Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On 02:39, Fri 20 Oct 06, Tzafrir Cohen wrote: On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote: On 23:04, Thu 19 Oct 06, Vidar wrote: Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz Thanks for the information. It's a shame we need to read this here and not see it on their website. Also note that the changelog entry for 0.3.0-PRE-1u is missing from the CHANGES file. Nevertheless, that is a version for Asterisk 1.2.13 , Zaptel 1.2.10 and libpri 1.2.4 . Also note that the changelog mentioned -1t and that that file is also available on FTP. I think junghann.net needs a webmaster ;-) I'm off to run diff to make the changelog myself... -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] question about CDR command
I don't believe there is any quick simple way of doing this. You would need to add the column in the DB and modify cdr.c. I'm sure someone out there has a step by step doc on how to do this. You may try the #asterisk channel on irc. bp On 10/19/06, unplug [EMAIL PROTECTED] wrote: Thanks!!Just one more question.Can I do the same add fieldname=1 if I adda field fieldname in the cdr table to perform the same action? On 10/19/06, William Piper [EMAIL PROTECTED] wrote: In cdr_mysql.conf add userfield=1 under the globals setting. bp On 10/18/06, unplug [EMAIL PROTECTED] wrote: I want to set some custom data in the field of userfield in table CDR as following. exten = s,19,Set(CDR(userfield)=1234) exten = s,20,Dial(SIP/1234) However, the userfield doesn't get update after making the call. After that, I relocate the command as following. exten = s,19,Dial(SIP/1234) exten = s,20,Set(CDR(userfield)=1234) The userfield doens't get update at all.I don't know why the field can't update after issuing the command.Anyone can help? ___ --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 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
Re: [asterisk-users] Embedded Asterisk
Doh! Turns out it won't be November. It will be a bit later. Sorry. On Thursday 19 October 2006 21:35, Mike Diehl wrote: On Thursday 19 October 2006 14:10, Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. I happen to know that November's Linux Journal will have an article about running Linux/Asterisk on a Linsys WRTGS54SL router. Nothing too technical, but I hope you enjoy it. Mike Diehl. ___ --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] Re: Sip Trunks
On 2006-10-19 20:30:03 -0700, Martin Joseph [EMAIL PROTECTED] said: On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said: Hello, well, I need to configure two asterisk box like SIP trunks to se nd sip calls from one asterisk to the other and visceversa. So How I setup con fi g files to get this working?.Thanks. You can do it via IAX2, there was a recipe posted here very recently that made this quite simple. Plus IAX2 saves bandwidth for trunked calls. Woops! Wrong link, sorry... Try 2. http://astrecipes.net/index.php?n=204 ___ --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] vISDN, mISDN, bristuff [was: Re: Bristuff qozap drivers problem]
Same similar problems here, with qozap and bristuff 0.3.0: with physical media connected, I get a layer 1 down message that keeps rolling up EVEN DURING AN INCOMING CALL on that BRI span, and prevents asterisk from placing outbound calls... until I restart asterisk (luckily, no kernel crashes so far). Can anyone give me his opion/experience about vISDN or mISDN alternatives? Maybe I'm running bristuff in the wrong way, but I'm starting to think that bristuff is not the best way to make a stable ISDN PBX. I don't like very much the way bristuff kind of messes up with asterisk sourcecode (especially chan_zap/libpri). Steve Davies ha scritto: Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The flaw in the messages is the Alarm cleared message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All ports are in TE mode.) Can anyone suggest a cleanup in qozap.c that will prevent it telling Asterisk that the channel is up unless it actually has come back up? I do not understand the zaptel/bristuff internals well enough to be able to find where this is occuring. I also get a solid kernel crash with no Oops if I unload the qozap module - Again this does not happen in the older versions of the qozap module. I am using Kernel 2.6.10. Many thanks for any pointers, Steve. Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 4: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 5: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 5 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 7: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 7 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 8: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 8 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 10: No Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 10 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 11: No Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 11 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 4 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 5 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 7 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 8 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 10 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 11 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 3 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 3 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 12 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 12 as D-channel anyway! ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [asterisk-users] Bristuff qozap drivers problem
