Re: [asterisk-users] SSH connection hangs on logout?
On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote: > On Thu, 24 Aug 2006, Jeremy McNamara wrote: > > >Rushowr wrote: > >>Hey all, I have an interesting issue that just recently started when I > >>grabbed a copy of the trunk about a week ago and compiled it. Ever since > >>that compile, if I start Asterisk (disconnected terminal, using > >>safe_asterisk to launch) and then continue on about my work with it, when > >>I > >>disconnect my SSH terminal (using latest version of PuTTY) the session no > >>longer closes it just hangs. I've even changed the Putty setting to close > >>the window even on unclean exit but it still hangs the connection... I had > >>something similar once with Zabbix a while back, but never Asterisk. > >> > >>Anyone else experience this? > > > >Start asterisk using safe_asterisk or via asterisk -f > > > >I prefer the safe_asterisk shell script, since if asterisk seg faults, > >there is a good chance asterisk will get automatically restarted. > > > >Jeremy McNamara > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >asterisk-users mailing list > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > You may need to redirect stdin, stdout, stderr like: > > run_asterisk\ > 0 1>/dev/null\ > 2>/dev/null\ > & > In other words: A plain 'asterisk' (without '-c' and such) that daemonizes and does exactly that for you, among others. Asterisk is a daemon, rather than an interactive program. Thus its handling for SIGHUP is to re-read configuration rather than detach from the terminal. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.4.3 Released!
Hello everyone, I have released AstLinux 0.4.3: http://sourceforge.net/projects/astlinux/ For all of those that have been waiting to switch to 0.4.x, this is your chance. The few remaining problems with uclibc have been fixed (i.e. voicemail timezones and voicemail -> email via MSMTP). Don't forget to peek around in SVN for all kinds of goodies. Especially trunk - the Gumstix is now a direct target for builds. That's right, build AstLinux for a Gumstix just as easily as a Soekris! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SSH connection hangs on logout?
On Thu, 24 Aug 2006, Jeremy McNamara wrote: Rushowr wrote: Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my SSH terminal (using latest version of PuTTY) the session no longer closes it just hangs. I've even changed the Putty setting to close the window even on unclean exit but it still hangs the connection... I had something similar once with Zabbix a while back, but never Asterisk. Anyone else experience this? Start asterisk using safe_asterisk or via asterisk -f I prefer the safe_asterisk shell script, since if asterisk seg faults, there is a good chance asterisk will get automatically restarted. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You may need to redirect stdin, stdout, stderr like: run_asterisk\ 0/dev/null\ 2>/dev/null\ & Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Annoying Bristuff
HiThe newest bristuff didn't change anything. Still the same. I was wondering if this is happening only to me or not. Does anyone has the same problem? Maybe I am messing something when loading the modules. Does anyone have any other tips.Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia E60/61/70 and SIP
On 08/24/06 09:02 El Flynn said the following: Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. SIP works well with asterisk, with some caveats: 1. you need qualify set as the wifi radio on the phone sucks big oranges 2. the phone routinely loses IP connectivity, leading to reg failures 3. when two simultaneous calls, GSM and SIP, come in the phone hangs more often than not 4. be prepared to reboot constantly for simple config changes on the phone. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Nokia E60/61/70 and SIP
We are using some E61 and E70's with asterisk. Only problem we have at this moment is that we are unable to use a password for the authentication. I haven't found out yet why this isn't working. They are working good, but I would like to see some small things changed in future firmware versions (like being able to select multiple WLAN points (Access groups) instead of just one. Rolph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of El Flynn Sent: donderdag 24 augustus 2006 3:03 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia E60/61/70 and SIP Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SIP client works. Link: http://reviews.cnet.com/4520-6454_7-6358771-1.html?tag=txt Thanks, El Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Extensions -- Comments?
Douglas Garstang wrote: It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. It doesn't seem like Asterisk is good enough for you Doug. Switch to one of the competitors' products. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia E60/61/70 and SIP
Hi, have you looked here http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html thanks atik On 8/24/06, El Flynn <[EMAIL PROTECTED]> wrote: Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SIP client works. Link: http://reviews.cnet.com/4520-6454_7-6358771-1.html?tag=txt Thanks, El Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SSH connection hangs on logout?
Rushowr wrote: Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my SSH terminal (using latest version of PuTTY) the session no longer closes it just hangs. I've even changed the Putty setting to close the window even on unclean exit but it still hangs the connection... I had something similar once with Zabbix a while back, but never Asterisk. Anyone else experience this? Start asterisk using safe_asterisk or via asterisk -f I prefer the safe_asterisk shell script, since if asterisk seg faults, there is a good chance asterisk will get automatically restarted. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SSH connection hangs on logout?
Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my SSH terminal (using latest version of PuTTY) the session no longer closes it just hangs. I've even changed the Putty setting to close the window even on unclean exit but it still hangs the connection... I had something similar once with Zabbix a while back, but never Asterisk. Anyone else experience this? SKM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] NAT problems
Try changing the configuration on your PAP2 linksys, more precisly the part where is the NAT parameters, try changing the options from NO to YES. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de andrutto Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] NAT problems Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting "488 Not Acceptable Here" and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. Cheers Andrutto -- Zostan Dziewczyna Lata! >>> http://link.interia.pl/f1997 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: problems with wevbmail
I could fix it. The problem was permissions on the directory /var/spool/asterisk/voicemail. Thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steven Enviado el: Miércoles, 23 de Agosto de 2006 08:01 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Re: problems with wevbmail Try running apache as the asterisk user instead of "apache" My assumption is that "apache" or your apache user does not have access to the voicemail folders. -- -- Steven http://www.glimasoutheast.org "Sergio R. D'Ippolito" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]... I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i can’t see the messages on any folder. Thanks, Sergio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?
Giorgio Incantalupo wrote: Hi, I have an asterisk box with a sangoma a102 (two PRI ports). Is is possible to connect port A to port B in order to use port B as a simulation of a telco PRI line? If yes, is there a special cable needed? How can I configure the card and zaptel.conf? Yes. You'll need a T1 crossover cable to do it. Google for which pins to swap. Configure one port as pri_net (acts as a central office switch) and the other port as pri_cpe (acts as a pbx). See the sample configs for other parameters (including /etc/zaptel.conf timing parameters). Your zapata.conf entries will look something like these: resetinterval=never ; gets rid of the many restart messages context=pri-in signalling=pri_net switchtype=national pridialplan=unknown channel=>1-23 context=pri-out switchtype=national signalling=pri_cpe pridailplan=unknown group=7 channel=>25-47 And, /etc/zaptel.conf something like this: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nokia E60/61/70 and SIP
Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SIP client works. Link: http://reviews.cnet.com/4520-6454_7-6358771-1.html?tag=txt Thanks, El Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About IVR and Oracle
On 8/23/06, Infobox Peru <[EMAIL PROTECTED]> wrote: maybe you could make it with PHP and its driver for Oracle. For PHP have a look here : http://phpagi.sourceforge.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SJPhone and Asterisk over H323
Hello all, I'm using Asterisk h323 (default/NuPhone) with some success with SJPhone. I say some success because while I'm able to receive audio from Asterisk, I seem unable to send audio to it... Any suggestions? Anybody managed to get this to work? Thanks, Matt King Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11
I upgraded to 1.2.11 and now I see two behaviors different than the existent in 1.2.10: 1.- I get 183 Session Progress instead of 180 Ringing. 2.- If I have three extensions, A, B and C. A using codec X, B using also codec X and C using codec Y. If C dials to B and A tries to pick up the call (using *8#), it start getting an endless output of: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64) (in this case, C was using GSM, B and A, G729). I tried this making all the combinations between A, B and C calling each other, and I only get the problem when the picked conversation needs to be transcoded (it means, if A calls to C and B pick it up, it worked fine). For some reason, I guess somebody initializes a variable as SLINEAR (64) in all cases. The result is that it's impossible to pick up the calls!!! Has anybody experienced this issue? Is this a bug in 1.2.11? I looked through Mantis, but didn't find a clue. Thanks a lot for your attention. -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk IAXmodem HylaFax?
Tzafrir Cohen wrote: On Wed, Aug 23, 2006 at 03:41:22PM +1000, Warrick Zedi wrote: Tzafrir, When last did you look at AsterFax? What do you believe is required to set it up? In what way are there "wheel reinventings" in either HylaFax or AsterFax? Tzafrir Cohen wrote: On Wed, Aug 23, 2006 at 08:28:51AM +1000, Warrick Zedi wrote: If you're looking for alternatives to Zetafax why not look at AsterFax (http://asterfax.sourceforge.net)? Your clients can use their existing e-mail client to send faxes. Running OpenOffice on the server to render OpenOffice/MS-Office documents automatically is something that will hopefully work well. If both copies have exactly the same fonts. And exactly the same definitions. And there are no other compatibility bugs. Defining the fax number in an email message is not exactly a natural operation, IMHO. I rather do the fax rendering on the client side. That way, if the client tries to send out a wrongly-rendered file, you automatically get it. Point taken. 1.0 has just been released. As AsterFax becomes more widely adopted we'll get more feedback and will have to watch out for compatibility issues. I believe we'll be able to address those if they arise. I suppose you've got a point about the fax number in the email message but then a fax number really is just an address so instead of putting a [EMAIL PROTECTED] you're putting a [EMAIL PROTECTED] to com. But you have just given me an idea. Maybe we should provide some sort of name resolution feature where you could put [EMAIL PROTECTED] and AsterFax can lookup the fax number. Hmmm, sounds like a good idea, I'll have to think about it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About IVR and Oracle
maybe you could make it with PHP and its driver for Oracle. Daniel Pizarro www.infobox-peru.com-- Forwarded message --From: Moises Silva < [EMAIL PROTECTED]>Date: 23-ago-2006 17:12Subject: Re: [asterisk-users] About IVR and OracleTo: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com>Sure is possible. Look into google 'asterisk agi fastagi'.RegardsOn 8/23/06, Javier Lara Sanchez <[EMAIL PROTECTED] > wrote:> Dear All, I need to buid an IVR that could make a request to a data base (oracle) in a> remote host.>>> > The idea is that an user dial a extension with 2 options and one of them> ask for a data (in the case a date). This data is the field that the data> base needs to find the information that the user are looking for.. Somebody know if this is posible or have any idea where can I find> information about this? Thank>> Regard>> Javier> > ___> --Bandwidth and Colocation provided by Easynews.com -- >> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>--"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audion on Sangoma
Hi List, I have an A200 with echo can. 2-FXO and 2 FXS. Today I went and upgraded asterisk, zaptel and libpri. I ran the sangoma util to patch asterisk. When I started up asterisk ZAP1 worked like a charm. However ZAP2 has been acting up. I only get one way audio on it. The person that I call can hear me however I can not hear them at all. I tired switching around the lines but to no avail. It seems that only zap2 is giving the problems. Anyone have any suggestions ? Can it be that ZAP2 just crapped out today or does it have to do with the upgrade. I also want to mention that I didnt use the system all day so I dont know if it was working earlier (before I upgraded asterisk) or not. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] howto install asterisk on freebsd release 4.11
