeout to 20 -- Playing 'vm-extension' (language 'jp') -- Redirecting Zap/1-1 to fax extension -- Timeout on Zap/1-1 -- Playing 'demo-thanks' (language 'jp') -- Hungup 'Zap/1-1' extension.conf is below... [mainmenu] exten = s,1,Wait,2 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,10; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,20 ; Set Response Timeout to 10 seconds exten = s,5,BackGround(vm-extension) ; Play some instructions exten = 1,2,Hangup exten = t,1,Playback(demo-thanks) ; Thanks for trying the demo exten = t,2,Hangup ; Hang them up. exten = fax,1,Goto(faxmenu,100,1) [faxmenu] exten = 100,1,Hangup ;exten = 100,1,RxFax(/var/spool/asterisk/vm/test3.tif) [default] include = mainmenu Yamamoto Tatsuya ___ 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] dtmfmode SIPDtmfMode
Try again ... with latest CVS. Martin On Fri, 14 Nov 2003, Jordi Haarman wrote: Hi, I would like to be able to switch dtmf mode of SIP calls of local clients so the server can understand them and it can also be used when connected to a remote location. I saw that there is an application called SIPDtmfMode in cvs so instead of using the debian package I recompiled the kernel and compiled asterisk from CVS. When I use the command ( exten = _XXX,1,SIPDtmfMode(inband) ) it does not seem to work. Even putting a false mode does not give me a warning or something. Did I miss something? Any help/suggestion is appreciated! gr Jordi ___ 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] dtmfmode SIPDtmfMode
You must be calling SIPDtmfMode on incoming calls that are not SIP calls. Eg: zap call that you send to SIP ... this way it doesn't work. regards Martin On Fri, 14 Nov 2003, Jordi Haarman wrote: I get a 'Segmentation fault' now. A false mode just shows the error message now. Gr Jordi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Friday, November 14, 2003 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dtmfmode SIPDtmfMode Try again ... with latest CVS. Martin On Fri, 14 Nov 2003, Jordi Haarman wrote: Hi, I would like to be able to switch dtmf mode of SIP calls of local clients so the server can understand them and it can also be used when connected to a remote location. I saw that there is an application called SIPDtmfMode in cvs so instead of using the debian package I recompiled the kernel and compiled asterisk from CVS. When I use the command ( exten = _XXX,1,SIPDtmfMode(inband) ) it does not seem to work. Even putting a false mode does not give me a warning or something. Did I miss something? Any help/suggestion is appreciated! gr Jordi ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap won't load after CVS update
make sure the modules for your boards are loaded. ztcfg -vv shouldn't return with any errors. regards Martin On Fri, 14 Nov 2003, Matt Lawson wrote: I've just finished updating my Asterisk to the CVS version. Unfortunately, chan_zap won't load anymore. The hardware has not changed and the config files have not changed. I can re-install the two packages back and forth. The old one will still work. The new one won't. I tried updating to a brand-new zaptel and wcfxo modules, with no difference. This has got to be the most frustrating thing about dealing with Asterisk. This is also the same error I get trying to get the FXS cards to work (I have never succeeded). There must be something else in the Makefile or configuration files. Is there anything different regarding zap interfaces in the config files since maybe 3 months ago? The other differences I noticed were the modules chan_alsa.so, chan_oss.so (which didn't appear to be there before, or maybe in a different order), and a new requirement for libpri.so Same error message: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 1, with 0 conference users ERROR[16384]: File chan_zap.c, Line 5287 (mkintf): Unable to get span status: Inappropriate ioctl for device ERROR[16384]: File chan_zap.c, Line 6838 (load_module): Unable to register channel '1' WARNING[16384]: File loader.c, Line 305 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 400 (load_modules): Loading module chan_zap.so failed! zapata.conf: [channels] echocancelwhenbridged=yes echocancel=yes stripmsd=1 callerid=asreceived language=en context=incoming3121 signalling=fxs_ks rxgain=3.0 txgain=0.0 usecallerid=yes group=1 channel=1 echocancelwhenbridged=yes echocancel=yes stripmsd=1 usecallerid=no callwaiting=no callerid=intercom 9876543210 context=incoming3130 language=en signalling=fxs_ks group=1 channel=2 zaptel.conf can be blank or: loadzone=us defaultzone=us fxoks=1-2 ___ 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] Error in Incoming SIP call
what does show dialplan incoming show ? Also try using Dial(Zap/bla,10) instead Maritn On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Any suggestions will be appreciated! ___ 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] Red Alarm
Check if you configured the clocking from their circuit correctly. You need to have span=1,1 ... in zaptel.conf Martin On Tue, 4 Nov 2003, Eduardo Goncalves wrote: On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz [EMAIL PROTECTED] wrote: Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? I'd suggest your telco doing loopup and checking the circuit. My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ 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] Does externalip= do anything to help with SIP behind a Linux based NAT router?
It should. YOu need to do port forwarding on the firewall and use externip not externalip in general section of sip.conf. Refer to asterisk/configs/sip.conf.sample Martin On Tue, 4 Nov 2003, Leif Madsen wrote: I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ 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] Red Alarm
If you use TE410P make sure you have a recent zaptel from CVS. Martin On Tue, 4 Nov 2003, Eduardo Goncalves wrote: On Tue, 4 Nov 2003 09:42:36 -0600 (CST) Martin Pycko [EMAIL PROTECTED] wrote: Check if you configured the clocking from their circuit correctly. You need to have span=1,1 ... in zaptel.conf This is my zaptel.conf: span=1,1,0,cas,hdb3 alaw=1-8 em=1-8 loadzone = us defaultzone=us [ ]'s Eduardo ___ 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] Asterisk behind LinkSys NAT Routing
You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin On Mon, 3 Nov 2003, WipeOut wrote: Robert Mann wrote: Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and outside through * is not possible at all. I know I can not be the only one who has tried to do this. Please any help would be greatly appreciated. Robert, You need to get Asterisk onto a public IP address.. Using the DMZ function on the router will not work.. If you search the archives you will see that it has been attempted many times.. The reason is not in the IP but in the SIP headers.. they will be sent out from the Asterisk server with the internal IP address of the server, this means that when the SIP UA reads the SIP message and responds it will respond to the incorrect IP address.. So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. SIP UA's can be behind NAT but need nat=yes, canreinvite=no and qualify=yes set in the phone configuration in sip.conf.. Hope that helps.. Later.. ___ 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] E100P troubles
Maybe you need the straight through cable. Martin On Mon, 3 Nov 2003 [EMAIL PROTECTED] wrote: Hi, At least I have one E1 to test my E100P. My telco company in Spain has installed one LiteSpan 1540 NT (UTR 2M) I make a crossover cable between E100P and UTR. 1 - 4 2 - 5 after loading drivers red led on e100p is blinking and alarm is flashing on UTR. What is wrong ? my zaptel.conf inf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 best regards, Jorge Castellet [EMAIL PROTECTED] - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 4:08 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi , I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the non-geeks. I only had a short time to run your program last night but it worked well. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Robert Germany ___ 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] Asterisk behind LinkSys NAT Routing
It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? Later.. ___ 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] Asterisk behind LinkSys NAT Routing
Download the new code and see in asterisk/configs/sip.conf.sample It can't be easier than that. Martin On Mon, 3 Nov 2003, listas iPfone wrote: Hi! How to use that externip new parameter? Where in sip.conf and what is the format? thanks - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 3:34 PM Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? Later.. ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
It's for setting asterisk box with SIP support behind a NAT.