On Thu, Oct 19, 2006 at 01:42:01PM +0100, Steve Davies wrote: Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. Have you tried version = 0.3.0-PRE-1s? It seems to have made many such messages debug messages. -- 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] Help: Problems about console color (FC5, XTerm)
Hi, I have just installed asterisk under Fedora5, the program runs fine except the color is not correct, it's very dark, difficult tosee. And if I select a part that part's color will be fine at least clear for read purpose. All I can do now is disabling color by changing the TERM environment variable to another value. Any help is greatly appreciated !! Thanks in advance. (Fedora5, asterisk-1.4.0-beta3)Sincerely,zuobf ___ --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] Asterisk 1.2.13 make problem
Hi all I have downloaded 1.2.13 installing on my FC5 when iam making, iam getting the following error could some one suggest me the what is the problem make[1]: Entering directory `/root/vici/asterisk-1.2.13/apps'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=k8 -fomit-frame-pointer -fPIC -c -o app_voicemail.o app_voicemail.c app_voicemail.c: In function âsendmailâ:app_voicemail.c:1796: error: âVM_CONVERTMP3â undeclared (first use in this function)app_voicemail.c:1796: error: (Each undeclared identifier is reported only onceapp_voicemail.c:1796: error: for each function it appears in.) make[1]: *** [app_voicemail.o] Error 1make[1]: Leaving directory `/root/vici/asterisk-1.2.13/apps'make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: Problems about console color (FC5, XTerm)
On Fri, Oct 20, 2006 at 03:34:32PM +0800, zuo bf wrote: Hi, I have just installed asterisk under Fedora5, the program runs fine except the color is not correct, it's very dark, difficult to see. And if I select a part that part's color will be fine at least clear for read purpose. All I can do now is disabling color by changing the TERM environment variable to another value. Or use -n What colors would you suggest? Any help is greatly appreciated !! Thanks in advance. (Fedora5, asterisk-1.4.0-beta3) Is this xterm? (the real xterms, not rxvt or anything else)? Hold the CTRL key and clock with the middle button on the window. Select enable reverse window from the menu. Better? -- 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
Re: [asterisk-users] Embedded Asterisk
Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, OpenWRT -running on Linksys WRT- has asterisk packages. [EMAIL PROTECTED]:~# ipkg list | grep asterisk asterisk - 1.0.10-1 - An open source PBX asterisk-chan-mgcp - 1.0.10-1 - a Media Gateway Control Protocol implementation for Asterisk asterisk-chan-skinny - 1.0.10-1 - a Skinny Client Control Protocol implementation for Asterisk asterisk-codec-ilbc - 1.0.10-1 - an Internet Low Bitrate Codec (ILBC) Translator for Asterisk asterisk-codec-lpc10 - 1.0.10-1 - an LPC10 (Linear Predictor Code) 2.4kbps Voice Coder for Asterisk asterisk-codec-speex - 1.0.10-1 - a Speex/PCM16 Codec Translator for Asterisk asterisk-mini - 1.0.10-1 - A minimal open source PBX asterisk-mysql - 1.0.10-1 - MySQL modules for Asterisk asterisk-pbx-dundi - 1.0.10-1 - Distributed Universal Number Discovery (DUNDi) support for Asterisk asterisk-pgsql - 1.0.10-1 - PostgreSQL modules for Asterisk asterisk-res-agi - 1.0.10-1 - Asterisk Gateway Interface module asterisk-sounds - 1.0.10-1 - a sounds collection for Asterisk asterisk-voicemail - 1.0.10-1 - VoiceMail related modules for Asterisk -- Daniel ___ --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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
Hi Doug, I do not use extensions.conf so I cannot show anything but I can assure that I do not set the callerid except for parameters inside zapata.conf: usecallerid = yes callerid = asreceived Hope may help. TIA Giorgio Incantalupo Doug Lytle wrote: Giorgio Incantalupo wrote: Hi, I have a sangoma PRI card on an Asterisk PBX. I have problem with outgoing caller ID: when I make an outbound call, the called party gets x1 instead of x240 where x is the my company prefix and 240 is the phone extensions I call from. I read something about usecallingpres on wiki but nothing is told about default and possible values. Is this parameter the real cause of my wrong caller id? How are you setting the caller id before dialing? Show us the section of your dial plan that handles this. 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 ___ --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] Embedded Asterisk