Hi There, Is there anybody who installed asterisk on freebsd 4.11 release ? I was not succesful. please guide me. I updated the ports and I installed the lib using ports but when I try to install zaptel it says cannot load it for release before than 5 I couldn't install the asterisk from ports also because it ask for zaptel. I don't need zaptel because I will not use hardware by asterisk. Please also let me know how to cancel the requirment of zaptel during insalling asterisk from ports ? Below please find my tries; Thanks Mansour Safaie dedi513# cd /usr/ports/misc/zaptel/ dedi513# make install clean ===> zaptel-1.0 does not build on FreeBSD \< 5.x. *** Error code 1 Stop in /usr/ports/misc/zaptel. dedi513# cd /usr/ports/net/asterisk dedi513# make install clean ===> asterisk-1.2.9.1_1 depends on executable in : mpg123 - found ===> asterisk-1.2.9.1_1 depends on package: libpri>=1.2.0 - found ===> asterisk-1.2.9.1_1 depends on file: /usr/local/include/zaptel.h - not found ===>Verifying install for /usr/local/include/zaptel.h in /usr/ports/misc/zaptel ===> zaptel-1.0 does not build on FreeBSD \< 5.x. *** Error code 1 Stop in /usr/ports/misc/zaptel. *** Error code 1 Stop in /usr/ports/net/asterisk. dedi513# cd /usr/local/asterisk-install/asterisk/asterisk-1.2.10 dedi513# ls .cleancount callerid.c jitterbuf.c .lastclean cdr jitterbuf.h .versioncdr.c keys BUGSchannel.c loader.c CHANGES channels logger.c COPYING chanvars.c manager.c CREDITS cli.c md5.c ChangeLog codecs mkpkgconfig HARDWAREcoef_in.h muted.c LICENSE coef_out.h muted.conf.sample Makefileconfig.c netsock.c README configs pbx README.fpm contrib pbx.c SECURITYcryptostub.cplc.c UPGRADE.txt cygwin poll.c acl.c db.c privacy.c aescrypt.c db1-ast redhat aeskey.cdevicestate.c res aesopt.hdlfcn.c rtp.c aestab.cdns.c sample.call agi dnsmgr.csay.c alaw.c doc sched.c app.c dsp.c slinfactory.c appsecdisa.hsounds ast_expr2.c editline sounds.txt ast_expr2.flenum.c srv.c ast_expr2.h file.c stdtime ast_expr2.y formats strcompat.c ast_expr2f.cframe.c tdd.c asterisk.8 fskmodem.c term.c asterisk.c funcs translate.c asterisk.sgml image.c ulaw.c astmm.c images utils autoservice.c include utils.c build_tools indications.c buildinfo.c io.c dedi513# make install clean "Makefile", line 28: Missing dependency operator "Makefile", line 32: Need an operator "Makefile", line 35: Need an operator Error expanding embedded variable. dedi513# make "Makefile", line 28: Missing dependency operator "Makefile", line 32: Need an operator "Makefile", line 35: Need an operator Error expanding embedded variable. dedi513# ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Registering IP Phone To Asterisk
I don't think you can use the template of another brand with your Fanvil. You must configure the phone manually. First time I ear about FAnvil IP phone so I cannot help you David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : 23 août 2006 11:44 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Registering IP Phone To Asterisk What is the process to get an IP phone registered to Asterisk? I bought an Asterisk with a GUI and it has templates for devices such as sipura, cisco, and xten but I am using a Fanvil IP phone. How do I load the template for my IP phone into astrisk so that it can work? Thanks Wyatt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
I tested the trunk two days ago and I agree that the presence feature is broken. I don't know if it's an error or if it's volunteer. I will post a bug report on the tracker and we will see. One thing is sure; hints are working well on 1.2.10 and not in the trunk. Is this because they did some change to "friend" I don't know. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Watkins, Bradley Envoyé : 23 août 2006 10:41 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] Hint extension issue - bug? I may have to eat my words, then. This is the case with trunk, and I can't recall the last time I built a 1.2.x system. I could have sworn that behavior didn't change, but I've been wrong before. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon Sent: Wednesday, August 23, 2006 10:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Hint extension issue - bug? On 1.2.10, presence is working very well using friend. The state is refreshing successfully. There is probably antoher problem with your installation cause I'm using hint with friend since 1 years in all my production system. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 23 août 2006 08:50 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Hint extension issue - bug? I'm using asterisk 1.2.10 David Gagnon wrote: >Are you having this problem with the trunk? > > > >-Message d'origine- >De : [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez >Envoyé : 22 août 2006 18:23 >À : Asterisk Developers Mailing List; Asterisk Users Mailing List - >Non-Commercial Discussion Objet : [asterisk-users] Hint extension issue >- bug? > >Hi, I'm using the hint extension to monitoring the status of some >extensions. If the extension is defined as a friend, the monitoring >doesn't work any more. It only work if I define it as a "peer". Is that >right ? I mean, I supposed that an extension defined as a friend should >have all the functionality of "user" and "peer" types. Is this >documented somewhere? How can I know the status of an extension of type >friend? I hope someone could bring me some light about this issue. >Thanks in advance. > >Lucas Alvarez > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About IVR and Oracle
Sure is possible. Look into google 'asterisk agi fastagi'. Regards On 8/23/06, Javier Lara Sanchez <[EMAIL PROTECTED]> wrote: Dear All, I need to buid an IVR that could make a request to a data base (oracle) in a remote host. The idea is that an user dial a extension with 2 options and one of them ask for a data (in the case a date). This data is the field that the data base needs to find the information that the user are looking for.. Somebody know if this is posible or have any idea where can I find information about this? Thank Regard Javier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About IVR and Oracle
Dear All, I need to buid an IVR that could make a request to a data base (oracle) in a remote host. The idea is that an user dial a extension with 2 options and one of them ask for a data (in the case a date). This data is the field that the data base needs to find the information that the user are looking for.. Somebody know if this is posible or have any idea where can I find information about this? Thank Regard Javier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB GSM gateway for Asterisk?
You can find cheap gsm (+/- 150$) gateways too, although the cheap ones will require a additional pstn card. (expensive ones could do sip) Zoa. Jay Milk wrote: There's been some (futile?) effort a while back attempting to get a Bluetooth capable phone integrated into asterisk as a channel. The idea, of course, was to make it possible to have asterisk utilize a cellular connection for backup, calls on free nights/weekends, or free in-network minutes. How about skipping the phone entirely? I just found this -- http://www.falcom.de/?id=283 It's an integrated quad-band GSM engine with a USB connection. Claims to be able to do data, fax, voice, SMS/MMS etc, even EDGE speeds. Prices I found are 170 Euros or US$225. Considering the alternative ($20 USB dongle, $180 BT capable phone), this seems like a very competitive option. I'd expect it to be in Falcom's best interest to support development efforts as it would open the asterisk market to them. Anyone up for creating a bounty-page for this? Voice traffic would be the first priority, but SMS would certainly be a desired option. Personally, I wouldn't mind being able to utilize EDGE as an emergency backup for my network, allowing at least mail traffic and such. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] isdn30 uk setup problem
Hi
Can anyone help with my following problem connecting asterisk to a new
provisioned isdn30e line.
At long last I have had BT install our new isdn 30 (I421) line for our
asterisk server after 3 months of waiting.
Before this we have used asterisk with a tdm400 and some analogue lines.
I have installed a digium TDM 205 e1 card with port 1 connected to the
isdn30 box via a rj45 network lead. (wired like a normal patch lead NOT
crossover). Is this correct ?. I have no red warning lights just green ok
lights.
Asterisk,libpri and zaptel are the latest ones from asterisk.org.
On trying to dial using the isdn30 im getting the following error
Circuit/Channel congestion
-- Executing Set("SIP/602-9989", "GROUP(901392275533)=OUTBOUND_GROUP") in
new stack
-- Executing GotoIf("SIP/602-9989", "0?5") in new stack
-- Executing Set("SIP/602-9989", "GROUP(602)=OUTBOUND_GROUP") in new stack
-- Executing Dial("SIP/602-9989", "zap/g1/14101392275533") in new stack
Aug 23 16:26:43 NOTICE[6823]: app_dial.c:1040 dial_exec_full: Unable to
create channel of type 'zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/602-9989' status is 'CONGESTION'
ztcfg - shows
PAN 1: CCS/HDB3 Build-out: 133-266 feet (DSX-1)
SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 0
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 1
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 2
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)
but this does show in the /var/log/messages
Aug 23 16:32:21 WARNING[6778]: chan_zap.c:6347 handle_init_event: Detected
alarm on channel 28: Recovering
Aug 23 16:32:21 WARNING[6778]: chan_zap.c:1435 zt_disable_ec: Unable to
disable echo cancellation on channel 28
FOR each channel and then
Aug 23 16:32:26 NOTICE[6778]: chan_zap.c:6340 handle_init_event: Alarm
cleared on channel 28
FOR each channel
Can anyone help as have I missed somthing simple.
Thanks
Nick
Config files:-
my zaptel.cong has the following
loadzone=uk
defaultzone=uk
span=1,1,1,ccs,hdb3,crc4
span=2,0,0,esf,b8zs
bchan=1-15
dchan=16
bchan=17-31
and zapata.conf has
[channels]
language=en
switchtype=euroisdn
signalling=pri_cpe
callwaiting=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
;UK CLID BT
usecallerid=yes
immediate=yes ;take line on first ring
callerid=asreceived ; propagate cid as received from bt
cidsignalling=v23
cidstart=polarity
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;400
rxgain=0.0 ;14.5
txgain=0.0 ;2
busydetect=yes
group=1
callgroup=1
pickupgroup=1
musiconhold=default
;ISDN 30e Uk SALES LINE Card 1 Line 1
switchtype=euroisdn
signalling=pri_cpe
group=1
context = incoming
channel => 1-15
channel => 17-31
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Re: [asterisk-users] Annoying Bristuff
HiThanks for your reply.I will check it first thing in a morning and of course will let you know about results.Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set externip in sip.conf automatically?
John Marvin wrote:
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for
Asterisk to access a Linux environment variable containing the IP
information in the form of "${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Actually, I don't think you read his response carefully enough. He was
giving you a method of doing it automatically.
But first, lets dismiss the environment variable solution. I haven't
played with using environment variables in Asterisk, so I can't help you
there. But I do know about environment variables in general, and you
cannot use them to solve your problem. Environment variables are not
"global", i.e. if you change one it does not effect the value in all
currently running programs. The "environment" (all of the environment
variables and their values) is inherited from the parent process (it is
passed in by the kernel on the new processes stack when the process
first starts). After that, only the process itself can change its own
values in order to pass on a changed value to a child (but again, only
when that child process is started, i.e. a parent cannot affect the
environment of an already running child process). In summary, you can't
change the values of Asterisk's environment variables after Asterisk has
already started. The values that Asterisk sees are the values that it
inherited from its parent process, i.e. most likely the rc scripts that
started Asterisk when you first booted the machine.
Now, back to the solution proposed by Brad. He was in effect proposing
that you dynamically change sip.conf. However, parsing a rewriting
sip.conf automatically is kind of ugly, but luckily Asterisk supports
#include. So, he suggested that you can periodically generate a file
with a single line in it, i.e. "externip=xx.xx.xx.xx" and then use
#include in your sip.conf to include it (i.e. sip.conf doesn't have to
ever change). The final part of the solution is to make Asterisk reread
sip.conf (and the included dynamically created file at the same time).
You can do that with:
asterisk -rx "sip reload"
which you can put into the same cron job that you currently are using to
refresh /etc/myip.
John
John you are quite correct about Brad's solution and I have to thank you
both. I thought again about Brad's post last night in bed - where I do
my best work .
The #include route is indeed a way to go.
In the meantime, I've found more about externhost which was implemented
in Asterisk 1.20+.
I put a line in sip.conf of
externhost=myhost.dyndns.org
with a refresh of 6 hours against the default of 10 seconds ala
externrefresh=21600
This seems to be working nicely but, if there are problems, I'll change
to #include.
You are also correct that environmental variables will not refresh until
the Asterisk shell is reloaded - hopefully not often!
It would still be nice to be able to access shell variables from within
the CLI and particularly frustrating since the "Asterisk - Future of
Telephony" mentions a specific method on page 92. Perhaps they got the
syntax wrong. I certainly would like a better connection between the
shell and CLI.
Thanks for your explanation and I apologize for not realizing that
Brad's suggestion was spot on.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] NoCDR()
Everybody, What is the proper usage of NoCDR()? I keep getting the following warning about lacks end: Aug 23 16:34:32 WARNING[23822]: cdr.c:443 ast_cdr_free: CDR on channel 'Local/[EMAIL PROTECTED],1' not posted Aug 23 16:34:32 WARNING[23822]: cdr.c:445 ast_cdr_free: CDR on channel 'Local/[EMAIL PROTECTED],1' lacks end Searching the archives don't reveal any answers and neither does the Wiki. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set externip in sip.conf automatically?
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for Asterisk
to access a Linux environment variable containing the IP information in
the form of "${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Actually, I don't think you read his response carefully enough. He was
giving you a method of doing it automatically.