You need to do port redirection of eg. 5060 and then setup
externip=ip_of_your_nat_gateway
Martin
On Mon, 3 Nov 2003, Andrew Thompson wrote:
According to the source, it goes in the general section of sip.conf:
} else if (!strcasecmp(v-name, externip)) {
if (!(hp = gethostbyname(v-value))) {
ast_log(LOG_WARNING, Invalid address for externip keyword: %s\n, v-value);
} else {
memcpy(__ourip, hp-h_addr, sizeof(__ourip));
use_external_ip = 1;
}
Apparently, it expects the IP address that you want to use instead of the default
(bindaddr, I guess?).
Can someone tell me, does the second line that I quoted, with the gethostbyname
function mean that it will accept a hostname instead of just an IP? This would be
really really good for Dynamic IP users.
Note: I'm not savy enough to figure out how often this variable is refreshed!
This was taken from the CVS Viewer at: http://asterisk.espia-net.net/
chan_sip.c:
http://asterisk.espia-net.net/horde/chora/co.php/asterisk/channels/chan_sip.c?login=2r=1.204
-
Andrew Thompson
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 12:34 PM
Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Martin
On Mon, 3 Nov 2003, WipeOut wrote:
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
Martin,
Is externip and new parameter??
Does it do a similar thing for the server as what nat=yes does for the
phone?
Later..
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Re: [Asterisk-Users] Voicemail servermail and fromstring
Are you guys using voicemail2 ? Martin On Mon, 3 Nov 2003, Philipp von Klitzing wrote: Hi! The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Same here - please open a bug report on this. Cheers, Philipp ___ 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] Asterisk behind LinkSys NAT Routing
It's not for phones, it's for asterisk behind a NAT. Martin On Mon, 3 Nov 2003, Robert L Mathews wrote: At 11/3/03 10:00 AM, Martin Pycko [EMAIL PROTECTED] wrote: Is externip and new parameter?? It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Does it take an IP address, like externip=1.2.3.4? And does it then force the SIP messages for that channel to use the externip value instead of the server's local IP address? If so, that's useful; it will help people who know in advance that a certain phone is on one side of a NAT or the other. However, it would be nicer still if it could fix the SIP messages only when necessary, using a subnet mask or STUN, as has been proposed. The reason is that hard-coding an IP address to use when communicating with a certain client means you can't have a phone in an office (on the same side of the NAT as Asterisk) during the day, then take the phone home at night (on the other side of the NAT) and have it work without changing sip.conf. -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ 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] Red Alarm
I'd suggest your telco doing loopup and checking the circuit. regards Martin On Mon, 3 Nov 2003, Eduardo Goncalves wrote: Hi list, Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo ___ 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] HELP HELP HELP G729
Try starting asterisk from /usr/src/asterisk with the console asterisk -vvvcng regards Martin On Fri, 31 Oct 2003, Bartosz Jozwiak wrote: I just download a new one! And now I have that, it is even worser WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (2). - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 4:10 PM Subject: Re: [Asterisk-Users] HELP HELP HELP G729 You are not using the new codec binary, On Fri, 2003-10-31 at 12:59, Bartosz Jozwiak wrote: Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 179 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File translate.c, Line 225 (calc_cost): Translator 'lintog729b' appears to be broken and will probably fail. == Registered translator 'lintog729b' from format SLINR to G729A, cost 9 -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] Setvar SIP_CODEC
[extensions.conf]
exten = 123456,1,SetVar,SIP_CODEC=ulaw
exten = 123456,2,Dial(${TRUNK}/${EXTEN})
The problem is with the SetVar function, the debug shows that the
function is executed, but after that, * sends the media capability to
the phone with g729 as preferred codec.
SIP_CODEC is was supposed to only change the codec of the incoming call,
eg: asterisk responds with ANSWER with ulaw codec ...
But it won't change anything with the 2nd call.
regards
Martin
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RE: [Asterisk-Users] Outgoing CallerID
Calling Number (len=12) [ Ext: 0 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '4330' ] It might be that the number plan is international Change pridialplan to unknown in zapata.conf Martin Called Number (len=11) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2840' ] Cant figure out what's wrong? regards Mickey Binder ÿÿÿÀ²×«ÉÿRÇ«²f¢)à+-Ë^®+$ýK ®ÏåËlýØ éÿr¡¶Úÿÿùb²Ûÿv(ºoÜ¢oæj)fjåËbú?jË^®+$þë ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting * with G729 licences
check 'screen -d -m asterisk -vvvcng' regards Martin On Thu, 16 Oct 2003, CW_ASN - Gus wrote: Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the old way? Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting * with G729 licences
If you use g729 codec The Voiceage part of the codec breaks it. regards Martin On Thu, 16 Oct 2003, CW_ASN - Gus wrote: Martin: This works ok. Doing a 'ps ax | grep aste' shows: 3071 ?S 0:00 SCREEN -d -m asterisk -vvvcng 3072 pts/2S 0:02 asterisk -vvvcng This means that I need to run * in this way forever? Gus - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 16, 2003 6:13 PM Subject: Re: [Asterisk-Users] Starting * with G729 licences check 'screen -d -m asterisk -vvvcng' regards Martin On Thu, 16 Oct 2003, CW_ASN - Gus wrote: Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the old way? Thanks in advance, Gus ___ 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] Asterisk Manager
It's an application and not a cli command, put it in extensions.conf [default] exten = s,1,System(ls /tmp/log) regards Martin On Tue, 14 Oct 2003, Chee Foong wrote: Hello mate, I tried that, i get No such command 'System(ls)'. I can't even make it work on CLI. I am able to execute linux command (via CLI) by prefix command with a !. I would like to know how to do it throut the manager appllication. Thanks for you reply. CF - Original Message - From: [EMAIL PROTECTED] To: Chee Foong [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 2:08 PM Subject: Re: [Asterisk-Users] Asterisk Manager On Tue, 14 Oct 2003, Chee Foong wrote: Can I execute linux command like(ls, mkdir) through the Manager interface? nain*CLI show application system nain*CLI -= Info about application 'System' =- [Synopsis]: Execute a system command [Description]: System(command): Executes a command by using system(). Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee. ___ 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] WARNING[49159]
REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.33:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.33:5060 From: sip:[EMAIL PROTECTED];tag=483a-f0f0b8ca To: 35 sip:[EMAIL PROTECTED];tag=as3028bf6d Call-ID: [EMAIL PROTECTED] CSeq: 26289 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 180 Contact: sip:[EMAIL PROTECTED];expires=180 Date: Tue, 14 Oct 2003 16:30:06 GMT Content-Length: 0 to 192.168.0.33:5060 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552 From: asterisk sip:[EMAIL PROTECTED];tag=as787ccf10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/1 (no NAT) to 192.168.0.33:5060 Sip read: CLI SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552 Call-ID: [EMAIL PROTECTED] Contact: 35 sip:[EMAIL PROTECTED] CSeq: 102 NOTIFY From: asterisk sip:[EMAIL PROTECTED];tag=as787ccf10 Supported: timer To: sip:[EMAIL PROTECTED];tag=02f8-f0f0f208 Server: ipDialog SipTone 1.2.0 rc V UAS Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY Content-Length: 0 11 headers, 0 lines localhost*CLI - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:39 PM Subject: Re: [Asterisk-Users] WARNING[49159] It means that your SIP device sends some SIP packets and we can't parse the CSeq numbers. Can you paste the 'sip debug' of that ? regards Martin On Tue, 14 Oct 2003, listas iPfone wrote: Hi All I receive that warning message: WARNING[49159]: File chan_sip.c, Line 2220 (__transmit_response): Unable to dete rmine sequence number from '' What is it? There is some documentation with all error messages? thanks miklos ___ 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] use of SIP SHOW CHANNELS question
Use tabulator button for asterisk to help you guess the name.