On 19 Oct 2006, at 21:10, Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. I've been using an nslu2 (slug) as a lightweight asterisk server. It isn't a broadband router, but it is cheap and works well. I'd be happy to chat with anyone at Astricon about it. I'll be on Booth 118 launching Corraleta - our zero install web softphone (Funny thing, I found a couple of bugs in Corraleta that only showed up when testing against the slug - byte order things if I remember.) Tim Panton www.mexuar.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] /dev/zap/channel ownership
Hi Mitch, I have same problemsometime I get that error in particular when I modprobe module as root to fix asterisk wrong configurations but not when rebooting. To be sure I chown /dev/zap inside my /etc/init.d/asterisk launch script after modprobe-ing zaptel and wctdm. Giorgio Incantalupo Mitch Miller wrote: * is having permission problems accessing /dev/zap/channel. When I look, these devices (everything in /dev/zap) shows root.root for uid and gid. If I start Asterisk from the command line, it runs fine (running as Root). When I start it as a service, I get Oct 19 23:02:55 WARNING[10587] chan_iax2.c: Unable to open IAX timing interface: Permission denied Oct 19 23:02:55 WARNING[10587] chan_zap.c: Unable to open '/dev/zap/channel': Permission denied Oct 19 23:02:55 ERROR[10587] chan_zap.c: Unable to open channel 1: Permission denied So ... I changed ownership on /dev/zap/* to asterisk.asterisk and now everything seems to be running fine. My question is ... how would the ownership on these devices have changed? (I've not yet rebooted, but I'm suspicious that they'll revert back to root.root). -- Mitch ___ --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] using asterisk to do remote control functions
Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... Many Thanks -- Gregory Machin [EMAIL PROTECTED] www.linuxpro.co.za ___ --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] rxfax problem
Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldn’t hand shake with other side well. HTH. M. Shokuie Nia. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim McIver Sent: 2006/10/19 06:17 ب.ظ To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rxfax problem Did you ever get an answer to this problem ? I too am seeing this and it’s driving me mad !!! Jim ___ --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] Asterisk Realtime... Help Me!!!
Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make "odbc show" I see that the DB connection is "UP", but if I make "realtime load family column value" both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the "realtime load" command? Please, help me! Maury ___ --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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
On 201006, 10:06, Giorgio Incantalupo wrote: Hi Doug, I do not use extensions.conf so I cannot show anything but I can assure that I do not set the callerid except for parameters inside zapata.conf: usecallerid = yes callerid = asreceived I guess the problem is at the telco's side, since the CLI that is shown seems to be the first one of the numbering scheme. I suppose what you should do is to call them up and ask them to open the CLI presentation for all your numbering scheme. If you need any more help, feel free to ask. Ciao -- Massimiliano Stucchi, CTO Director of Operations WillyStudios.com - IT Consulting, Web and VoIP Services [EMAIL PROTECTED] | Tel (+39) 0244417203 | Fax (+39) 0244417204 IT-20040, Carnate (Milano), via Carducci 9 ___ --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] plainvoip - down ???
Oh this brings back memories. - Original Message - From: Andres [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:44 AM Subject: Re: [asterisk-users] plainvoip - down ??? Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails This is starting to sound like a rerun of Livevoip. Remember that company? ___ --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] Asterisk Realtime... Help Me!!!
Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury ___ --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 paste your relevant config files and also an example command (realtime load etc) that you are using. also.. if u can.. turn on logging(DEBUG) in logger.conf, or better still, go change the code n put in ur own debug lines duznt take too long to figure out where u r going wrong. - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury ___ --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 paste your relevant config files and also an example command (realtime load etc) that you are using. also.. if u can.. turn on logging(DEBUG) in logger.conf, or better still, go change the code n put in ur own debug lines duznt take too long to figure out where u r going wrong. - Ben ___ --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] Asterisk Realtime... Help Me!!!