But first, lets dismiss the environment variable solution. I haven't
played with using environment variables in Asterisk, so I can't help you
there. But I do know about environment variables in general, and you
cannot use them to solve your problem. Environment variables are not
"global", i.e. if you change one it does not effect the value in all
currently running programs. The "environment" (all of the environment
variables and their values) is inherited from the parent process (it is
passed in by the kernel on the new processes stack when the process
first starts). After that, only the process itself can change its own
values in order to pass on a changed value to a child (but again, only
when that child process is started, i.e. a parent cannot affect the
environment of an already running child process). In summary, you can't
change the values of Asterisk's environment variables after Asterisk has
already started. The values that Asterisk sees are the values that it
inherited from its parent process, i.e. most likely the rc scripts that
started Asterisk when you first booted the machine.
Now, back to the solution proposed by Brad. He was in effect proposing
that you dynamically change sip.conf. However, parsing a rewriting
sip.conf automatically is kind of ugly, but luckily Asterisk supports
#include. So, he suggested that you can periodically generate a file
with a single line in it, i.e. "externip=xx.xx.xx.xx" and then use
#include in your sip.conf to include it (i.e. sip.conf doesn't have to
ever change). The final part of the solution is to make Asterisk reread
sip.conf (and the included dynamically created file at the same time).
You can do that with:
asterisk -rx "sip reload"
which you can put into the same cron job that you currently are using to
refresh /etc/myip.
John
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Re: [asterisk-users] How to set externip in sip.conf automatically?
Thank you Greg and RR.
externhost=myhost.dyndns.org works perfectly so figuring out how to
access a shell variable from within the CLI is no longer necessary -
although it would be nice to know!
externhost works in 1.20 onwards.
Thanks for finding the solution.
Larry
Greg Delgado wrote:
The easiest way is to register for free dynamic DNS
service at www.dyndns.com. Then use externhost=
instead of externip= in sip.conf . If you are using a
Linksys router like the WRT54G, it already has a
dyndns client which will update the dyndns servers
with your ip address everytime it changes.
Greg
--- Larry Alkoff <[EMAIL PROTECTED]> wrote:
As stated in the original post, when I entter the IP
with an editor
directly into sip.conf calls work just fine but I am
looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is
possible for Asterisk
to access a Linux environment variable containing
the IP information in
the form of "${ENV{variable}}.
It doesn't seem to work. I am asking how to make it
work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't
you set it up so the
file itself says: externip=xx.xx.xx.xx and then
do a #include in
sip.conf for the /etc/myip file? I believe you'll
have to do a sip
reload either way (which can obviously be part of
your cron job) if
you're not already, but that should do what you're
looking to do.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of Larry
Alkoff
Sent: Tuesday, August 22, 2006 9:34 PM
To: Asterisk-users; Austin-asterisk-users
Subject: [asterisk-users] How to set externip in
sip.conf automatically?
I need to give Asterisk access to my external IP
address to prevent
the NAT problem where caller cannot hear the
callee's voice.
According to Asterisk - The Future of Telephony
page 92 Environment
Variables:
"Environment variables are a way of accessing
Unix environment
variables from within Asterisk. They are
referenced in the form of
${ENV{var}}
where var is the Unix environment variable you
wish to reference."
My external IP is placed each night in a file call
/etc/myip and placed
in the $MYIP variable by /etc/bashrc when an shell
is loaded.
So I have /etc/myip refreshed each night in a cron
job and when a shell
is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret
the book?
I say neither seems to work because When I hard
code
externip=69.91.84.176
there are no NAT problems but when I try to access
the $MYIP variable
either of the ways above NAT prevents me hearing
the callee's voice.
I have tried but not found a way to directly
access the contents of MYIP
to the console using the CLI. Is there a way to
see or set _any_ Linux
enviromnent variable using the CLI? More
generally, how do I access the
Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't
know an easy way to
automatically write the value into sip.conf
programatically.
I could have just said "how do I do this" but
wanted to show that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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The contents of this e-mail are intended for the
named addressee only. It contains information that
may be confidential. Unless you are the named
addressee or an authorized designee, you may not
copy or use it, or disclose it to anyone else. If
you received it in error please notify us
immediately and then destroy it.
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Using Thunderbird on Linux
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[asterisk-users] MySQL undefined symbol: __pure_virtual
Hey guys, I'm getting the following message when I start asterisk: Aug 23 13:42:40 WARNING[29258] loader.c: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __pure_virtual Aug 23 13:42:40 WARNING[29258] loader.c: Loading module res_config_mysql.so failed! I don't know how it happened because everything was working fine before, now it doesnt. rpm -qa | grep -i mysql php-mysql-4.3.9-3.1 perl-DBD-MySQL-2.9004-3.1 MySQL-client-standard-5.0.18-0.rhel4 mysql-4.1.7-4.RHEL4.1 I'm running MySQL 5.1.11-beta as a binary in /usr/local/mysql. I'm hoping this is not a problem. I have recompiled the asterisk-addon packages (make clean ; make ; make install) to see if this would help but it didnt. Any ideas on how I can fix this error and get my Asterisk to run? Red Hat Enterprise Linux 4 Note: I have an exact copy of this server (other node in the cluster) with the same packages installed and using the exact same configuration file and it works great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to start special tone
Can anyone tell me where this is coming from? I can’t seem to find any information on it anywhere. I don’t believe I’m using “special tones” anywhere. Any ideas? Aug 23 14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special tone on 15 _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI initiate call probs
On Tue, 22 Aug 2006 16:55:37 +0200, Niklas Larsson wrote: > I'm using AMI to initiate a call, first calling the agent and when > he picks up, the call is placed to the customer. The prob is if the > user rejects the call (or they don't have cw...), the call is still > placed to the customer... > > I havn't found a variable that tells me that the first leg is still > up. I removed the Local part and called the extension directly, and the problem went away... > This is what i send to AMI: > > Action: Originate > Channel: Local/[EMAIL PROTECTED] Channel: Sip/8542 > Context: from-turbo > Async: yes > Variable: ext_turbo=8542 > Exten: 0703123456 > Priority: 1 > Timeout: 3 > Callerid: "Turbo" /Niklas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
[EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim Jim, Did you ever find a solution to this, other then buying another TDM? I've had a query from someone with the same issue. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Annoying Bristuff
On Wed, Aug 23, 2006 at 09:35:17PM +0200, Andrew Nowrot wrote: > Hi > > I am trying to set up * box with the ISDN hfc-s cards. One in NT mode and > two in TE. I am using the bristuff-0.3.0-PRE-1r.tar.gz . The installation > went well, but soon after the zaphfc was loaded I started to receive these > message in kernlog: > > Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC frame or bad CRC > received (framelen = 6, stat = 0xff, card = 1). > Aug 23 21:00:09 asterisk kernel: zaphfc: empty HDLC frame or bad CRC Hmmm give bristuff 0.3.0-PRE1s a shot. Reading from top of the CHANGES file: 0.3.0-PRE-1s: - added "FASTBUSYONBUSY" Makefile option to libpri - fixed "BAD CRC" error on layer 1 activation in TE mode /me off to merge bristuff 1s in the official deb... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] client socket to asterisk manager gets disconnected
I have a test application, what it does is just connect to the asterisk manager, and listen for events. I also set the connection to receive on user, call and agent events. I Noticed that everytime the queue is empty and a caller joins in, asterisk tends to throw too many queuememberstatus events, overwhelming the connection and therefore closes it abruptly. I also got the same result when I used telnet to connect to asterisk, and then make a call which is then forwarded to an empty queue. I'm using version 1.2.7.1 and even if the queue has its eventwhencalled set to no, the problem still persists. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3COM NBX and Digium Cards...
> Message: 6 > Date: Wed, 23 Aug 2006 13:13:17 -0500 > From: Carlos Chavez <[EMAIL PROTECTED]> > Subject: [asterisk-users] 3COM NBX and Digium Cards... > To: Asterisk > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="utf-8" > > I have a customer that has returned two cards, a TE210P and a TE110P > because they are no longer working. Both cards were connected to an > 3COM NBX system but not to the same one. On the TE210P only the port > that was connected to the NBX failed, the other works perfectly. > > The cards are on separate sites, one is on an IBM server, the other one > in a Dell. The only thing in common is that both cards were connected > to a NBX. Both Asterisk systems worked fine for about two weeks and > suddenly stopped working. Does anyone have any idea what could be > causing the failures? Is the NBX capable of doing something that can > damage the Digium cards? I have a TE205P card connected to a 3COM NBX system with no problems for the past 6 months. But as it is used for testing and demos only the amount of traffic is typically around 10 calls per day, and sometimes no calls are exchanged on the line between the Digium TE205P and the 3COM NBXfor days at a time. > -- > Carlos Chavez Prats > Director de TecnologÃa > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > -- next part -- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20060823/486aabc3/attachment-0001.pgp > > -- __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 incent Delporte wrote: > Hello > > I'm having a problem with the Linksys 3102: With incoming PSTN > calls, I can hear the caller through the X-Ten softphone, but he can't > hear me. The problem is worse with Sjphone and the GrandStream 100 > hardphone, as I get no sound in either direction. > > FWIW... > > - the SIP client, the PBX and the Linksys are all connected to a switch, > with no firewall anywhere > > - the only way I can get the Linksys to notify the PBX of an incoming > PSTN call is using the following settings: > > * PSTN Line > PSTN-To-VoIP Gateway Setup > PSTN Ring Thru Line 1 = yes > * User 1 > Call Forward Settings > Cfwd All Dest = fxo (where "fxo" is > the account also used in PSTN Line > Subscriber Information to register > with the PBX) > > Dial plans in either "Line 1" or "PSTN Line" don't make it. > > Could someone upload his configuration of the Linksys (File > Save as > file) so I can compare with what I have? > > Since both ends use G711u as their default codec and there's no firewall > between them, I suspect I'm totally wrong when it comes to configuring > the Linksys as a simple SIP gateway (no use for the FXS port at this > point). Possibly some routing issue. Here are mine (with UK regional settings/A-law). http://www.wellsted.org.uk/spa3102router.html for the router configuration as bridge and http://www.wellsted.org.uk/spa3102voice.html for my voice configuration with UK regionalisation (A-law, UK tones/cadences). - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBROyzPEtP/KMNOfRbAQLZrgf+NJse+LpdmROsaZCChmXbBMvLz3vMexMi IuZTOetaj6/XYwZUXmuHbMg5sqzK1fU70XmnEhUqtGVPyIVlSM1ZCmmh9bnpK26b 0V4E8T7ToTG1wG7QPahGKg4ly51UoTq3lKLCRxLvaPlEtxxAJjHMYTm9bHD+TufU 2zA7519UvC7C9UWrIKUFJZhcanzdI90VXYJj3mD8qzjIXdxAr1/jKiLHlw++3PLX TWlQLkGeuBM665LE4vCl4K0tmqHs9MO2f5E6+qCLgnF28pPpLVeShnQvSAEXV/p9 vHoCBzjMcUIvVHlLe/klGII6gOX7XEt13Ymm0XtJXdhnuhg5lDv9FQ== =5PE4 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: NAT problems
Strange?!? These three phones are using g726 (this codec is configured in sip.conf and in SIP ATA as well). -- Zostan Dziewczyna Lata! >>> http://link.interia.pl/f1997 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Annoying Bristuff
HiI am trying to set up * box with the ISDN hfc-s cards. One in NT mode and two in TE. I am using the bristuff-0.3.0-PRE-1r.tar.gz . The installation went well, but soon after the zaphfc was loaded I started to receive these message in kernlog:Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1). Aug 23 21:00:09 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 2).Aug 23 21:00:35 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). Aug 23 21:00:39 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1).Aug 23 21:01:01 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 2). Aug 23 21:01:09 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff, card = 1).Aug 23 21:01:27 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). Aug 23 21:01:39 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1).Aug 23 21:01:53 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). Aug 23 21:02:10 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1).Aug 23 21:02:19 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). Aug 23 21:02:45 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 2).Aug 23 21:03:04 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1). Aug 23 21:03:11 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2).Aug 23 21:03:34 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff, card = 1). Aug 23 21:03:37 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2).I know that there was a discussion on the list about this issue, but unfortunately it didn't point me anywhere further. >alternatively you can try this solution on debian: >4)Modules configuration for startup with zaphfc AND wcfxs on debian stable 3.1 sarge kernel >2.4.27-2-386. Here a configuration to fix this issue at boottime emacs /etc/modutils/zaptel>match it to: >post-install zaphfc /sbin/ztcfg>#post-install tor2 /sbin/ztcfg>#post-install wcusb /sbin/ztcfg>#post-install wcfxo /sbin/ztcfg>#post-install ztdynamic /sbin/ztcfg>#post-install ztd-eth /sbin/ztcfg >#post-install wct1xxp /sbin/ztcfg>#post-install wct4xxp /sbin/ztcfg>#post-install wcte11xp /sbin/ztcfg>alias wctdm wcfxs>#post-install torisa /sbin/ztcfg>#post-install wcfxs /sbin/ztcfg ># end of file /etc/modutils/zaptel >Update the /etc/modules.conf file with:>[EMAIL PROTECTED] update-modules >[EMAIL PROTECTED] emacs /etc/modules>add a line like the following at end of the file >zaphfc>and finaly reboot for testing>[EMAIL PROTECTED] rebootI tried to set up this like described above but it didn't help (I am still getting these messages). This messages sometimes cause my ISDN link to hang (I can receive the calls, but I am not able to make any outgoing calls). I am using Debian Sarge with custom 2.4.30 kernel.Does anyone have got any idea how can I make this to work.Please help I need it very badly.CheersAndrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT problems
andrutto wrote: Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting "488 Not Acceptable Here" and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. "488 Not Acceptable Here" is almost always a codec issue. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Direct to Voicemail
Aaron Daniel wrote: Since you're using the variables to decide what to do next (VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the Thank you very much, this took care of it. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan "or" matching
Glad I could help. I agree, these mailing lists are a life saver. I
personally have only been using Asterisk for about 5 months now, in fact
I have never even delt with any PBX's before (complete newbie) but
everyone here is very helpful and I am picking up a lot.