regards
Martin
On Tue, 14 Oct 2003, Walker Haddock wrote:
I am trying to figure out the correct syntax for the CLI command SIP SHOW
CHANNELS. I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected
such as:
-- Zap/15-1 is ringing
-- Zap/15-1 answered SIP/206-4299
asterisk*CLI sip show channel SIP/206-4299
No such SIP Call ID 'SIP/206-4299'
I always get the No such SIP Call ID ...
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] On an RH9 box, where does wcusb get loaded?
If you do make config in the zaptel then it's going to be loaded during bootup. Otherwise it's not being loaded unless you do 'modprobe wcusb' regards Martin On 14 Oct 2003, tom wrote: From - Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by sccrmxc11.comcast.net (sccrmxc11) with ESMTP id 20031014185753s1100nos46e; Tue, 14 Oct 2003 18:57:53 + Received: from rwcrmhc12.comcast.net (localhost[127.0.0.1]) by comcast.net (rwcrmhc12) with ESMTP id 20031014185753014006qmtie; Tue, 14 Oct 2003 18:57:53 + From: Mail Delivery Subsystem [EMAIL PROTECTED] Subject: Returned mail: delivery problems encountered Message-Id: [EMAIL PROTECTED]@[EMAIL PROTECTED] Date: 14 Oct 2003 18:57:51 + To: [EMAIL PROTECTED] Mime-Version: 1.0 Content-Type: multipart/report; report-type=delivery-status; boundary=_3f8c472f.7355.0+comcast.net=_ X-Evolution-Source: pop://[EMAIL PROTECTED]/ --_3f8c472f.7355.0+comcast.net=_ Content-Type: text/plain A message (from [EMAIL PROTECTED]) was received at 14 Oct 2003 18:57:41 +. The following addresses had delivery problems: [EMAIL PROTECTED] Permanent Failure: 550_5.1.1_[EMAIL PROTECTED]..._User_unknown Delivery last attempted at Tue, 14 Oct 2003 18:57:50 - --_3f8c472f.7355.0+comcast.net=_ Content-Type: message/delivery-status Reporting-MTA: dns; comcast.net Arrival-Date: 14 Oct 2003 18:57:41 + Final-Recipient: rfc822; [EMAIL PROTECTED] Action: failed Status: 5.0.0 550_5.1.1_[EMAIL PROTECTED]..._User_unknown Diagnostic-Code: smtp; Permanent Failure: Other undefined Status Last-Attempt-Date: Tue, 14 Oct 2003 18:57:50 - --_3f8c472f.7355.0+comcast.net=_ Content-Type: message/rfc822 Received: from [192.168.0.33] (dnvr-dsl-gw28-poolc85.dnvr.uswest.net[65.101.254.85]) by comcast.net (rwcrmhc12) with SMTP id 20031014185741014008t5g9e (Authid: landslide_x); Tue, 14 Oct 2003 18:57:42 + Subject: On RH9, where is wcusb loaded? From: tom [EMAIL PROTECTED] To: [EMAIL PROTECTED] Content-Type: text/plain Organization: Message-Id: [EMAIL PROTECTED] Mime-Version: 1.0 X-Mailer: Ximian Evolution 1.2.2 (1.2.2-4) Date: 14 Oct 2003 12:58:57 -0600 Content-Transfer-Encoding: 7bit I have a dev kit lite, and I'd like to have asterisk up and running when I boot my linux box, but there are couple of things that are preventing this from happening. First and foremost, the wcusb and zaptel modules are loaded at startup, but wcxfo is not. In order to get everything running (and loaded in the correct order), I have to remove wcusb, load wcfxo, reload wcusb, and then run ztcfg. Any ideas on how I might all of this loading in the correct order so that I won't have to keeps putzing with it when I boot. Thanks. Tom --_3f8c472f.7355.0+comcast.net=_-- ___ 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] Digium cards just for timing
With the musiconhold and SIP-SIP call it turnes out that you need to disable silence supporesion on your phones/gateways since the timing is taken from the coming stream (but only for musiconhold AFAIK) regards Martin On Tue, 14 Oct 2003, Michael Ulitskiy wrote: Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those cards provide some reliable timing. So I have no choice and wish to buy a cheapest Digium card just for timing. I have no PSTN ports, it's pure voip environment here. So my question is whether any Digium card would be ok or I have to buy some specific card? I'm looking at X100P card as it is the cheapest one. Would it be enough? Thank you. Michael ___ 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] Strange message !! Unknown IE 76 (Unknown Information Element)
It means that this IE is not implemented in the libpri or is not very standarized. regards Martin On Mon, 13 Oct 2003, Marcel Prisi wrote: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? 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
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
My fault then :) I was thinking only in terms of Q931 spec ... Martin On 13 Oct 2003, Klaus-Peter Junghanns wrote: Hi Martin, it's not implemented in libpri but very well standarized (ETS 300 097). regards, kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email:[EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko: It means that this IE is not implemented in the libpri or is not very standarized. regards Martin On Mon, 13 Oct 2003, Marcel Prisi wrote: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? 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 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] VoiceMail fromstring?