I'm having the same issue overhere with a nearly identical config: res_odbc.conf: [mysql2] enabled = yes dsn = MySQL-asterisk username = asterisk password = asterisk pre-connect = yes extconfig.conf [settings] sipusers = odbc,MySQL-asterisk,sip_buddies sippeers = odbc,MySQL-asterisk,sip_buddies sip_buddies is identical to what is described at http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip ++--+-+--+---+---+-++---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+---+-+++--+++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context| defaultip | dtmfmode | fromuser | fromdomain | fullcontact | host| insecure | language | mailbox | md5secret | nat | deny | permit | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | musiconhold | regseconds | ipaddr | regexten | cancallforward | setvar | ++--+-+--+---+---+-++---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+---+-+++--+++ | 1 | 1006 | NULL| NULL | 2 | Test account 1006 | no | ciscophone | NULL | NULL | NULL | NULL | NULL| dynamic | NULL | NULL | NULL| NULL | no | NULL | NULL | NULL | 2 | | NULL| NULL | NULL | NULL | 1234 | friend | 1006 | all | alaw | NULL| 0 || | yes || ++--+-+--+---+---+-++---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+---+-+++--+++ modules.conf includes: preload = res_odbc.so preload = res_config_odbc.so When I execute odbc show, I can see a query coming in from asterisk in the mysql query log, thus the odbc connection mysql work. 061020 13:25:47 10 Connect [EMAIL PROTECTED] on asterisk 061020 13:27:03 10 Query select 1 Other than that, I have the same problem as Maurizio Pederneschi. *CLI realtime load sipusers username 1006 No rows found matching search criteria. Same DB problem occurs when I register the 1006 phone: *CLI Oct 20 13:29:58 NOTICE[32135]: chan_sip.c:11084 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.16.5.206' - Username/auth name mismatch At that time I see no incoming query whatsoever passing in the MySQL log. I'm running debian/unstable asterisk package 1.2.12.1.dfsg-1. Perhaps it is a problem with just this release? greetings Tijl Van den Broeck On 10/20/06, Maurizio Pederneschi [EMAIL PROTECTED] wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using
[asterisk-users] Xorcom Astribank
Hello, I want to use Astribank from Xorcom, has anybody some experience or references with it? Sincerely, Stepan -- tel./fax: +420 552 305 306 email: [EMAIL PROTECTED] www: http://www.ha-vel.cz ___ --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 to do remote control functions
Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... Many Thanks Hi, yes it is possible using AGI scripts ! Greg ___ --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: echotraining=yes in misdn.conf is invalid or out of range.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Jarkko, I had the same problem..It worked with an old version of misdn-install (taken from beronet site) but not with actual mqueue-misdn-install. I tried to put it in every misdn.conf section I have without success. The updated beronet install manual doesn't mention that parameter anymore so I removed it from misdn.conf. I have also BeroNet card but I'm unable to start Asterisk with chan_misdn. This is the error that I get on CLI. Oct 18 15:10:21 ERROR[5860] chan_misdn.c: Unable to initialize mISDN Oct 18 15:10:21 WARNING[5860] loader.c: chan_misdn.so: load_module failed, returning -1 Oct 18 15:10:21 VERBOSE[5860] chan_misdn.c: -- Unregistering mISDN Channel Driver -- Oct 18 15:10:21 WARNING[5860] loader.c: Loading module chan_misdn.so failed! And I have started misdn-init start [EMAIL PROTECTED] ~]# /etc/init.d/misdn-init start which: no lsusb in (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/lo cal/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/root/bin) [!!] FATAL: lsusb not in path, please install. - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x4 protocol=0x2,0x2,0x2,0x2 layermask=0 xf,0xf,0xf,0xf poll=128 debug=0 modprobe mISDN_dsp debug=0x0 options=0 poll=128 dtmftreshold=100 [i] creating device node: /dev/mISDN And I believe I have all modules loaded: [EMAIL PROTECTED] ~]# lsmod Module Size Used by mISDN_dsp 202764 0 mISDN_capi103180 0 l3udss145020 0 mISDN_l2 41812 0 mISDN_l1 12732 0 capi 18049 0 capifs 5961 2 capi kernelcapi 46689 2 mISDN_capi,capi md5 4033 1 ipv6 266433 10 parport_pc 28805 0 lp 13001 0 parport39689 2 parport_pc,lp autofs427333 2 rfcomm 42589 0 l2cap 30021 5 rfcomm bluetooth 55109 4 rfcomm,l2cap sunrpc162821 1 ztdummy 3924 0 wcusb 19488 0 wctdm 35392 0 wcfxo 13216 0 wctdm24xxp120384 0 wcte11xp 36384 0 wct1xxp2 0 wct4xxp 312000 0 tor2 92704 0 zaptel206468 9 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wc t1xxp,wct4xxp,tor2 crc_ccitt 2113 1 zaptel video 15941 0 button 6609 0 battery 9413 0 ac 4805 0 ohci1394 39817 0 ieee1394 304057 1 ohci1394 uhci_hcd 34897 0 ehci_hcd 39757 0 shpchp 91205 0 i2c_viapro 8145 0 i2c_core 21825 1 i2c_viapro hfcmulti 79144 0 mISDN_core 79840 6 mISDN_dsp,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1,h fcmulti via_rhine 27465 0 mii 5569 1 via_rhine dm_snapshot17669 0 dm_zero 2113 0 dm_mirror 25261 0 ext3 132297 2 jbd79449 1 ext3 dm_mod 58997 6 dm_snapshot,dm_zero,dm_mirror Do you know what could be the problem? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Linksys PAP2 dial plan help please
Hi,I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345.Can someone help me to add this dialplan.Thanks in advanceDan ___ --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] call center status viewer
Can anyone point me in the direction of a good status viewer for agents. I have looked at the voip-info wiki and saw some good commercial ones. I just need opinions on any products. I am currently using FOP. I am looking for login/out, ringing, hangup and the like. I do strictly monitor agent proxies and not actual devices.___ --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 to do remote control functions