Kevin
David Cook wrote:
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those "Doh!" moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3="800", etc. rather than by 2's, (so it doesn't overlap
with 8XX area codes) and select my local loop as the "first pick".
dbc.
Kevin Smith wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten => _18XXNXX,1, NoOP()
exten => _18XXNXX,n,gotoif("${EXTEN}:2:2" = "00" | "${EXTEN}:2:2"
= "66" | "${EXTEN}:2:2" = "77" | "${EXTEN}:2:2" = "88")?TRUE:FALSE
exten => _18XXNXX,n(TRUE),Dial()
exten => _18XXNXX,n(FALSE), HangUp()
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[asterisk-users] NAT problems
Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting "488 Not Acceptable Here" and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. Cheers Andrutto -- Zostan Dziewczyna Lata! >>> http://link.interia.pl/f1997 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set externip in sip.conf automatically?
That's a very nice idea Greg. I'm not sure that my Asterisk 1.2 has the
externhost= function but it would solve my problem.
I have a dyndns.org account already that reports my externip.
Larry
Greg Delgado wrote:
The easiest way is to register for free dynamic DNS
service at www.dyndns.com. Then use externhost=
instead of externip= in sip.conf . If you are using a
Linksys router like the WRT54G, it already has a
dyndns client which will update the dyndns servers
with your ip address everytime it changes.
Greg
--- Larry Alkoff <[EMAIL PROTECTED]> wrote:
As stated in the original post, when I entter the IP
with an editor
directly into sip.conf calls work just fine but I am
looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is
possible for Asterisk
to access a Linux environment variable containing
the IP information in
the form of "${ENV{variable}}.
It doesn't seem to work. I am asking how to make it
work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't
you set it up so the
file itself says: externip=xx.xx.xx.xx and then
do a #include in
sip.conf for the /etc/myip file? I believe you'll
have to do a sip
reload either way (which can obviously be part of
your cron job) if
you're not already, but that should do what you're
looking to do.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of Larry
Alkoff
Sent: Tuesday, August 22, 2006 9:34 PM
To: Asterisk-users; Austin-asterisk-users
Subject: [asterisk-users] How to set externip in
sip.conf automatically?
I need to give Asterisk access to my external IP
address to prevent
the NAT problem where caller cannot hear the
callee's voice.
According to Asterisk - The Future of Telephony
page 92 Environment
Variables:
"Environment variables are a way of accessing
Unix environment
variables from within Asterisk. They are
referenced in the form of
${ENV{var}}
where var is the Unix environment variable you
wish to reference."
My external IP is placed each night in a file call
/etc/myip and placed
in the $MYIP variable by /etc/bashrc when an shell
is loaded.
So I have /etc/myip refreshed each night in a cron
job and when a shell
is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret
the book?
I say neither seems to work because When I hard
code
externip=69.91.84.176
there are no NAT problems but when I try to access
the $MYIP variable
either of the ways above NAT prevents me hearing
the callee's voice.
I have tried but not found a way to directly
access the contents of MYIP
to the console using the CLI. Is there a way to
see or set _any_ Linux
enviromnent variable using the CLI? More
generally, how do I access the
Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't
know an easy way to
automatically write the value into sip.conf
programatically.
I could have just said "how do I do this" but
wanted to show that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] 3COM NBX and Digium Cards...
I have a customer that has returned two cards, a TE210P and a TE110P because they are no longer working. Both cards were connected to an 3COM NBX system but not to the same one. On the TE210P only the port that was connected to the NBX failed, the other works perfectly. The cards are on separate sites, one is on an IBM server, the other one in a Dell. The only thing in common is that both cards were connected to a NBX. Both Asterisk systems worked fine for about two weeks and suddenly stopped working. Does anyone have any idea what could be causing the failures? Is the NBX capable of doing something that can damage the Digium cards? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy calls on IAX trunk but no problems on internal calls
We are running the default asterisk package on Ubuntu Dapper (which has the advanced timing options used by ztdummy). Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order and often completely breaks up. But on internal calls between extensions, even with call recording turned on, which goes through our asterisk server, everything sounds fine. We also have some test SIP accounts with our provider. Phones connected directly to our provider on these accounts have no problem either, so we are confident that our network conditions are good and QoS is working properly. We are also confident that our provider is not the problem, since the phones that connect directly to our provider without going through our asterisk server are working fine. We thought the problem might be hardware related, so we tried three different machines on it, each with adequate CPU, memory and disk performance. Every machine had the same problem. One of the machines we borrowed from our provider. They were using it with a hardware PRI and said their zttest results were consistenly 99.99 or greater and the server had performed great for them. But with our Ubuntu installation and no hardware, the same server gets results around 99.92. In fact, every one of the machines we tried got fairly bad zttest results, although we have discovered various info that indicate that zttest might not be a very accurate test (http://bugs.digium.com/view.php?id=4301), but it is the only benchmark we know of. We suspect there may be a problem with with the build options in the kernel or in the default asterisk package on dapper, so we are trying out trixbox at the moment. In the mean time, does anyone else have any suggestions? Are there some specific build options or kernel flags we should try? Are there any other approaches that someone might recommend? Thanks in advance for your time. Carl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USB GSM gateway for Asterisk?
There's been some (futile?) effort a while back attempting to get a Bluetooth capable phone integrated into asterisk as a channel. The idea, of course, was to make it possible to have asterisk utilize a cellular connection for backup, calls on free nights/weekends, or free in-network minutes. How about skipping the phone entirely? I just found this -- http://www.falcom.de/?id=283 It's an integrated quad-band GSM engine with a USB connection. Claims to be able to do data, fax, voice, SMS/MMS etc, even EDGE speeds. Prices I found are 170 Euros or US$225. Considering the alternative ($20 USB dongle, $180 BT capable phone), this seems like a very competitive option. I'd expect it to be in Falcom's best interest to support development efforts as it would open the asterisk market to them. Anyone up for creating a bounty-page for this? Voice traffic would be the first priority, but SMS would certainly be a desired option. Personally, I wouldn't mind being able to utilize EDGE as an emergency backup for my network, allowing at least mail traffic and such. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura SPA-3000 vs Sangoma A200
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups the call, Asterisk (Sangoma) does not notice that and keeps ringing. Call is hangup only after 8s! That's totally unacceptable in busy call center. Imagine the agent which picks up the call and hears total silence. I tried all solutions found in Google. Most of them are on AsteriskGuru website. That didn't help. ringtimeout= in zapata.conf drives Asterisk crazy. Sangoma support (you will find it bellow) acknowledged that they don't know how to fix it on A200. Notice/question to Sangoma and others - why cheap SPA-3000 does not have this problem and more expensive A200 can't solve it? Can somebody suggets me working FXO alternative? Regards/Pagarbiai, Mindaugas Kezys Sangoma response to this problem -- Hi Mindaugas, Yuan asked me to respond because I have more experience in this. The way the phone system works is that when the phone is on hook, the line voltage is about 48V DC. To ring, the voltage increases to about 90v AC. When the phone or the FXO goes off hook, the line voltage is about 7 volts for the duration of the call. If the call is cleared at the far end, the voltage goes back to about 48 volts, and that tells us that the call has been terminated. On some systems they use a polarity reversal, or a 500ms drop of carrier current but the principle is the same. ON a good PSTN system, this change in voltage at the end of the call is almost instantaneous. In Canada, for instance, it takes over 10 seconds for the voltage signal to come through. The result is that on our Asterisk PBX at Sangoma, we have exactly the same problem as you: People call in, and hang up when they hear that you are not available, and we get messages about 10 seconds long with no audio. It is very annoying. We certainly would like to find a way around this ourselves. Bell Canada is not interested in our problems. We have tried using a silence filter to cut calls, but it happens often that there is a few seconds of silence in a call. Busydetect works, but the busy tone only comes much later, long after the 10 seconds has passed. I have no idea why some telcos have this delay before sending the disconnect signal. You may have better luck with your telco than we have had with ours. Please let me know if you find anything that helps. Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura SPA-3000 vs Sangoma A200
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups the call, Asterisk (Sangoma) does not notice that and keeps ringing. Call is hangup only after 8s! That's totally unacceptable in busy call center. Imagine the agent which picks up the call and hears total silence. I tried all solutions found in Google. Most of them are on AsteriskGuru website. That didn't help. ringtimeout= in zapata.conf drives Asterisk crazy. Sangoma support (you will find it bellow) acknowledged that they don't know how to fix it on A200. Notice/question to Sangoma and others - why cheap SPA-3000 does not have this problem and more expensive A200 can't solve it? Can somebody suggets me working FXO alternative? Regards/Pagarbiai, Mindaugas Kezys Sangoma response to this problem -- Hi Mindaugas, Yuan asked me to respond because I have more experience in this. The way the phone system works is that when the phone is on hook, the line voltage is about 48V DC. To ring, the voltage increases to about 90v AC. When the phone or the FXO goes off hook, the line voltage is about 7 volts for the duration of the call. If the call is cleared at the far end, the voltage goes back to about 48 volts, and that tells us that the call has been terminated. On some systems they use a polarity reversal, or a 500ms drop of carrier current but the principle is the same. ON a good PSTN system, this change in voltage at the end of the call is almost instantaneous. In Canada, for instance, it takes over 10 seconds for the voltage signal to come through. The result is that on our Asterisk PBX at Sangoma, we have exactly the same problem as you: People call in, and hang up when they hear that you are not available, and we get messages about 10 seconds long with no audio. It is very annoying. We certainly would like to find a way around this ourselves. Bell Canada is not interested in our problems. We have tried using a silence filter to cut calls, but it happens often that there is a few seconds of silence in a call. Busydetect works, but the busy tone only comes much later, long after the 10 seconds has passed. I have no idea why some telcos have this delay before sending the disconnect signal. You may have better luck with your telco than we have had with ours. Please let me know if you find anything that helps. Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 extn not registering on 4569
Hi all, Just having a strange situation with no clues how to solve. I have an Asterisk/TRIXBOX located in US and an IAX extn running on PA168V ATA in another country. All my configs seems to be on 4569 but i see my extn connected at a different port like 13569. How can i make it to register at 4569 on my asterisk? Please help Thanks all Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy calls on IAX trunk but no problems on internal calls
We are running the default asterisk package on Ubuntu Dapper. Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order and often completely breaks up. But on internal calls between extensions, even with call recording turned on, which goes through our asterisk server, everything sounds fine. We also have some test SIP accounts with our provider. Phones connected directly to our provider on these accounts have no problem either, so we are confident that our network conditions are good and QoS is working properly. We are also confident that our provider is not the problem, since the phones that connect directly to our provider are working fine. We thought the problem might be hardware related, so we tried three different machines on it, each with adequate CPU, memory and disk performance. Every machine had the same problem. One of the machines we borrowed from our provider. They were using it with a hardware PRI and said their zttest results were consistenly 99.99 or greater and the server had performed great for them. But with our Ubuntu installation and no hardware, the same server gets results around 99.92. In fact, every one of the machines we tried got fairly bad zttest results, although we have discovered various info that indicate that zttest might not be a very accurate test (http://bugs.digium.com/view.php?id=4301 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silent Calls (Ghost Calls) When Picking Up Queue Calls
We're having a problem with calls coming in from our TE110P (an E&M wink
T1) through to our queues and then when someone picks up the calls goes
dead or silent. They are becoming known as "ghost calls" in our
organization. It's seems to only have cropped up in the last couple
weeks though we had the problem many months ago when the T1 clock timing
wasn't set to synchronize with the T1 (that is not the case now). Any
body else have this problem or have suggestions on where to start
looking to fix? Here's our zapata.conf, zaptel.conf and the excerpt from
the call log when the call appears to get hung up on from the T1 circuit.