I just tested fromstring and emailbody with voicemail2 and a farily new code and it's working. I don't know what you're doing wrong ... but something for sure. regards Martin On Mon, 13 Oct 2003, John Todd wrote: I would recommend then doing grep fromstring /usr/src/asterisk/apps/app_voicemail2.c Martin On Fri, 19 Sep 2003, Ben Bloomberg wrote: I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? [snip] Martin - I examined the source, but I am still un-enlightened. :-) I cannot get fromstring or emailbody working reliably. Even with the minimalist settings below, the header or body did not change (other than serveremail which seems to be set appropriately.) Interestingly and perhaps as an additional problem, the timezones also don't seem to work correctly in the voicemail message, either - the time in the email message is Eastern time (the TZ to which that server is set.) My CVS is Asterisk CVS-10/13/03-18:38:10. What I am doing incorrectly? JT [general] format=wav [EMAIL PROTECTED] attach=yes fromstring=Foo emailbody=New vm now [zonemessages] eastern=US/NewYork|'vm-received' Q 'digits/at' IMp central=US/Central|'vm-received' Q 'digits/at' IMp mountain=US/Mountain|'vm-received' Q 'digits/at' IMp pacific=US/Pacific|'vm-received' Q 'digits/at' IMp [default] 2413669780 = ,john todd,[EMAIL PROTECTED],,|tz=pacific A message left in that mailbox results in: Date: Mon, 13 Oct 2003 18:53:49 -0400 From: Asterisk PBX [EMAIL PROTECTED] To: john todd [EMAIL PROTECTED] Subject: [PBX]: New message 2 in mailbox 2413669780 Dear john todd: Just wanted to let you know you were just left a 0:01 long message (number 2) in mailbox 2413669780 from 2155821314, on Monday, October 13, 2003 at 06:53:49 PM so you might want to check it when you get a chance. Thanks! --Asterisk Content-Type: audio/x-wav; name=msg0002.wav Content-Description: Voicemail sound attachment. Content-Disposition: attachment; filename=msg0002.wav ___ 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] VoiceMail fromstring?
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines defined
by the user's tz= setting.
Well it's the task for those who add features to have a global-system
thinking. The emailbody was added way before the timezones ...
Also, there seems to be a character limit for the length of
emailbody= that is a bit short - I get the last part of my messages
chopped off at a predictable point (seems to be around the 500th
character of the emailbody= line that it gets snipped.)
That can be easily changes since the static array is used.
Martin
JT
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Re: [Asterisk-Users] Call to 06302 aborted, insufficient bandwidth
What protocol ? H323 ? Which channel driver ? chan_oh323 or chan_h323 ? Martin On Wed, 8 Oct 2003 [EMAIL PROTECTED] wrote: Hi! When I try to make a call with ohphone, that is the message I get: Call to 06302 aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia ___ 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] modprobe wct1xxp: unresolved symbol zt_alarm_notify, but zaptel module IS loaded
Compile zaptel without PPP support or compile PPP support into your kernel. You can do the first in zaptel/Makefile Martin On Wed, 8 Oct 2003, Ron Arts wrote: This is probably not a direct asterisk problem, but I am quite at a loss here. I am experiencing problems with zaptel drivers Am trying to install asterisk on a system that is managed by a third party. They only accept software in .rpm format for various reasons. Anyway I created my own rpms for zaptel. This rpm compiles the driver modules for the specific kernel on the particular machine. When I install the rpm, depmod -a gives: [EMAIL PROTECTED] misc]# depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/kernel/drivers/char/drm/sis.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/tor2.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/wcfxo.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/wcfxs.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/wct4xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-0NBSsmp/misc/ztdynamic.o I installed zaptel.o like this: [EMAIL PROTECTED] misc]# modprobe zaptel /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_unit_number /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_input /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_input_error /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_unregister_channel /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_output_wakeup /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_channel_index /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: unresolved symbol ppp_register_channel /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: insmod /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o failed /lib/modules/2.4.21-0NBSsmp/misc/zaptel.o: insmod zaptel failed [EMAIL PROTECTED] misc]# modprobe ppp_generic [EMAIL PROTECTED] misc]# modprobe zaptel After that: [EMAIL PROTECTED] misc]# modprobe wct1xxp /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_ec_chunk /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_unregister /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_alarm_notify /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_transmit /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_rbsbits /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_receive /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: unresolved symbol zt_register /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: insmod /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o failed /lib/modules/2.4.21-0NBSsmp/misc/wct1xxp.o: insmod wct1xxp failed How can this be, becasue zaptel DID load successfully. I must be missing something really obvious. [EMAIL PROTECTED] misc]# lsmod Module Size Used byNot tainted zaptel189760 0 (unused) ppp_generic22284 0 [zaptel] slhc6356 0 [ppp_generic] ipt_MASQUERADE 2816 1 (autoclean) ip_nat_ftp 4608 0 (unused) e1000 57528 1 ipt_TOS 1696 4 (autoclean) ipt_REJECT 3872 1 (autoclean) ipt_LOG 4352 17 (autoclean) ipt_limit 1696 18 (autoclean) ipt_state 1056 4 (autoclean) ip_conntrack_ftp5376 2 [ip_nat_ftp] 8139too17248 1 mii 3996 0 [8139too] iptable_mangle 2848 1 (autoclean) iptable_nat28468 3 (autoclean) [ipt_MASQUERADE ip_nat_ftp] ip_conntrack 37428 4 (autoclean) [ipt_MASQUERADE ip_nat_ftp ipt_state ip_conntrack_ftp iptable_nat] iptable_filter 2368 1 (autoclean) ip_tables 17312 11 [ipt_MASQUERADE ipt_TOS ipt_REJECT ipt_LOG ipt_limit ipt_state iptable_mangle iptable_nat iptable_filter] md 50240 0 (unused) rtc 8316 0 (autoclean) ext3 70432 3 jbd5 3 [ext3] gdth 81152 5 I hope someone can give me any clues. BTW any software I use is straight from CVS. Thanks, Ron -- Netland Internet Services bedrijfsmatige internetoplossingen http://www.netland.nl Kruislaan 419 1098 VA Amsterdam info: 020-5628282 servicedesk: 020-5628280 fax: 020-5628281 (A)bort, (R)etry, (G)et a beer? ___
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
cd /usr/src/asterisk; make config; cd /usr/src/zaptel; make config regards Martin On Tue, 7 Oct 2003, john lawler wrote: Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? I appreciate your willingness to share your knowledge and expertise. jl ___ 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] Answer on second ring - need it on first.
Yeah, I'd put usecallerid=no since I bet it's set by default as yes. Martin On Sat, 4 Oct 2003, Richard Scobie wrote: Martin Pycko wrote: take out usecallerid=yes in zapata.conf Martin Thanks Martin, but my zapata.conf is : [channels] echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=6 context=incoming signalling=fxs_ks group=1 channel = 1-2 Perhaps I need a usecallerid=no in there. I'll test this when I'm back at work. I have not changed this prior to rebuilding the source. Regards, ___ 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] Small problem with FAX and Modem.