HiIm very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the userdials a predefined number ...And what hardware is required ...We use it in a variety of situations in an industrial setting. We use it for some control, status checking, and also we use asterisk for overhead paging when certain alert situations occur. The hardware I'm using is Opto22 Snap ehternet brains. One of the models I use is the SNAP-B3000-ENET which can sometimes be found on Ebay. The hardware isn't cheap but they provide excellent documentation and software for free, and free training. They have a free linux developers kit which has, as an example program, a handy command line utility which I use with asterisk. If you just want to control a couple of digital points this hardware may be overkill, but it is cool stuff.Marv Horst ___ --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] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
Hi all after patching to my asterisk when iam try to make, iam getting the following error GNU_SOURCE -O6 -march=k8 -fomit-frame-pointer -fPIC -c -o app_voicemail.o app_voicemail.capp_voicemail.c: In function âsendmailâ:app_voicemail.c:1796: error: âVM_CONVERTMP3â undeclared (first use in this function) app_voicemail.c:1796: error: (Each undeclared identifier is reported only onceapp_voicemail.c:1796: error: for each function it appears in.)make[1]: *** [app_voicemail.o] Error 1make[1]: Leaving directory `/root/vici/asterisk- 1.2.12.1/apps'make: *** [subdirs] Error 1 Ram On 10/16/06, Marco Mouta [EMAIL PROTECTED] wrote: Error syntax:is Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gain On 10/16/06, Marco Mouta [EMAIL PROTECTED] wrote: Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough. Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies On 10/16/06, kjcsb [EMAIL PROTECTED] wrote: The problem is:Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. There's a patch for this: http://bugs.digium.com/file_download.php?file_id=10824type=bug Cameron ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta -- Com os melhores cumprimentos,Marco Mouta ___--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] Linksys PAP2 dial plan help please
Just adapt the Dial line you use like this
Dial(SIP/22${EXTEN})
On 10/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to
add 2 characters in front of the dialled number always when it send the call
to my asterisk. I dont know how to do that. I will summarise my requirement.
My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345.
Can someone help me to add this dialplan.
Thanks in advance
Dan
___
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Re: [asterisk-users] plainvoip - down ???
- Original Message - From: J. Oquendo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 19, 2006 3:22 PM Subject: Re: [asterisk-users] plainvoip - down ??? Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails I can get there just fine. Your routes might be toasted [EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com PING plainvoip.com (66.199.240.2) 56(84) bytes of data. ... --- plainvoip.com ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9013ms rtt min/avg/max/mdev = 75.531/78.550/80.349/1.418 ms, pipe 2 Depending on your location thought, there are issues with GBLX possibly due to a fiber cut either in VA or DC. No, the service is down. If you turn asterisk off, isn't your box still pingable? ___ --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] rxfax problem
M. Shokuie Nia wrote: Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldn’t hand shake with other side well. rxfax isn't sensitive to noise at all. At a gain of 12 you've caused overloading and distortion, and the signal cannot be decoded. Many people seem to be nearly deaf. They run systems at massive gain with awful distortion, and seem content until they find something like a modem or DTMF detection doesn't work too well. Steve ___ --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 to do remote control
If you just want to control a couple of digital points this hardware may be overkill, but it is cool stuff. For smaller implementations you can just use the outbound control lines (DTR RTS) on an RS232C port. That can give you control of two on/off devices. They only sink about 20ma so isolate them with a solid state relay or something. A C program to turn on/off is fairly trivial and run it from AGI. I don't want to clutter the list with code but I can supply if anyone needs it. dbc. -- David Cook ___ --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 to do remote control functions
Thank you for sharing this information .. Many Thanks , have a grate day :-) On 10/20/06, marvin horst [EMAIL PROTECTED] wrote: Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... We use it in a variety of situations in an industrial setting. We use it for some control, status checking, and also we use asterisk for overhead paging when certain alert situations occur. The hardware I'm using is Opto22 Snap ehternet brains. One of the models I use is the SNAP-B3000-ENET which can sometimes be found on Ebay. The hardware isn't cheap but they provide excellent documentation and software for free, and free training. They have a free linux developers kit which has, as an example program, a handy command line utility which I use with asterisk. If you just want to control a couple of digital points this hardware may be overkill, but it is cool stuff. Marv Horst ___ --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 -- Gregory Machin [EMAIL PROTECTED] www.linuxpro.co.za ___ --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: echotraining=yes in misdn.conf is invalid or out of range.