Note: in the log below I saw a line that says "Released clone lock on
'Local/[EMAIL PROTECTED],1'". Could that be part of the
problem?
Thanks in advance - Sascha
[Zapata.conf]
; T1 CARD
language=en
signalling=em_w
context=from-pstn
rxwink=300
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
callerid=asreceived
group=1
progzone=us
channel => 1-24
[Zaptel.conf]
span=1,1,0,esf,b8zs
e&m=1-24
[Asterisk Log Exerpt]
Aug 23 12:23:56 DEBUG[13218] channel.c: Set channel Zap/9-1 to write
format slin
Aug 23 12:23:56 DEBUG[13218] channel.c: Scheduling timer at 160 sample
intervals
Aug 23 12:24:20 DEBUG[13218] chan_zap.c: DTMF digit: 2 on Zap/9-1
Aug 23 12:24:20 DEBUG[13218] channel.c: Scheduling timer at 0 sample
intervals
Aug 23 12:24:20 DEBUG[13218] channel.c: Set channel Zap/9-1 to write
format ulaw
Aug 23 12:24:20 DEBUG[13218] pbx.c: Oooh, got something to jump out with
('2')!
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'Goto'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'Answer'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Expression result is '0'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'GotoIf'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'Set'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'Set'
Aug 23 12:24:23 DEBUG[13218] pbx.c: Launching 'Queue'
Aug 23 12:24:23 DEBUG[13218] app_queue.c: queue: 454, options: t, url: ,
announce: , expires: 1156350383, priority: 0
Aug 23 12:24:23 DEBUG[13218] app_queue.c: Queue '454' Join, Channel
'Zap/9-1', Position '1'
Aug 23 12:24:23 DEBUG[13218] channel.c: Scheduling timer at 160 sample
intervals
Aug 23 12:24:23 DEBUG[13218] app_queue.c: It's our turn (Zap/9-1).
Aug 23 12:24:23 DEBUG[13218] app_queue.c: Zap/9-1 is trying to call a
queue member.
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel Zap/9-1 to write
format ulaw
Aug 23 12:25:11 DEBUG[13218] channel.c: Scheduling timer at 0 sample
intervals
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel Zap/9-1 to read
format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel
Local/[EMAIL PROTECTED],1 to write format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel
Local/[EMAIL PROTECTED],1 to read format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel Zap/9-1 to write
format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Got clone lock for masquerade on
'SIP/102-8bd3' at 0x8f21b4c
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel SIP/102-8bd3 to
write format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel SIP/102-8bd3 to read
format slin
Aug 23 12:25:11 DEBUG[13218] channel.c: Putting channel SIP/102-8bd3 in
64/64 formats
Aug 23 12:25:11 DEBUG[13218] channel.c: Released clone lock on
'Local/[EMAIL PROTECTED],1'
Aug 23 12:25:11 DEBUG[13218] channel.c: Done Masquerading SIP/102-8bd3 (6)
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel Zap/9-1 to read
format ulaw
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel SIP/102-8bd3 to
write format ulaw
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel SIP/102-8bd3 to read
format ulaw
Aug 23 12:25:11 DEBUG[13218] channel.c: Set channel Zap/9-1 to write
format ulaw
Aug 23 12:25:11 VERBOSE[13218] logger.c: -- ***[JB LOG]*** fixed
jitterbuffer created on channel Zap/9-1
Aug 23 12:25:15 DEBUG[13218] channel.c: Nobody there, continuing...
Aug 23 12:25:15 DEBUG[13218] channel.c: Nobody there, continuing...
Aug 23 12:25:16 DEBUG[13218] rtp.c: Got RTCP report of 52 bytes
...
Aug 23 12:25:18 DEBUG[13218] channel.c: Nobody there, continuing...
Aug 23 12:25:18 DEBUG[13218] channel.c: Didn't get a frame from channel:
SIP/102-8bd3
Aug 23 12:25:18 DEBUG[13218] channel.c: Bridge stops bridging channels
Zap/9-1 and SIP/102-8bd3
Aug 23 12:25:18 DEBUG[13218] channel.c: Hanging up channel 'SIP/102-8bd3'
Aug 23 12:25:18 DEBUG[13218] chan_sip.c: Hangup call SIP/102-8bd3, SIP
callid [EMAIL PROTECTED])
Aug 23 12:25:18 DEBUG[13218] chan_sip.c: update_call_counter(102) -
decrement call limit counter
Aug 23 12:25:18 DEBUG[13218] pbx.c: Spawn extension (ext-queues,454,6)
exited non-zero on 'Zap/9-1'
Aug 23 12:25:18 DEBUG[13511] app_queue.c: Device 'SIP/102' changed to
state '1' (Not in use) but we don't care because they're not a member of
any queue.
Aug 23 12:25:18 DEBUG[13218] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Aug 23 12:25:18 DEBUG[13218] cdr_add
Re: [asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?
I use Speakeasy.net and have been satisfied for a good 4 years now... mogorman wrote: I have used bellsouth dsl and comcast cable. In my experience they both have there problems, but at least in my area I have consistently always gotten anywhere from 2x to 3x more bandwith and reliable rates. but thats just my 2 cents. Mog ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco PIX firewall and nat=yes
If you are running a new version of PIX sw (6.3.4 or 6.3.5), then leave fixup on and set "nat=no". The PIX is the only firewall that I have seen that truly does nat correctly. It nat's both the source and dest inside the packet. You can even do reinvite with multiple phones behind a PIX and it works correctly. One other thing to check. If you have qualify off, then you need to set the phone to re-register in less time that the SIP timeout value in the PIX. For example, if the timeout is 10 mins, then the phone needs to have a register value less than 10 mins. Scott Pinhorne wrote: Hi I use a PIX 515 and had a similar problem when I started. I turned on the fixup for SIP (as well as having nat in sip entry) and it seems to do the trick for me. Good Luck SP Bill Gibbs wrote: Also the phone can dial out from behind the PIX…but obviously not receive calls. Bill *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bill Gibbs *Sent:* Wednesday, August 23, 2006 11:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702x.x.x.x D N 54297UNREACHABLE 701/701x.x.x.x D N 54297UNREACHABLE 700/700x.x.x.x D N 54297UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting Asterisk to Avaya Definity over H.323
Hello, Does anyone out there have experience or settings they can share to help connect Asterisk to an Avaya Definity system over H.323? If so we need your help! Please email me directly. Many thanks, Matt King Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Compilation
The archives should contain these details, but here they are again- Near line 29- change this line: app_test.so app_forkcdr.soTo this: app_test.so app_forkcdr.so app_cbmysql.so Near line 88 add thes lines (just above this line " look: look.c" app_cbmysql.o: app_cbmysql.c $(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c -o app_cbmysql.o app_cbmysql.c**It is important that the $(CC) line start with a tab and not spaces. I posted Web-MeetMe-2.1.0 up on SourceForge yesterday, and it has a self-contained build environment for app_cbmysql. I would recommend that you use 2.1.0 if you are just getting started. The process to build app_cbmysql is much more straight forward and there are a number of key improvements. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled ChehabSent: Wednesday, August 23, 2006 4:09 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Compilation Dear dan Thanks for your help, I am using Web-MeetMe_v2.0.0.gz ,I copied app_cbmysql.c to /usr/src/asterisk/apps/ ,can you please tell me how to include it at the Makefile Regards From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan AustinSent: Tuesday, August 22, 2006 7:19 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Compilation Which version of Web-MeetMe did you download? The process up to 2.0.1 is, well, annoying. Copy app_cbmysql.c to ./asterisk/apps and modify the Makefile to include the application. The project is now hosted on SourceForge and has a much improved build process, but I have not built a release tarball yet. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Khaled ChehabSent: Tuesday, August 22, 2006 4:01 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Cc: [EMAIL PROTECTED]Subject: [asterisk-users] Compilation Dear I am installing Web-MeetMe ,one of the requirements is app_cbmysql.c I have it but ,how can I compile it . Regards *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco PIX firewall and nat=yes
Hi I use a PIX 515 and had a similar problem when I started. I turned on the fixup for SIP (as well as having nat in sip entry) and it seems to do the trick for me. Good Luck SP Bill Gibbs wrote: Also the phone can dial out from behind the PIX…but obviously not receive calls. Bill *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bill Gibbs *Sent:* Wednesday, August 23, 2006 11:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702x.x.x.x D N 54297UNREACHABLE 701/701x.x.x.x D N 54297UNREACHABLE 700/700x.x.x.x D N 54297UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [asterisk-users] Snom360 with 6.2.2 firmware
Well, you could just press the transfer button when the line starts to ring instead of waiting for someone to answer. -Brodie On Wednesday 23 August 2006 02:07 am, Giordano Grandis wrote: > Thanks, but my problem is that I need to transfer a call, while the called > party is ringing. I cannot wait that the called > to call. > > Thanks again > > Giordano > > -Messaggio originale- > Da: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Per conto di Brodie > Macleod Inviato: martedì 22 agosto 2006 16.44 > A: Asterisk Users Mailing List - Non-Commercial Discussion > Oggetto: Re: [asterisk-users] Snom360 with 6.2.2 firmware > > Although I'm not using this firmware, attended transfers on these phones > are done like this (while talking to the person you want to transfer): > > 1. Press one of the other line keys and dial the destination number of the > person you are transferring to (your caller on line 1 will be put on hold). > 2. If the person answers and is ready to accept the call, press the > Transfer button, and line 1 & line 2 will be bridged along with you, after > which time you can hangup the phone, leaving the caller and callee > connected. > > -Brodie > > On Tuesday 22 August 2006 02:44 am, Giordano Grandis wrote: > > Hi all, > > I'm using a Snom360 with bristuffed asterisk and i want to known if is > > possibile realize somthing of this: I receive an incoming call and then > > answered I want to transfer it to a cell phone (or another pubblic > > number), so press "transfer" on the phone, call the number and only if > > the called party is avaible i want to transfer the call. Infact with the > > transfer key, when i send the number, i lost the state of call, and i do > > not known if the called party was avaible or no. > > Is there a way to realize this ? > > > > Thanks very much in advance > > > > Giordano > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registering IP Phone To Asterisk
What is the process to get an IP phone registered to Asterisk? I bought an Asterisk with a GUI and it has templates for devices such as sipura, cisco, and xten but I am using a Fanvil IP phone. How do I load the template for my IP phone into astrisk so that it can work? Thanks Wyatt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco PIX firewall and nat=yes
Also the phone can dial out from behind the PIX…but obviously not receive calls. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, August 23, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702 x.x.x.x D N 54297 UNREACHABLE 701/701 x.x.x.x D N 54297 UNREACHABLE 700/700 x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco PIX firewall and nat=yes
I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702 x.x.x.x D N 54297 UNREACHABLE 701/701 x.x.x.x D N 54297 UNREACHABLE 700/700 x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird compile problem
I'm in the process of upgrading an asterisk to 1.2.10 and started by upgrading libpri-1.2.3 (make & make install) and zaptel (make & make install). Was about to install asterisk, but doing a "ls" I get the following error: ls: relocation error: /lib/libpthread.so.0: symbol _h_errno, version GLIBC_2.0 not defined in file libc.so.6 with link time reference How can compiling and installing zaptel and libpri cause errors like this in other programs ? Running a gentoo with 2.6.11 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk with multiple IPs?