What if you separate the fax machine channels to diffrent contexts that don't call application Monitor ? It's for outgoing calls and for incoming calls if you have certain extensions for faxes you can call StopMonitor application. regards Martin On Fri, 3 Oct 2003, Nicholas Romero wrote: Is there a good way to detect FAX and Modem on a call that is established and then take some sort of action? What I have is a situation that all calls going out through an asterisk system are being recorded. Some of those calls are internal fax machines or modems. When monitoring is turned on is causes some funky transmission errors with the modems and sometimes Faxes to the point where about 50% of the time the devices just give up. What I would like to do is if the systems detects a modem or a fax jump to another point in the sequence that disables the monitoring. On inbound calls this is more obvious but on outbound calls I am not sure of how to accomplish it. Sort of an aside to this is that after I started recording all the calls I have also seemed to introduce some echo somewhere. The echo was not present before monitoring but not sure why it would introduce it either. Any help appreciated. -Nicholas ___ 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] Answer on second ring - need it on first.
take out usecallerid=yes in zapata.conf Martin On Sat, 4 Oct 2003, Richard Scobie wrote: After some months of Make updates, I have just deleted my Zaptel and Asterisk source directories and done cvs checkout 's of asterisk and zaptel, in order to clean up the trees. After re-installing, I am finding that when dialling into an X100P, that Answer is now answering on the second ring, where it always used to answer on the first before. In the console, Starting simple switch on 'Zap/1-1' appears halfway through the first ring and Executing Answer(Zap/1-1, ) in new stack appears at the end of the second. I cannot recall having changed anything previously, in order for it answer on the first, but I would really like the old behaviour back. Thanks, Richard ___ 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] error message 49159
It's a WARNING, so if you want to know why your phone doesn't work you can read it or ignore it. regards Martin On Thu, 2 Oct 2003, Brian Capouch wrote: Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but an artifact of something asterisk is trying to do? I have seen these messages pretty much since the beginning of time, and I figured something was out of spec with my phones. I can't tell from what you say whether it is normal or not to see those messages? Thanks. B. ___ 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] Any way to get out of a remote console without stopping *
use quit or ctrl-D Martin On Thu, 2 Oct 2003, Andy Hester wrote: This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero ___ 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] X100P - Busydetect / calls being disconnected - Australia; tip.
Because of the nature of busydetect algorithm busycount shouldn't be set to less than 8. It's 10 by default. Just imagine that you dial a number that is attached to some speed dial key. It'll surely cause hangup if busydetect 8. Martin On Sat, 27 Sep 2003, Shaun Ewing wrote: Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still happened. I thought that there was some remote possibility that the phone company was reversing the line on answering long distance calls, so I switched to fxs_ls instead of fxs_ks - no difference. Various things were tried to no avail, until I made a long distance call over a different carrier to our usual carrier (we use Optus, I made the call over Telstra). When the remote end answered, my end disconnected. What was happening was, when the call is answered, 5 quick chirps are sent down the line. However, because of the bug in the Cisco 7960 causing the first 1/2 a second or so of a conversation to be cut off - I didn't hear these chirps and as such I didn't think of the next bit: Basically, because I had busycount set to 3 and busydetect set to yes, these chirps were being detected by the busydetect function and causing the call to be disconnected. I raised the busycount to something safe (8) and this no longer happened. This has me worried for a while, especially as I'd just disconnected the old PBX a few days ago and spent a nice amount of money on Cisco 7960 and 7940 IP phones (and will probably be ordering more in the near future). Anyway, I'm pleased to report that everything is now working perfectly and I'm extremely happy with Asterisk. I'd contribute, but alas I'm not much of a C programmer. -Shaun ___ 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] Set context based on CID...
[incoming]
exten = _X.,1,DBGet(NEWCONTEXT=context/${CALLERIDNUM})
exten = _X.,2,Goto(${NEWCONTEXT},${EXTEN},1)
exten = _X.,102,Goto(allother,${EXTEN},1)
Martin
On Fri, 26 Sep 2003, Matt McIntyre wrote:
I was wondering if someone might be able to offer a suggestion to me
about how I might go about dropping a caller into a context specific to
their CID. For example, I would like to be able to dial Asterisk from a
specific number (a mobile phone) and have it drop me into a context
other then the one that normal callers receive that has more options
tailored to things I might want to do. I assume that answer can
somehow be used to do this but I thought I might ask the experts and see
what they might have to say.
Thanks in advance,
(You guys are great)
Matt
^
! Matt McIntyre (KF4FGZ)
! Certified Novell Administrator
! (336) 334-1134 (Campus telephone)
! (336) 215-7199 (Mobile telephone) - Please note the
change
! (336) 334-1134 (Facsimile)
! E-MAIL: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
! AIM: MixMANJaVa
! ICQ: 11956085
^
___
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Re: [Asterisk-Users] Sometimes pri channels restart during * is runnig ?
Asterisk does restart on idle channels every (I think) 20 minutes to ensure that the remote switch treats the idle channels (on our side) as idle. I don't get the channelid problem that you're reporting, maybe the pri debug span span_no is a good idea to post. regards Martin On Thu, 25 Sep 2003, Thomas Haeger wrote: Hi all, i have observed, that sometimes all BChannels on my Zaptel Pri device (E400P) will be restarted. The E400P is connected to another pri switch. In the traces from the other side (pri switch) i can see that libpri request for the channelid is 255. Is this a bug or a feature ...? Or, can it be a bug on the other side (terminator switch) ? Have anyone an idea ? Thanks, Thomas. ___ 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: AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
The call does not get compleated on the PRI so you should check the pri debug span 1 on your 2nd box. regards Martin On Tue, 23 Sep 2003, Thomas Haeger wrote: I have tried it with a timeout and without... here the * output for the first side: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack -- Called thaeger:[EMAIL PROTECTED]/99033283077731 -- Call accepted by 62.180.50.212 (format ALAW) -- Format for call is ALAW -- Hungup 'IAX2[62.180.50.212:4569]/2' == No one is available to answer at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' and here from the other side: -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format = 8 -- Executing SetCallerID([EMAIL PROTECTED]:4569]/1, 033283077731) in new stack -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g3/033283077731) in new stack -- Called g3/033283077731 -- Channel 1, span 3 got hangup -- Hungup 'Zap/63-1' == No one is available to answer at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steven Critchfield Gesendet: Dienstag, 23. September 2003 17:13 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote: Hi all, can somebody explain this ? Do you have something like a |15 in the dial string? Do you have logs to show what asterisk did? -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Windows Media Player Error
make sure the 'format=wav' in voicemail.conf Martin On Tue, 23 Sep 2003, Steve Totaro wrote: I am getting the following error in Windows Media Player Version 9 when listening to voice mails. ClassFactory cannot supply requested class (Error=80040111) Any ideas? I tried searching the net but only found references to DivX. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
gdb /usr/src/asterisk core.6044 then 'bt' Martin On Tue, 23 Sep 2003, jerk face wrote: I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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] Segmentation Fault on reload (gdb output included)
actually gdb /usr/sbin/asterisk core.6044, sorry On Tue, 23 Sep 2003, jerk face wrote: I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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] G.729A + Cisco AS5300
It does support it but you have to uncomment -DWANT-G729 in h323/Makefile On Mon, 22 Sep 2003, Eric Wieling wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. On Mon, 2003-09-22 at 04:55, Chee Foong wrote: hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile
Re: [Asterisk-Users] built in dial functions?