Hi Tomislav, may sound stoopid but have you checked if /usr/lib/asterisk/modules/chan_misdn.so is present? Giorgio Incantalupo Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Jarkko, I had the same problem..It worked with an old version of misdn-install (taken from beronet site) but not with actual mqueue-misdn-install. I tried to put it in every misdn.conf section I have without success. The updated beronet install manual doesn't mention that parameter anymore so I removed it from misdn.conf. I have also BeroNet card but I'm unable to start Asterisk with chan_misdn. This is the error that I get on CLI. Oct 18 15:10:21 ERROR[5860] chan_misdn.c: Unable to initialize mISDN Oct 18 15:10:21 WARNING[5860] loader.c: chan_misdn.so: load_module failed, returning -1 Oct 18 15:10:21 VERBOSE[5860] chan_misdn.c: -- Unregistering mISDN Channel Driver -- Oct 18 15:10:21 WARNING[5860] loader.c: Loading module chan_misdn.so failed! And I have started misdn-init start [EMAIL PROTECTED] ~]# /etc/init.d/misdn-init start which: no lsusb in (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/lo cal/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/root/bin) [!!] FATAL: lsusb not in path, please install. - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x4 protocol=0x2,0x2,0x2,0x2 layermask=0 xf,0xf,0xf,0xf poll=128 debug=0 modprobe mISDN_dsp debug=0x0 options=0 poll=128 dtmftreshold=100 [i] creating device node: /dev/mISDN And I believe I have all modules loaded: [EMAIL PROTECTED] ~]# lsmod Module Size Used by mISDN_dsp 202764 0 mISDN_capi103180 0 l3udss145020 0 mISDN_l2 41812 0 mISDN_l1 12732 0 capi 18049 0 capifs 5961 2 capi kernelcapi 46689 2 mISDN_capi,capi md5 4033 1 ipv6 266433 10 parport_pc 28805 0 lp 13001 0 parport39689 2 parport_pc,lp autofs427333 2 rfcomm 42589 0 l2cap 30021 5 rfcomm bluetooth 55109 4 rfcomm,l2cap sunrpc162821 1 ztdummy 3924 0 wcusb 19488 0 wctdm 35392 0 wcfxo 13216 0 wctdm24xxp120384 0 wcte11xp 36384 0 wct1xxp2 0 wct4xxp 312000 0 tor2 92704 0 zaptel206468 9 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wc t1xxp,wct4xxp,tor2 crc_ccitt 2113 1 zaptel video 15941 0 button 6609 0 battery 9413 0 ac 4805 0 ohci1394 39817 0 ieee1394 304057 1 ohci1394 uhci_hcd 34897 0 ehci_hcd 39757 0 shpchp 91205 0 i2c_viapro 8145 0 i2c_core 21825 1 i2c_viapro hfcmulti 79144 0 mISDN_core 79840 6 mISDN_dsp,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1,h fcmulti via_rhine 27465 0 mii 5569 1 via_rhine dm_snapshot17669 0 dm_zero 2113 0 dm_mirror 25261 0 ext3 132297 2 jbd79449 1 ext3 dm_mod 58997 6 dm_snapshot,dm_zero,dm_mirror Do you know what could be the problem? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] call center status viewer
Yeah, try the Flash Operator Panel. You can view/download it at: http://www.asternic.org/ Good luck! Jordan Novak wrote: Can anyone point me in the direction of a good status viewer for agents. I have looked at the voip-info wiki and saw some good commercial ones. I just need opinions on any products. I am currently using FOP. I am looking for login/out, ringing, hangup and the like. I do strictly monitor agent proxies and not actual devices. ___ --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] Linksys PAP2 dial plan help please
Thanks Tijl,That was a nice one but i would like to have my PAP2 programmed with this dialplan.This way i can program it for others PAP2's too. I want to have the PAP2 dialplan help not the asterisk dialplan. My PAP2 should send the 55-1-210-1234345 this way rather than 1-210-1234345.
ThanksOn 20/10/06, Tijl Van den Broeck [EMAIL PROTECTED] wrote:
Just adapt the Dial line you use like thisDial(SIP/22${EXTEN})On 10/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement.