In my case it was not a class c, but just 4 separate addresses, one each in NY, Seattle, Miami and London on the Level 3 network. I ended up creating separate entries for each, in and out, and for the outbound route, put all 4 in the order of their ping times. That is working nicely. W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?
I have used bellsouth dsl and comcast cable. In my experience they both have there problems, but at least in my area I have consistently always gotten anywhere from 2x to 3x more bandwith and reliable rates. but thats just my 2 cents. Mog ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom360 with 6.2.2 firmware
This bridges the call on the phone and not the switch unless I am mistaken On 8/22/06, Brodie Macleod <[EMAIL PROTECTED]> wrote: Although I'm not using this firmware, attended transfers on these phones are done like this (while talking to the person you want to transfer): 1. Press one of the other line keys and dial the destination number of the person you are transferring to (your caller on line 1 will be put on hold). 2. If the person answers and is ready to accept the call, press the Transfer button, and line 1 & line 2 will be bridged along with you, after which time you can hangup the phone, leaving the caller and callee connected. -Brodie On Tuesday 22 August 2006 02:44 am, Giordano Grandis wrote: > Hi all, > I'm using a Snom360 with bristuffed asterisk and i want to known if is > possibile realize somthing of this: I receive an incoming call and then > answered I want to transfer it to a cell phone (or another pubblic > number), so press "transfer" on the phone, call the number and only if > the called party is avaible i want to transfer the call. Infact with the > transfer key, when i send the number, i lost the state of call, and i do > not known if the called party was avaible or no. > Is there a way to realize this ? > > Thanks very much in advance > > Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk with multiple IPs?
I'm thinking I used deny and permit statements on broadvoice.com way back when, and the configs/sip.conf.sample suggests its still valid for v1.2.10 code. You might take another look at that for sip. Benjamin Lawetz wrote: Agreed that with a other IAX and SIP that have registration information and secrets that works. The problem is when you have a provider that just sends you a SIP call and the only way to identify it is by IP address. In those cases (if I understand correctly) we need a host line don't we? (Or at least I remember when I was testing a while back that it wouldn't work with deny and permit) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: August 23, 2006 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk with multiple IPs? Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? Here's an iax.conf example of what I'm using: [teliax] context=teliax-incoming type=user auth=md5 secret=mysecret jitterbuffer=yes disallow=all allow=gsm deny=0.0.0.0/0.0.0.0 permit=207.174.202.0/255.255.255.0 The last two statements essentially restrict incoming calls from teliax to one of their class-c networks (regardless of how many servers or IP's they have). Note that on incoming calls the host= line is not used. If you're really asking how to do that for outgoing calls, you'll probably have to do it through three/four sections (type=peer) and deal with those sections in your dialplan. As a side note, there are a large percentage of * implementors that don't understand the search terms when an incoming call is being negotiated (eg, is host= used, is secret= used). Without that understanding, calls likely come into different sections then what the implementor actually expected. The deny & permit statements are very useful to tighten down security for each incoming context. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I like using the TOS header bits personally as lots of other protocols (eg, dns) will eventually match on udp/4569. For the TOS bits & v1.2.10, use tos=lowdelay in iax.conf and on the cisco use an access list to match on the tos bits. Something like: access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay <= same as tos=lowdelay access-list 103 permit ip any any tos 12 For the TOS bits & svn truck, the tos= settings have changed in asterisk. Look in the supplied documentation (eg, readme's, sample configs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access list item "dscp cs3" and "dscp ef". If you're not all that experienced on cisco qos, then the following is an example of a working config that you should be able to translate into your router config one way or another. class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue ! interface Dialer0 bandwidth 555 service-policy output voice-policy ! access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any The above config provides low-latency "priority" to voice-rtp, then provides an additional qos piece to ensure www-traffic is given bandwidth before all of the "class-default" traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of bandwidth=555 above) "if" voice traffic is present. If voice traffic isn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic "after" the router deals with voice-rtp traffic. The default class always gets what bandwidth is left over (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a "show access-list 103" from the CLI (on the router) and watch for matching packets in each access list line. Once you've structured the access list to truly match asterisk traffic, then do a "show policy-map interface dialer0" to display how the overall qos structure is functioning. Note that cisco didn't get real serious about IOS qos until v12.2 of their IOS code. In v12.2 (and later versions of IOS) there has been a significant amount of work to bring all of their products into industry standard implementations / conformance / expectations. If you want to get real serious with cisco's qos stuff, purchase the book "End-to-end QoS Network Design" and read the 700+ pages devoted to the subject. It is an excellent book with lots of examples, etc. The book (and actual practice) suggests IOS v12.3 has more QoS funtionality then v12.2, and v12.4 has more then v12.3. (The authors of the book back that statement up 100% as well, and they are cisco employees.) In the above config, the "bandwidth=555" statement is very important. It should represent the "actual" outgoing bandwidth for whatever interface you are using and not the theoretical max that someone said you should get. Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time is consumed sending a non-voice 1500-byte packet, the time is likely to be more then the 20 millisecond interval for sip/iax traffic. If that is your case, then you may need to forcibly reduce the MTU size of packets originating from other non-voice workstations/servers. The later cisco IOS versions have a parameter to do that if you can't do it via the workstation/server configuration parameters. If memory serves correctly, that parameter appeared around v12.4 of their IOS. One last item... all of the above deals only with "outgoing" traffic. You would need to talk to your ISP about QoS for your incoming traffic, and most of the local ISP's don't have a clue. Increasingly, some of the larger backbone isp's are beginning to understand QoS and some have actually implemented something. However, those isp's are heading towards providing QoS as a value-add chargeable service (as in MPLS, etc). R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. I just spent some time doing this myself. If your routers already can prioritize traffic based on the TOS bits in your IP traffic, setting the tos in iax.conf should be the first place to start. If your routers don't do that already you will need to mess with setting up queuing/policies on the routers. Here are some places to start: Here are some awesome descriptions on how QoS works and what the different methods of implementing queuing and traffic shaping on Cisco hardware are. They may be a bit dated depending on what kind or routers you are using: http://www.netcraftsmen.net/welcher/papers/qos1.html http://www.netcraftsmen.net/welcher/papers/qos2.html http://www.netcraftsmen.net/welcher/papers/qos3.html Here are some pages from the wiki that talk about QoS on cisco hardware. Not sure what "type" of queueing this uses, but it allocates a certain amount of available traffic to voice traffic. Any unused voice traffic will be shared by what's left: http://www.voip-info.org/wiki/view/QoS+Cisco This page seems to talk about CAR (Committed Access Rate): http://www.voip-info.org/wiki/view/QoS+Cisco+IOS I ended up using priority queuing on my routers, giving voice first priority and everything else lower priorities. Not the best solution but it was the easiest for me to implement with the versions of IOS I have. If you would like to see my configs, let me know. -Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
I may have to eat my words, then. This is the case with trunk, and I can't recall the last time I built a 1.2.x system. I could have sworn that behavior didn't change, but I've been wrong before. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon Sent: Wednesday, August 23, 2006 10:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Hint extension issue - bug? On 1.2.10, presence is working very well using friend. The state is refreshing successfully. There is probably antoher problem with your installation cause I'm using hint with friend since 1 years in all my production system. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 23 août 2006 08:50 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Hint extension issue - bug? I'm using asterisk 1.2.10 David Gagnon wrote: >Are you having this problem with the trunk? > > > >-Message d'origine- >De : [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez >Envoyé : 22 août 2006 18:23 >À : Asterisk Developers Mailing List; Asterisk Users Mailing List - >Non-Commercial Discussion Objet : [asterisk-users] Hint extension issue >- bug? > >Hi, I'm using the hint extension to monitoring the status of some >extensions. If the extension is defined as a friend, the monitoring >doesn't work any more. It only work if I define it as a "peer". Is that >right ? I mean, I supposed that an extension defined as a friend should >have all the functionality of "user" and "peer" types. Is this >documented somewhere? How can I know the status of an extension of type >friend? I hope someone could bring me some light about this issue. >Thanks in advance. > >Lucas Alvarez > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set externip in sip.conf automatically?
The easiest way is to register for free dynamic DNS
service at www.dyndns.com. Then use externhost=
instead of externip= in sip.conf . If you are using a
Linksys router like the WRT54G, it already has a
dyndns client which will update the dyndns servers
with your ip address everytime it changes.
Greg
--- Larry Alkoff <[EMAIL PROTECTED]> wrote:
> As stated in the original post, when I entter the IP
> with an editor
> directly into sip.conf calls work just fine but I am
> looking for a way
> to have that done _automatically_.
>
> The Asterisk - Future of Telephony book says it is
> possible for Asterisk
> to access a Linux environment variable containing
> the IP information in
> the form of "${ENV{variable}}.
>
> It doesn't seem to work. I am asking how to make it
> work.
>
> Larry
>
> Watkins, Bradley wrote:
> > If you already have the IP in a file, why don't
> you set it up so the
> > file itself says: externip=xx.xx.xx.xx and then
> do a #include in
> > sip.conf for the /etc/myip file? I believe you'll
> have to do a sip
> > reload either way (which can obviously be part of
> your cron job) if
> > you're not already, but that should do what you're
> looking to do.
> >
> > - Brad
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of Larry
> > Alkoff
> > Sent: Tuesday, August 22, 2006 9:34 PM
> > To: Asterisk-users; Austin-asterisk-users
> > Subject: [asterisk-users] How to set externip in
> sip.conf automatically?
> >
> > I need to give Asterisk access to my external IP
> address to prevent
> > the NAT problem where caller cannot hear the
> callee's voice.
> >
> > According to Asterisk - The Future of Telephony
> page 92 Environment
> > Variables:
> >
> >"Environment variables are a way of accessing
> Unix environment
> > variables from within Asterisk. They are
> referenced in the form of
> >${ENV{var}}
> > where var is the Unix environment variable you
> wish to reference."
> >
> > My external IP is placed each night in a file call
> /etc/myip and placed
> > in the $MYIP variable by /etc/bashrc when an shell
> is loaded.
> >
> > So I have /etc/myip refreshed each night in a cron
> job and when a shell
> > is opened /etc/bashrc does:
> > export MYIP=`cat /etc/myip`
> >
> > To access the variable in sip.conf I have tried:
> >
> > externip=${ENV(EXTERNIP)}
> > and
> > ${ENV($EXTERNIP)}
> > but neither seems to work.
> > Is this the correct syntax? Did I misinterpret
> the book?
> >
> > I say neither seems to work because When I hard
> code
> > externip=69.91.84.176
> > there are no NAT problems but when I try to access
> the $MYIP variable
> > either of the ways above NAT prevents me hearing
> the callee's voice.
> >
> > I have tried but not found a way to directly
> access the contents of MYIP
> > to the console using the CLI. Is there a way to
> see or set _any_ Linux
> > enviromnent variable using the CLI? More
> generally, how do I access the
> > Linux shell from the CLI?
> >
> > The problem with simply using
> > externip=69.91.94.176
> > is that number is subject to change and I don't
> know an easy way to
> > automatically write the value into sip.conf
> programatically.