The implementation of *72 is done for FXS port (the one that gives the dialtone). However you could implement that with some extensions.conf logic. regards Martin On Sat, 20 Sep 2003, Rich Adamson wrote: Martin, That makes sense... but how would one actually use *72#, as an example, when * has two x100p FX ports? i.e, can one enable call forwarding on one fx port and not the other? If I want to call forward my extension (say extn 3000) to extn 3001, is there a way for the user to do that without changing config files? Rich These functions are implemented only for chan_zap (zaptel hardware) and work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I know. regards Martin On Fri, 19 Sep 2003, Rich Adamson wrote: Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial *78# and hang up. From another sip phone I Dial that extenstion and the 7960 rings. I expected the call to roll over to voicemail or something. Am I missing something here, or are these functions not expected to work on a per-extension basis? I was assuming (probably incorrectly) these functions were custom calling features implemented within * for all extensions. Are my assumptions wrong or do I have to implement something for these to work? TIA, Rich ___ 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 ---End of Original Message- ___ 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] RE: Very bad echo (appears that...)
So 'zap show chanenl channel-no' shows that the echocan is turned on ? Martin On Sun, 21 Sep 2003, Asterisk PBX wrote: Oh, I forgot to say, zaptel/wcfxo is compiled with: KFLAGS+=-DECHO_CAN_MARK2 KFLAGS+=-DAGGRESSIVE_SUPPRESSOR (and, Brian, my jack is wired correct..) -Original Message- From: Lenny Tropiano [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk PBX Sent: Sunday, September 21, 2003 5:02 PM To: '[EMAIL PROTECTED]' Subject: Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ 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] built in dial functions?
These functions are implemented only for chan_zap (zaptel hardware) and work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I know. regards Martin On Fri, 19 Sep 2003, Rich Adamson wrote: Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial *78# and hang up. From another sip phone I Dial that extenstion and the 7960 rings. I expected the call to roll over to voicemail or something. Am I missing something here, or are these functions not expected to work on a per-extension basis? I was assuming (probably incorrectly) these functions were custom calling features implemented within * for all extensions. Are my assumptions wrong or do I have to implement something for these to work? TIA, Rich ___ 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] Hangups after voicemail
Do you have silence in the channel when the remote user hangs up or busy
tone ?
If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung up.
Therefore, I've added the following hideous hack - er, code - to voicemail2.c.
It starts right after the call to play_and_record() in leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3)) {
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been recorded, if
the # key wasn't pressed, if the channel still exists, and if it's a Zap
channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel is controlled
by another thread. I also modified channel.c so that ast_channel_free() sets
chan to NULL after it's freed, just in case. Is there anything else I should
be aware of? The code seems to work in my testing, resulting in a proper
hangup right after the voicemail has been recorded. I'm not up on my Asterisk
internals, so I'm not totally confident about this.
Thanks,
Chris
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Re: [Asterisk-Users] calls terminating abnormally
Can you send a pri debug span span_no trace ? Or do you have an analog T1/E1 ? regards Martin On Wed, 17 Sep 2003, denzel-infotechs wrote: hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangups after voicemail
set silencethreshold to 50 and before voicemail call
responsetimeout,0
regards
Martin
On Wed, 17 Sep 2003, Christian Hecimovic wrote:
Hi Martin,
Yes, silence detection in voicemail is working. I am using Voicemail2 with the
silencethreshold set to 256. However, the line doesn't hang up after the
silence is detected; instead, Voicemail2 exits after recording the voicemail
correctly, and Asterisk loops back into the main menu as if the # key was
pressed because the channel is still alive. Then it times out after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the incoming context.
It all works great, except for when the line gets tied up by the DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it didn't work. The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem detecting
hangups when a voice conversation is taking place, since Asterisk obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to outside line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up or busy
tone ?
If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop
start lines. So, after someone leaves a voicemail and then hangs up,
Asterisk doesn't know it, exits VoicemailMain2, and loops back to the
corporate greeting, tying up the line even though the outside caller has
hung up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c. It starts right after the call to play_and_record() in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3)) {
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been recorded,
if the # key wasn't pressed, if the channel still exists, and if it's a
Zap channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel is
controlled by another thread. I also modified channel.c so that
ast_channel_free() sets chan to NULL after it's freed, just in case. Is
there anything else I should be aware of? The code seems to work in my
testing, resulting in a proper hangup right after the voicemail has been
recorded. I'm not up on my Asterisk internals, so I'm not totally
confident about this.
Thanks,
Chris
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Re: [Asterisk-Users] Distinctive ringing
The X100P together with asterisk does not support the distinctive ringing detection on the line. Asterisk however can generate the distinctive ring over FXS ports. regards Martin On Tue, 16 Sep 2003, Robert Boardman wrote: Hi I've just signedup for Distinctive ringing on my PSTN line in the UK, could anyone explain what I need to add in the conf files to detect and route based on in comming Distinctive ringing Thanks in advance for your help Robb ___ 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] Voicemail time limit?