My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan
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Re: [asterisk-users] Bristuff qozap drivers problem
On 10/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Oct 19, 2006 at 01:42:01PM +0100, Steve Davies wrote: Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. Have you tried version = 0.3.0-PRE-1s? Yes, I tried it before I posted the message - I believe this is an intentional change in the alerting, but I am concerned that it shows Alarm cleared which is NOT correct. The cable is still unplugged, and the alarm has not cleared. The latest (PRE-1u) also still crashes on unload :( Regards, Steve ___ --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] considering purchasing a t1 card, any recommendations?
Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? Thanks in advance. R.R. Libera Zoa escribió: I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855
Re: [asterisk-users] Linksys PAP2 dial plan help please
Friday, October 20, 2006, 2:09:56 PM, [EMAIL PROTECTED] wrote: My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. you must add :55 before every rule, where you want to add 55. eg. this rule matches your example 1-210-1234345 and adds 55 before the dialed number: :551210xxx -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 dial plan help please
On Fri, Oct 20, 2006 at 03:09:56PM +0300, [EMAIL PROTECTED] wrote:
I have a Linksys PAP2-NA connectd to my asterisk. I would like the
device to add 2 characters in front of the dialled number always when
it send the call to my asterisk. I dont know how to do that. I will
summarise my requirement.
My friend dials 1-210-1234345, i want the asterisk to get
55-1-210-1234345.
What actual problem are you trying to solve?
It's very likely that you can solve it better in Asterisk without touching
the ATA.
For example, if what you actually want is for calls from the PAP2 which
Asterisk routes outwards via a POTS card to be prefixed with 55, you can do
that in your dialplan (extensions.conf) by modifying the rule for placing
outbound calls.
[internal]
exten = _1.,1,Dial(Zap/4/55${EXTEN})
^^
Brian.
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Re: [asterisk-users] Linksys PAP2 dial plan help please
http://www.netphonedirectory.com/pap2_dialplan.htm
might help
remember google?
Bails
[EMAIL PROTECTED] wrote:
Thanks Tijl,
That was a nice one but i would like to have my PAP2 programmed with
this dialplan.
This way i can program it for others PAP2's too. I want to have the PAP2
dialplan help not the asterisk dialplan. My PAP2 should send the
55-1-210-1234345 this way rather than 1-210-1234345.
Thanks
On 20/10/06, *Tijl Van den Broeck* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Just adapt the Dial line you use like this
Dial(SIP/22${EXTEN})
On 10/20/06, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
I have a Linksys PAP2-NA connectd to my asterisk. I would like
the device to
add 2 characters in front of the dialled number always when it
send the call
to my asterisk. I dont know how to do that. I will summarise my
requirement.
My friend dials 1-210-1234345, i want the asterisk to get
55-1-210-1234345.
Can someone help me to add this dialplan.
Thanks in advance
Dan
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Re: [asterisk-users] /dev/zap/channel ownership
On Thu, Oct 19, 2006 at 11:08:09PM -0500, Mitch Miller wrote: * is having permission problems accessing /dev/zap/channel. When I look, these devices (everything in /dev/zap) shows root.root for uid and gid. If I start Asterisk from the command line, it runs fine (running as Root). When I start it as a service, I get Oct 19 23:02:55 WARNING[10587] chan_zap.c: Unable to open '/dev/zap/channel': Permission denied Oct 19 23:02:55 ERROR[10587] chan_zap.c: Unable to open channel 1: Permission denied Please see README.udev of zaptel. Basically, those files are generated by udev. You might as well tell udev to chown them to asterisk.asterisk (or root.dialout, the standard on Debian systems) The default permissions.rules file on Debian Etch now contains: SUBSYSTEM==zaptel,GROUP=dialout A more complete rule would be: But you may choose to use: SUBSYSTEM==zaptel, MODE=0660, USER=asterisk, GROUP=asterisk BTW: that line is missing from the udev package in Debian Sarge, leading to a similar problem to the one described here once the uder decides to use udev. -- 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] Astricon - post show Saturday?
Is anyone on this list familiar with Dallas? Anyone want to recommend something to do on the Saturday/Sunday? Never been to Dallas so Im hoping for a restaurant recommendation for Saturday night.(somewhere a little more up market would be good) also any sights that have to be visited while in town? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 dial plan help please
Hi,Thanks once again,Let me put is clear. I'm using TRIXBOX which many out here feel - its for kids- but i do like it an use it for sometime.As all i do have extensions and trunks configured on it. I want one of my extensions to use a particular outbound route only. I have rightly done all the setup in the extensions_custom.conf. Hence i do want the 55 to be an identifier for his route hence the 55 will be sent from the PAP2 without the user knowing the setup itself.