> >
> > I could have just said "how do I do this" but
> wanted to show that I've
> > done my homework.
> > Thanks for any help.
> >
> > Larry
> >
> > --
> > Larry Alkoff N2LA - Austin TX
> > Using Thunderbird on Linux
> > ___
> > --Bandwidth and Colocation provided by
> Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > The contents of this e-mail are intended for the
> named addressee only. It contains information that
> may be confidential. Unless you are the named
> addressee or an authorized designee, you may not
> copy or use it, or disclose it to anyone else. If
> you received it in error please notify us
> immediately and then destroy it.
> > ___
> > --Bandwidth and Colocation provided by
> Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> Larry Alkoff N2LA - Austin TX
> Using Thunderbird on Linux
> ___
> --Bandwidth and Colocation provided by Easynews.com
> --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
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RE: [asterisk-users] Trunk with multiple IPs?
Agreed that with a other IAX and SIP that have registration information and secrets that works. The problem is when you have a provider that just sends you a SIP call and the only way to identify it is by IP address. In those cases (if I understand correctly) we need a host line don't we? (Or at least I remember when I was testing a while back that it wouldn't work with deny and permit) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: August 23, 2006 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk with multiple IPs? Benjamin Lawetz wrote: > Still no answers huh? > > I've asked a couple of time how to do this, and by the lack of > answers, I'm guessing there is no way. > The workaround unfortunately is to create an entry for each IP address > in the range (I hope you don't have to open up a whole C class) > > -Original Message- > How do I enter a trunk with multiple IPs. > > xyz voip provider has 4 IPs and I want to allow incoming from any of > them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 > > Do I put 4 separate host= lines, do I put a single host=line that is > comma separated or do I have to set up 4 separate incoming trunks? > Here's an iax.conf example of what I'm using: [teliax] context=teliax-incoming type=user auth=md5 secret=mysecret jitterbuffer=yes disallow=all allow=gsm deny=0.0.0.0/0.0.0.0 permit=207.174.202.0/255.255.255.0 The last two statements essentially restrict incoming calls from teliax to one of their class-c networks (regardless of how many servers or IP's they have). Note that on incoming calls the host= line is not used. If you're really asking how to do that for outgoing calls, you'll probably have to do it through three/four sections (type=peer) and deal with those sections in your dialplan. As a side note, there are a large percentage of * implementors that don't understand the search terms when an incoming call is being negotiated (eg, is host= used, is secret= used). Without that understanding, calls likely come into different sections then what the implementor actually expected. The deny & permit statements are very useful to tighten down security for each incoming context. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Direct to Voicemail
Since you're using the variables to decide what to do next
(VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the
general section of extensions.conf, unless you're using the n+101
priority jumping elsewhere.
On Wed, 2006-08-23 at 08:39 -0400, Doug Lytle wrote:
> Hey everybody,
>
> I've set up an extension that allows users to send a call directly to
> voice mail. Yesterday, someone accidentally sent a call to an extension
> that didn't exist and the call was dropped. I found the option to check
> if a mailbox exists and it works fine, but I get the following 'warning':
>
> Spawn extension (sip, 04258, 0) exited non-zero on 'Zap/3-1'
> -- Executing Set("Zap/3-1", "_direct_vm=4258") in new stack
> -- Executing MailboxExists("Zap/3-1", "[EMAIL PROTECTED]|") in new stack
> Aug 23 08:26:30 WARNING[8313]: app_voicemail.c:5697 vm_box_exists: VM
> box [EMAIL PROTECTED] exists, but extension 04258, priority 103 doesn't exist
>
>
> Is there a way to avoid this warming? Code fragment below:
>
> [direct-to-voicemail]
>
> ; **
> ; Allow anybody to send a call directly to voicemail
> ; by pre-pending a 0 to the destination extension.
> ; Checks to see if voice mail box exists, if not
> ; Tells the callee that no such vm box exists and
> ; then transfers them to the operator
> ; **
>
> exten => _04XXX,1,Set(_direct_vm=${EXTEN:1})
> exten => _04XXX,2,MailboxExists([EMAIL PROTECTED])
> exten => _04XXX,3,Goto(s-${VMBOXEXISTSSTATUS},1)
> exten => s-FAILED,1,SayDigits(${direct_vm})
> exten => s-FAILED,2,Playback(vm-nobox)
> exten => s-FAILED,3,Playback(pbx-transfer)
> exten => s-FAILED,4,Goto(incoming,s,1)
> exten => s-SUCCESS,1,Set(CALLBACK=${DB(vmcallback/${direct_vm})})
> exten => s-SUCCESS,2,GotoIf($["${CALLBACK}" =
> "YES"]?s-SUCCESS,3:s-SUCCESS,4)
> exten => s-SUCCESS,3,System(/usr/local/bin/vm-callout.sh ${direct_vm})
> exten => s-SUCCESS,4,Voicemail([EMAIL PROTECTED])
> exten => s-SUCCESS,5,Hangup()
>
>
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce, this might be able to help give you some hints or a place to start: http://www.voip-info.org/wiki/view/QoS+Cisco Hope that helps \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
Woah there... Relax, man. I will concur that there are some inconsistencies and things are not exactly how they should be. I'm mostly just pointing out that, for various reasons that I'm not particularly well-equipped to discuss (oej would be able to regale you with the necessary history if you require it), this is how it works. It works that way intentionally, and is not a bug per se. It is an unfortunate design decision, but one which was necessary in order to implement the functionality without requiring a full rewrite. But what I previously said stands: do not use friend, it gains you nothing and breaks certain things. I know that when Olle gets the time, the full rewrite of the sip channel will do away with the user/peer/friend nonsense (among many, many other things) that exists today. But that is not what exists in Asterisk at the moment, and I was mostly trying to point out how things are now. Anyway, I'll be the last to acuse you of not putting your money where your mouth is. I know you've contributed to Asterisk (both code and wisdom on the lists), probably much more than I have. No worries there. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, August 23, 2006 9:30 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hint extension issue - bug? On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote: > It's not a bug. When you use type=friend, it will create a user > object > *and* a peer object. This will make call-limit not function, thereby > breaking hints. There is no reason to use friend anyway. It does not > gain you any functionality, and in fact breaks some. This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of "calltype" taking "incoming", "outgoing" or "both" would be far clearer and eliminate all this inconsistency. Call-limiting not working with users is just as dumb an idea as users not being able to trunk calls in iax2. They're artificial boundaries set up for no reason other than to force a distinction between the two types of entries. Eliminate all the crap and let me use the damn PBX how I want; that's one of the biggest features of Asterisk. Stop trying to protect me for my own good. Document the shit, make it consistent and let the community support the clueless. You don't see this kind of crap with apache, openswan, postfix or even the kernel itself. There's no need to tie my hands behind my back in order to protect the newb. All you'll end up with is a system only newbs want to use. Before anyone accuses me of not putting my money where my mouth is: I've submitted a number of patches over the years to correct or address what I consider inconsistencies, and I do what I can to test out trunk, report bugs and document. I'm doing what I can to help the system. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
On 1.2.10, presence is working very well using friend. The state is refreshing successfully. There is probably antoher problem with your installation cause I'm using hint with friend since 1 years in all my production system. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 23 août 2006 08:50 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Hint extension issue - bug? I'm using asterisk 1.2.10 David Gagnon wrote: >Are you having this problem with the trunk? > > > >-Message d'origine- >De : [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez >Envoyé : 22 août 2006 18:23 >À : Asterisk Developers Mailing List; Asterisk Users Mailing List - >Non-Commercial Discussion >Objet : [asterisk-users] Hint extension issue - bug? > >Hi, I'm using the hint extension to monitoring the status of some >extensions. If the extension is defined as a friend, the monitoring >doesn't work any more. It only work if I define it as a "peer". Is that >right ? I mean, I supposed that an extension defined as a friend should >have all the functionality of "user" and "peer" types. Is this >documented somewhere? How can I know the status of an extension of type >friend? I hope someone could bring me some light about this issue. >Thanks in advance. > >Lucas Alvarez > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
Brad, It works with friend. I'm using this config since 1 year. I dunno why it didn't work for Andrew. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Watkins, Bradley Envoyé : 23 août 2006 08:48 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] Hint extension issue - bug? It's not a bug. When you use type=friend, it will create a user object *and* a peer object. This will make call-limit not function, thereby breaking hints. There is no reason to use friend anyway. It does not gain you any functionality, and in fact breaks some. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, August 23, 2006 8:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hint extension issue - bug? On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote: > This is actually working as designed. You need to use type=peer in > order for call-limit to work properly, which in turn is what allows > hints to work properly. I'm sorry, but type=friend is *SUPPOSED* to equal both user and peer. If friend does not work and peer does, then it's broken. Period. Lucas, I'd file a bug. It's probably something very simple, but I'd have to do a little digging to see for sure. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
On 8/23/06, Bruce Reeves <[EMAIL PROTECTED]> wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. Can't you just setup a policy class based on the host/UDP ports participating in your IAX networking? The RTP isn't separated in IAX so you don't need to keep track of signalling and RTP traffic separately like you would with SIP. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VM - advanced options?
Rich Adamson wrote: running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Rich, If you don't have the dialout option enabled in the voicemail.conf, then nothing will be presented in the Advanced menu. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?
On 8/23/06, Alistair Cunningham <[EMAIL PROTECTED]> wrote: Does anyone have an opinion of: 1. Comcast Cable 2. Bellsouth DSL for residential internet and VoIP service? I'm particularly interested in reports on: 1. VoIP voice quality. 2. Any NAT or firewall problems with SIP. 3. How long they take to install the service from date of order. 4. How friendly they are to 3rd party routers and firewalls. In Northern New Jersey here I've got residential Comcast at the house and I have a backup Comcast business connection here at the office. Both have been real reliable and after moving back in March I was actually able to get support on the line at 15 minutes before midnight to reset the MAC address lock on the cable modem and get me back online again at my new place. We don't have any POTS connectivity into the house and haven't had it for a little over a year now, and the voice quality has been fine. We've also got Linksys routers with customized firmware in place at both locations and have not had any inter-op issues. I really don't have anything bad to say about them. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hint extension issue - bug?
Andrew Kohlsmith wrote: This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of "calltype" taking "incoming", "outgoing" or "both" would be far clearer and eliminate all this inconsistency. This is standard telephony nomenclature. Makes much more sense, even to the newbe, than the way it is now. This is just one of many side effects from the original design not learning from the industry Call-limiting not working with users is just as dumb an idea as users not being able to trunk calls in iax2. They're artificial boundaries set up for no reason other than to force a distinction between the two types of entries. Eliminate all the crap and let me use the damn PBX how I want; that's one of the biggest features of Asterisk. Stop trying to protect me for my own good. Document the shit, make it consistent and let the community support the clueless. With consistency and documentation, even the clueless will need less support! JMO John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Router QOS and IAX2
I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VM - advanced options?
running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Have I missed a config option, sound file, or is the advanced option not totally implemented as yet? R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Hint extension issue - bug?
Hello Wouldn't the correct way of handling call limits, be using the Call Group Applications available in Asterisk? Regards Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andrew Kohlsmith Sendt: 23. august 2006 15:30 Til: asterisk-users@lists.digium.com Emne: Re: [asterisk-users] Hint extension issue - bug? On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote: > It's not a bug. When you use type=friend, it will create a user object > *and* a peer object. This will make call-limit not function, thereby > breaking hints. There is no reason to use friend anyway. It does not > gain you any functionality, and in fact breaks some. This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of "calltype" taking "incoming", "outgoing" or "both" would be far clearer and eliminate all this inconsistency. Call-limiting not working with users is just as dumb an idea as users not being able to trunk calls in iax2. They're artificial boundaries set up for no reason other than to force a distinction between the two types of entries. Eliminate all the crap and let me use the damn PBX how I want; that's one of the biggest features of Asterisk. Stop trying to protect me for my own good. Document the shit, make it consistent and let the community support the clueless. You don't see this kind of crap with apache, openswan, postfix or even the kernel itself. There's no need to tie my hands behind my back in order to protect the newb. All you'll end up with is a system only newbs want to use. Before anyone accuses me of not putting my money where my mouth is: I've submitted a number of patches over the years to correct or address what I consider inconsistencies, and I do what I can to test out trunk, report bugs and document. I'm doing what I can to help the system. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 22-08-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 22-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trunk with multiple IPs?
Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: August 22, 2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trunk with multiple IPs? How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote: > It's not a bug. When you use type=friend, it will create a user object > *and* a peer object. This will make call-limit not function, thereby > breaking hints. There is no reason to use friend anyway. It does not > gain you any functionality, and in fact breaks some. This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of "calltype" taking "incoming", "outgoing" or "both" would be far clearer and eliminate all this inconsistency. Call-limiting not working with users is just as dumb an idea as users not being able to trunk calls in iax2. They're artificial boundaries set up for no reason other than to force a distinction between the two types of entries. Eliminate all the crap and let me use the damn PBX how I want; that's one of the biggest features of Asterisk. Stop trying to protect me for my own good. Document the shit, make it consistent and let the community support the clueless. You don't see this kind of crap with apache, openswan, postfix or even the kernel itself. There's no need to tie my hands behind my back in order to protect the newb. All you'll end up with is a system only newbs want to use. Before anyone accuses me of not putting my money where my mouth is: I've submitted a number of patches over the years to correct or address what I consider inconsistencies, and I do what I can to test out trunk, report bugs and document. I'm doing what I can to help the system. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set externip in sip.conf automatically?
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for Asterisk
to access a Linux environment variable containing the IP information in
the form of "${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but that should do what you're looking to do.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Larry
Alkoff
Sent: Tuesday, August 22, 2006 9:34 PM
To: Asterisk-users; Austin-asterisk-users
Subject: [asterisk-users] How to set externip in sip.conf automatically?
I need to give Asterisk access to my external IP address to prevent
the NAT problem where caller cannot hear the callee's voice.
According to Asterisk - The Future of Telephony page 92 Environment
Variables:
"Environment variables are a way of accessing Unix environment
variables from within Asterisk. They are referenced in the form of
${ENV{var}}
where var is the Unix environment variable you wish to reference."
My external IP is placed each night in a file call /etc/myip and placed
in the $MYIP variable by /etc/bashrc when an shell is loaded.
So I have /etc/myip refreshed each night in a cron job and when a shell
is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret the book?
I say neither seems to work because When I hard code
externip=69.91.84.176
there are no NAT problems but when I try to access the $MYIP variable
either of the ways above NAT prevents me hearing the callee's voice.
I have tried but not found a way to directly access the contents of MYIP
to the console using the CLI. Is there a way to see or set _any_ Linux
enviromnent variable using the CLI? More generally, how do I access the
Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't know an easy way to
automatically write the value into sip.conf programatically.
I could have just said "how do I do this" but wanted to show that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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RE: [asterisk-users] PRI and Asterisk
I have tested Redfone’s boxes. Tried two of them and was able to re-create some issues. I did not have PRI lines but a 24 channel e&m wink line so not sure if PRI is affected as well. I found that over time we had issues with hanging zap channels. Asterisk reported everything was just fine yet people got busy signals calling in and when calling out all they got was silence. The CLI never showed any incoming calls that were attempted and when dialing out it showed Dialing but nothing happened. I worked with Mark Warren at Redfone and he was very co-operative and had an idea to fix this but sadly we just didn’t have any more time to fight with it and went with Digium cards. As of this writing I am starting to get problems with inbound calls. Seems for a couple minutes no one can dial into our office and then it just clears up. No errors or anything in Asterisk to indicate a problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Varanini Sent: Tuesday, August 22, 2006 6:35 PM To: Julian Varanini Subject: RE: [asterisk-users] PRI and Asterisk Hi Everyone Any opinions on this? Thanks Julian From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Tue, 18 Jul 2006 01:29:57 + Subject: [asterisk-users] PRI and Asterisk Hi All, I am planning to order a PRI and would like to know your opinions on a devices like the Redfone redbridge. Basically any PRI to Asterisk interface that has worked well for you. Thanks, Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slightly off-topic: Opinions of Comcast and Bellsouth?
Does anyone have an opinion of: 1. Comcast Cable 2. Bellsouth DSL for residential internet and VoIP service? I'm particularly interested in reports on: 1. VoIP voice quality. 2. Any NAT or firewall problems with SIP. 3. How long they take to install the service from date of order. 4. How friendly they are to 3rd party routers and firewalls. -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange SIP response
Have you done a "show channels" to see if Asterisk thinks that SIP Device is in use? I experienced this problem once after doing a Blind-Transfer from a Cisco 7940 SIP Phone. The transferred call had long since been disconnected, but the Cisco phone thought it still had control of the call, so it wouldn't accept any new calls. The only way I was able to get past it was to re-boot the Cisco phone. Diego Andrés Asenjo González wrote: Rushowr wrote: Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites Yes, I have used it. The lines are extracted from a sip debug on the CLI. I'm going to paste more lines: Sip read: SIP/2.0 480 Temporarily Unavailable To: ;tag=e4331437 From: "24307022";tag=as288765a2 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.50 Transmitting: ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1 From: "24307022" ;tag=as288765a2 To: ;tag=e4331437 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.50:6198 -- SIP/EXT25-a454 is circuit-busy == Everyone is busy/congested at this time I have not detected packet losses even. Thanks for your response. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple CDR parser to print to webpage
Yeah, use Asterisk-Addons and configure the CDR to go into a MySQL database. Once, there, it's really easy to use PHP or Perl to create a custom web-page that shows whatever you want to see. I've got one set up to search for a specific period of time, or for a specific extension. Christopher Aloi wrote: Hello - I'm searching for a simple php or perl script to parse Asterisk's CDR csv into a formatted webpage - anyone have any suggestions? -- -- Christopher T Aloi -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hint extension issue - bug?
I'm using asterisk 1.2.10 David Gagnon wrote: Are you having this problem with the trunk? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 22 août 2006 18:23 À : Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Hint extension issue - bug? Hi, I'm using the hint extension to monitoring the status of some extensions. If the extension is defined as a friend, the monitoring doesn't work any more. It only work if I define it as a "peer". Is that right ? I mean, I supposed that an extension defined as a friend should have all the functionality of "user" and "peer" types. Is this documented somewhere? How can I know the status of an extension of type friend? I hope someone could bring me some light about this issue. Thanks in advance. Lucas Alvarez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Missing Extension
When I do a sip show peers I see that some phones have lost their registration or is no longer reachable. When this occurs I would like the system to send someone an email that the extension is no longer reachable. Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 FWD Network: 56505 Website: http://www.upperclassman.net Billing Questions: billing @ upperclassman.net Rental Questions: rentals @ upperclassman.net Maintenance: help @ upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to Asset Management LLC, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 23, 2006 12:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Missing Extension Can you explain what you mean by disappears? or by disconnected? On 8/22/06, Roger Workman <[EMAIL PROTECTED]> wrote: > Is there a way to have asterisk send an email when a extension disappears or > is disconnected? > > Roger Workman > Business Development > Upperclassman/Universal Holdings LLC > Voice: 304.324.3800 > Fax: 304.324.3801 > ICQ: 4447584 > FWD Network: 56505 > Website: http://www.upperclassman.net > Billing Questions: billing @upperclassman.net > Rental Questions: rentals @upperclassman.net > Maintenance: help @upperclassman.net > > > > This e-mail and any of its attachments may contain sensitive information, > which is privileged, confidential, or subject to copyright belonging to RW > Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is > intended solely for the use of the individual or entity to which it is > addressed. If you are not the intended recipient of this e-mail, you are > hereby notified that any dissemination, distribution, copying, or action > taken in relation to the contents of and attachments to this e-mail is > strictly prohibited and may be unlawful. If you have received this e-mail in > error, please notify the sender immediately and permanently delete the > original and any copy of or printout of this e-mail. > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warrick Zedi > Sent: Tuesday, August 22, 2006 6:29 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Asterisk IAXmodem HylaFax? > > If you're looking for alternatives to Zetafax why not look at AsterFax > (http://asterfax.sourceforge.net)? Your clients can use their existing > e-mail client to send faxes. > > > Warrick > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hint extension issue - bug?
It's not a bug. When you use type=friend, it will create a user object *and* a peer object. This will make call-limit not function, thereby breaking hints. There is no reason to use friend anyway. It does not gain you any functionality, and in fact breaks some. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, August 23, 2006 8:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hint extension issue - bug? On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote: > This is actually working as designed. You need to use type=peer in > order for call-limit to work properly, which in turn is what allows > hints to work properly. I'm sorry, but type=friend is *SUPPOSED* to equal both user and peer. If friend does not work and peer does, then it's broken. Period. Lucas, I'd file a bug. It's probably something very simple, but I'd have to do a little digging to see for sure. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No retry after DNS failure
On Wednesday 23 August 2006 07:07, Remco Barendse wrote: > I am aware that it could mean serious delays for a call to be completed if > the dns lookup was done for every call but surely it should be possible > for * to keep re-trying to resolve an ip address for previous failed > entries let's say every minute or so? The added load is negligable. kpfleming has done some work on this, using a separate DNS resolver thread. Perhaps he can chime in and give us some details? This was done quite a while ago, and actually caused me some trouble, which is the only reason I know about it. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote: > This is actually working as designed. You need to use type=peer in order > for call-limit to work properly, which in turn is what allows hints to work > properly. I'm sorry, but type=friend is *SUPPOSED* to equal both user and peer. If friend does not work and peer does, then it's broken. Period. Lucas, I'd file a bug. It's probably something very simple, but I'd have to do a little digging to see for sure. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Direct to Voicemail
Hey everybody,
I've set up an extension that allows users to send a call directly to
voice mail. Yesterday, someone accidentally sent a call to an extension
that didn't exist and the call was dropped. I found the option to check
if a mailbox exists and it works fine, but I get the following 'warning':
Spawn extension (sip, 04258, 0) exited non-zero on 'Zap/3-1'
-- Executing Set("Zap/3-1", "_direct_vm=4258") in new stack
-- Executing MailboxExists("Zap/3-1", "[EMAIL PROTECTED]|") in new stack
Aug 23 08:26:30 WARNING[8313]: app_voicemail.c:5697 vm_box_exists: VM
box [EMAIL PROTECTED] exists, but extension 04258, priority 103 doesn't exist
Is there a way to avoid this warming? Code fragment below:
[direct-to-voicemail]
; **
; Allow anybody to send a call directly to voicemail
; by pre-pending a 0 to the destination extension.
; Checks to see if voice mail box exists, if not
; Tells the callee that no such vm box exists and
; then transfers them to the operator
; **
exten => _04XXX,1,Set(_direct_vm=${EXTEN:1})
exten => _04XXX,2,MailboxExists([EMAIL PROTECTED])
exten => _04XXX,3,Goto(s-${VMBOXEXISTSSTATUS},1)
exten => s-FAILED,1,SayDigits(${direct_vm})
exten => s-FAILED,2,Playback(vm-nobox)
exten => s-FAILED,3,Playback(pbx-transfer)
exten => s-FAILED,4,Goto(incoming,s,1)
exten => s-SUCCESS,1,Set(CALLBACK=${DB(vmcallback/${direct_vm})})
exten => s-SUCCESS,2,GotoIf($["${CALLBACK}" =
"YES"]?s-SUCCESS,3:s-SUCCESS,4)
exten => s-SUCCESS,3,System(/usr/local/bin/vm-callout.sh ${direct_vm})
exten => s-SUCCESS,4,Voicemail([EMAIL PROTECTED])
exten => s-SUCCESS,5,Hangup()
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety."
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[asterisk-users] Call Handoff
Hi All I have 2 phones registered to an asterisk server. The phones are sat behind a NAT. If I have the asterisk sat inline on the call after setting it up (with transfer option specified as an example) the call works fine. If I take out all options so the asterisk should bridge the call and hand it off I get the phones ringing and then when the call is answered there is no media. I know this is probably somehting related to the NAT as to why the asterisk cant handoff the call but was wondering if anyone else has been able to over come this at all? Many Thanks SP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