Did you see /etc/asterisk/voicemail.conf ? maxmessage=120 is 2 minutes Martin On Fri, 12 Sep 2003, Rich Adamson wrote: Is there a way to limit the duration of any single voicemail recording? I'd like to put a cap on that limit, say 2 minutes or whatever, for those long winded individuals and can't seem to find a reference for it. ___ 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] Voicemail 1 and 2
The Voicemail2 is better one, has more bug fixes, more functionality and Voicemail (1) should stop existing soon. regards Martin On Fri, 12 Sep 2003, Olle E. Johansson wrote: While on the subject of Voicemail - what is the difference between voicemail() and voicmail2() ? The show application commands contains exactly the same text, giving no hints. /Olle ___ 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] Asterisk using a h323 gateway
exten = _9X.,1,Dial(H323/[EMAIL PROTECTED]) If it's not working it's worth looking at the reson: h.323 debug h.323 trace 3 regards Martin On Fri, 12 Sep 2003, Cerrajetto wrote: Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 - PSTN gw)? - Asterisk ip: 192.168.1.10 - h323-PSTN gw: 192.168.1.20 I've tried: exten = _9,1,Dial(OH323/192.1.1.20) or exten = _9,1,Dial(OH323/[EMAIL PROTECTED]) but it does not work at all. If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are routed to the PSTN perfectly. What is the correct way to route some calls from Asterisk to another h323 gateway? Thank you, Mark ___ 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] E400P woes
So you don't receive any answer from the other side ? Is the circuit in alarm ? Can they do remote loopup test ? It might be that they don't have their D-channel turned on ... Martin On Fri, 12 Sep 2003, Alastair Maw wrote: OK, so I've done this: *CLI pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 Sending Set Asynchronous Balanced Mode Extended [00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended The above is repeated about once a second. Our provider does indeed have a System-X switch. Is this a known problem then? Is there no way to resolve it? I couldn't find anything on Google about it... I'd put a TE410P card in instead, but the 1u servers we have are all P4s and don't have 3.3V PCI slots. The official Word from Digium is that they'll have a 5V version of the TE410P out in about six weeks' time, but we have some services that need to go live on these new E1 lines in about three weeks time, plus we need to do some testing, etc. Time to buy a Xeon with 3.3V slots, I guess. :( -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ 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] Voicemail 1 and 2
you can copy voicemail.conf.sample to be your voicemail.conf ... Martin On Fri, 12 Sep 2003, Olle E. Johansson wrote: Steven Critchfield wrote: On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote: While on the subject of Voicemail - what is the difference between voicemail() and voicmail2() ? From the application stand point there is little difference, but from the configuration stand point there is a fair amount of difference. Consult the sample configs to start you on your path to deciding what you want. Steven, Thank your for responding. I find only one config in the sample directory - voicemail.conf.sample and it looks the same as my voicemail.conf - should I look in another place? /Olle ___ 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] (no subject)
What does 'dmesg' says ? Martin On Fri, 12 Sep 2003, James Sharp wrote: On Fri, 12 Sep 2003, Jim Paraschou wrote: I have problem with a TDM40B installation. When i modprobe wcfxs the error i get is the following: /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod /lib/modules/2.4.19-4GB/misc/wcfxs.o failed /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod wcfxs failed Does anybody know the poblem? Means the module can't find the card anywhere. Is the card inserted properly? Does it show up if you do an lspci? ___ 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] IAX, IAX2 and authenticatyion
IAX2 uses 4569 UDP port. You can see iax2 calls with iax2 show channels. Also you can send the calls in IAX2 simply by Dial(IAX2/blahblah) Also IAX2 is more recent, has more fixes and has the trunking mode to save bandwidth if you're sending more than 10 calls to another destination. regards Martin On Fri, 12 Sep 2003, Dan wrote: Hi, I have some questions regarding IAX, IAX2 and encrypted authentication. How can I know if IAX or IAX2 is used between two * servers? There is any guide about how to configure encrypted authentication (not in clear text)between two * servers? I hear on this list a couple of days ago that port 5036 is the default one for IAX and something else (4XXX) for IAX2. Trying 'iax show channels' in CLI during an active channel between the two * servers shows me that IAX is used. What to do to use IAX2 instead? Which are the main differences between IAX and IAX2? Thanks a lot, Dan ___ 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] TDM40B Installation problem
what does 'dmesg' says ? Martin On Fri, 12 Sep 2003, Jim Paraschou wrote: I have problem with a TDM40B installation. When i modprobe wcfxs the error i get is the following: /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod /lib/modules/2.4.19-4GB/misc/wcfxs.o failed /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod wcfxs failed Does anybody know the poblem? __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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] IAX, IAX2 and authenticatyion
Because IAX2 in trunking mode adds the 10 bytes header ... So It might not be a good idea if you're going to have only two calls. Martin On Fri, 12 Sep 2003 [EMAIL PROTECTED] wrote: On Fri, 12 Sep 2003, Martin Pycko wrote: Also IAX2 is more recent, has more fixes and has the trunking mode to save bandwidth if you're sending more than 10 calls to another destination. martin, why 10 calls? is this codec dependent? thanks in advance for the info... - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PROBLEM RECIVING CALLS AT FXO
Do you have an error about receiving the callerid ?
What happens when you pick up the Zap/2 phone ?
regards
Martin
On Thu, 11 Sep 2003, Alvaro Parres wrote:
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook
in strange state 6 on channel 1
My config files are:
zaptel.conf -
fxsks=1
fxoks=2-3
loadzone = us
defaultzone=us
--- zapata.conf -
[channels]
relaxdtmf=yes
busydetect=yes
callprogress=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1-2
;immediate=no
context=bell
signalling=fxs_ks
mailbox=yes
;callerid=asrecive
channel=1
context=home
group=2
signalling=fxo_ks
channel=2-3
callerid=FIJO 200
channel=3
callerid=INALAMBRICO 100
channel=2
extensions.conf
[dialout]
ignorepat = 9
exten = _9.,1,Dial(${PSTN}/${EXTEN:1},120,T)
exten = 9,1,Dial(Zap/g1/)
exten = 9,2,Congestion
Thanks.
--
Alvaro I. Parres Peredo
Director de IT
Xmarts, Soluciones Inteligentes
Bernardo de Balbuena #35
Tel: 36301294
http://www.xmarts.com
-
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Re: [Asterisk-Users] Request for best practices
It should work but you need to do Goto(extensions,666${EXTEN},1)
Martin
On Wed, 10 Sep 2003, Ernest W. Lessenger wrote:
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to direct-dial each other's extensions. We want this to
work like a real centrex, in that seven-digit numbers should try (1)
local VoIP extensions, and then (2) local PSTN numbers. Ten-digit
numbers should dial (1) long-distance VoIP extensions, and then (2)
long-distance PSTN numbers.
Here's my plan so far, does anyone have a better way? Will Goto() work the
way I expect it to (i.e. will the extension I specify be pattern matched)?
==Extensions.conf==
[area555]
exten = _NXXNXXX, 1, Goto(extensions,555${EXTEN})
include = extensions
[area666]
exten = _NXXNXXX,1, Goto(extensions,666${EXTEN})
include = extensions
[extensions]
exten = 5551234567, 1, Macro(stdexten, 1234, SIP/user1)
exten = 6661234567, 1, Macro(stdexten, 1235, SIP/user2)
include = longdistance
[longdistance]
exten = _NXXNXX, 1, Dial(${Nufone},${ARG1})
exten = _NXXNXX, 2, Congestion()
[macro-stdexten]
... as in demo ...
==Sip.conf===
[user1]
...
context = area555
[user2]
...
context = area666
Thanks,
--Ernest
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Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
yes, i had 'callprogress=yes', and i commented it. now the time out is working. Thank you very much by disabling callprogress in an analog environment, does it affet the call disconnection? IT does but you should only use it for analog channels ... and propably only FXOs. Martin Surajee - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 08, 2003 10:32 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! Do you have callprogress=yes in zapata.conf ? If yes, then comment it out. Also you could send some trace from the console including pri debug span span-no Martin On Mon, 8 Sep 2003, Surajee Ratnayake wrote: Is it a problem with E1, bcos, when we dial a SIP extension from the same asterisk box it timeouts but not the Zap ones.. We tried without |t, but it didn't work.. still keeps on ringing forever... :-( Surajee - Original Message - From: [EMAIL PROTECTED] To: Surajee Ratnayake [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:29 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! surajee: what happens if you remove the |t ? still no timeout ? -wasim On Mon, 8 Sep 2003, Surajee Ratnayake wrote: hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
Do you have callprogress=yes in zapata.conf ? If yes, then comment it out. Also you could send some trace from the console including pri debug span span-no Martin On Mon, 8 Sep 2003, Surajee Ratnayake wrote: Is it a problem with E1, bcos, when we dial a SIP extension from the same asterisk box it timeouts but not the Zap ones.. We tried without |t, but it didn't work.. still keeps on ringing forever... :-( Surajee - Original Message - From: [EMAIL PROTECTED] To: Surajee Ratnayake [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:29 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! surajee: what happens if you remove the |t ? still no timeout ? -wasim On Mon, 8 Sep 2003, Surajee Ratnayake wrote: hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee ___ 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] The old versus new TDM400P board
Hello Asterisk Community! There have been some complaints made by those customers that purchased the TDM400P board and it didn't work properly in their boxes. Digium promised to swap such boards for the new - revised version and will keep the promise. However since we were backordered we're currently shipping boards to the customers that paid and are waiting. But if you want to receive your new board *very soon* you can call Greg or Malcolm at 256-428-6262 and ask them to ship you the board along with the new orders. If you're not in a hurry we will proceed soon with the list of people that reported the problems with their TDM400Ps before. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'
comment out register = user:[EMAIL PROTECTED] from sip.conf Martin On Sat, 6 Sep 2003, fredrik chabot wrote: Hello, Is there any way to get rid of this message. NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.123.110' There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that this message was gone for about 2 day's. Its back however. [100] type=friend secret= host=192.168.123.110 username=100 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [101] type=friend secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ___ 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] Bug in my head or bug in the code?
What does this step show on the CLI ?
exten = 1,1,SetVar(FOO=123**)
exten = 1,2,SetVar(CHECK=${FOO:-1:1})
? If you're going to see CHECK=* then there is a bug in = operator ...
Martin
On Fri, 5 Sep 2003, John Todd wrote:
I am having Yet Another Regular Expression problem, but this one
might not be my fault, or at least it might not be obviously my
fault. :-)
exten = 2212,1,SetVar(FOO=123456**)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
This script continues with a value of 0 in BAR.
Similarly, none of the following changes made a difference in that
result, which is expected since the * is not listed in
README.variables as a character that must be escaped:
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
I have also tried setting the variable ${BAZ}=* and then using that
in my comparison, with the same unexpected results.
Oddly enough, this almost-identical example below has different, but
normal, results: BAR=1
exten = 2212,1,SetVar(FOO=123456##)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = #])
What gives? Am I colliding with a problem that is the result of the
* character being used in expr evaluations and somehow not being
handled correctly, or am I simply not implementing the syntax
correctly?
JT
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Re: [Asterisk-Users] Regular expression matching for : - examples needed
Examples I'd like to see:
1)
${FOO} contains 12345#
${HASH} contains #
something like this:
exten = 123,1,Gotoif($[${FOO} : 12345#]?2|102)
If ${FOO} contains the contents of ${HASH} anywhere, go to 2. If not, goto 102
exten= 123,1,GotoIf($[...???...]?2|102)
1.1)
If the last digit of ${FOO} is ${HASH}, then goto 2. If not, goto 102.
exten = 123,1,GotoIf($[...???...]?2|102)
exten = 123,1,GotoIf($[${FOO:-1:1} = ${HASH}]?2|102)
assuming ${HASH} is one digit ...
Martin
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Re: [Asterisk-Users] The sounds of silence: silent soundfiles available
You could use ResponseTimeout together with Background instead of playing silence files. Martin On Thu, 4 Sep 2003, John Todd wrote: As has been noted before on this list, the Wait() application does not listen for keystrokes from users. Many of you, like me, have looping Background(), Wait(), and Goto() application priority chains that prompt users to enter some data, and then repeat the instructions if no keys are pressed. The problem of course is if the user doesn't start pressing keys during the Background() call and delays until the Wait() application is called, those keys are lost. I had solved this some time back by creating a few random length files of silence, that would replace Wait() routines in some circumstances. I have finally created a formal measured group of files, each with 1-10 seconds of silence, and put them in my sounds directory for public consumption. Not a big deal for most of you to create these files yourselves, but perhaps a minor pain that hopefully I've removed for some people who don't have sound tools handy. http://www.loligo.com/asterisk/sounds/silence/ JT ___ 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] call parking -- what was the key combination?
It's defined in /etc/asterisk/parking.conf and set by deafult as 700 Martin On Fri, 5 Sep 2003, Dave Alan Caruana wrote: what i'm asking is what is the key sequence you have to dial for the transfer .. it was something like *7# if I remember well, I know I had it working, but the client lost the paper I wrote it on for him, and I can't trace the email I got it from! cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 05, 2003 3:11 PM Subject: Re: [Asterisk-Users] call parking -- what was the key combination? To park a call you simply transfer the call into extension 700 (this is the default and can be changed).. To get the call back you just dial the parked location.. If you are using an IP phone this is a problem becasue it will not tell you the location of the parked call so you will not know where to collect it from.. hi great gurus of asterisk :) could somebody remind me the key combination to send a call into the parking queue ? while you're at it, are there any other key combinations I should know?? eg. put a call on hold etc. thanks Dave ___ 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] X100P in Spain Busy Detect
If you have 0.4 ms silence every 3 cycles then try to uncommnet BUSYDETECT_TONEONLY in asterisk/Makefile and recompile. regards Martin On Fri, 5 Sep 2003, Norberto Garcia Prieto wrote: Martin Pycko wrote: What's the Spain busy tone ? x ms tone, y ms of silence etc ... If I remember correctly, 0.2 ms on 0.2 ms off repeated. All tones are 425 Hz, -10dBm It may also add 0.4ms off after every 3 on/off cycles --- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de Indra. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail ___ 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] IAX2 ports usage
RTP ports are not applying to IAX/IAX2. Martin On Thu, 4 Sep 2003, WipeOut . wrote: Yes, The RTP ports in * are configurable in rtp.conf.. The default is 1 - 2 Later HI! but when making iax2 calls, a packet monitor would only reveal this UDP port. (Between two * servers) ?? 4569 proto: U ( I would assume even the RTP headers get enclosed by UDP, so there should have been more UDP port variants. Not the case when monitored.) I've got these in my rtp.conf rtpstart=1 rtpend=2 Does it mean RTP use the above udp port range ?( 1~2). denzel - Original Message - From: Wade J. Weppler To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 8:42 AM Subject: RE: [Asterisk-Users] IAX2 ports usage The RDP packets need to be dealt with as well. They are specified in rtp.conf -wade -Original Message- From: denzel-infotechs [mailto:[EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX2 ports usage hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2. DenZel. -- __ 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