Hope you got what i intend. I don't know if there is any other solution other than this as all user will be using the same number format to dial.Thanks for the support.Dan
On 20/10/06, Brian Candler [EMAIL PROTECTED] wrote:
On Fri, Oct 20, 2006 at 03:09:56PM +0300, [EMAIL PROTECTED] wrote:I have a Linksys PAP2-NA connectd to my asterisk. I would like thedevice to add 2 characters in front of the dialled number always when
it send the call to my asterisk. I dont know how to do that. I willsummarise my requirement.My friend dials 1-210-1234345, i want the asterisk to get55-1-210-1234345.
What actual problem are you trying to solve?It's very likely that you can solve it better in Asterisk without touchingthe ATA.For example, if what you actually want is for calls from the PAP2 which
Asterisk routes outwards via a POTS card to be prefixed with 55, you can dothat in your dialplan (extensions.conf) by modifying the rule for placingoutbound calls.[internal]exten = _1.,1,Dial(Zap/4/55${EXTEN})
^^Brian.
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Re: [asterisk-users] Astricon - post show Saturday?
Title: Re: [asterisk-users] Astricon - post show Saturday? I really recommend Pappas Brothers Steakhouse. My wife and I went there for our first anniversary, and it was really nice. Beyond that, check out the west end, its pretty nice place to be for the night scene. On 10/20/06 8:45 AM, Dean Collins [EMAIL PROTECTED] wrote: Is anyone on this list familiar with Dallas? Anyone want to recommend something to do on the Saturday/Sunday? Never been to Dallas so Im hoping for a restaurant recommendation for Saturday night.(somewhere a little more up market would be good) also any sights that have to be visited while in town? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb ___ --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 -- James Texter ___ --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] noise gate for asterisk?
Hi list, I have a client with a strange requirement: putting a noise gate on the Asterisk channel. For those who are not familiar with them, noise gates are used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of the channel, and when it's too low they just let the channel close, i.e send perfect silence, therefore killing low-level buzzing sounds. My client has such a need because they have analog voice-operated push-to-talk half-duplex devices on the other side, and low level noise from the Asterisk side will keep the channel open. I will try diminishing the TXgain, but I wondered if there were other options too. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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] considering purchasing a t1 card, any recommendations?
I've been using the Digium TE110P for more than a year now without any hitches in a mission critical environment. It's in a Dell Poweredge 750. I had a problem compiling libpri at first and the digium help desk was more than helpful. I also had some echo issues, but those have mostly been resolved using the built in cancellation routines. -- Tom Hayden On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote: Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? Thanks in advance. R.R. Libera Zoa escribió: I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com
Re: [asterisk-users] Polycom boot error
The correct way to perform a factory reset on the Polycom phone is documented in one of our knowledge-base Articles, number: KB-001032http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-13.htmYou want to do a "factory format" to COMPLETELY erase everything on the phone.After this is done you can have the Polycom phone get the latest bootrom (important for this type of error) and firmware from your FTP or TFTP server set up at your office. If you don't have one, nows the time to learn (admin guide and normal IT knowledge), get one and use that to get the latest bootrom (3.2.2) and firmware (2.0.1 or 1.6.7 if you don't want to attempt version 2 yet) on your phones.That should resolve your problem.You can get the bootrom and firmware from your service provider, distributor, or reseller as long as they are Polycom certified.Hope that helps. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 19, 2006, at 10:26 PM, Neider, Clint wrote: I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message-From: Dovid Bender [mailto:asteriskusers at dovid.net]Sent: Tuesday, August 15, 2006 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom upgrade issue I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt mailto:cshaffer at gmail.com Shaffer To: 'Asterisk Users Mailing List - mailto:asterisk-users at lists.digium.com Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PMSubject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt Clint NeiderEmail Administrator[EMAIL PROTECTED]Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | This email message is intended only for the addressee(s) and contains information that may be confidential and/or copyright. If you are not the intended recipient please notify the sender by reply email and immediately delete this email. Use, disclosure or reproduction of this email by anyone other than the intended recipient(s) is strictly prohibited. No representation is made that this email or any attachments are free of viruses. Virus scanning is recommended and is the responsibility of the recipient. ___--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
Re: [Asterisk-Users] rxfax problem
On 10/20/06, Steve Underwood [EMAIL PROTECTED] wrote: M. Shokuie Nia wrote: Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldn't hand shake with other side well. rxfax isn't sensitive to noise at all. At a gain of 12 you've caused overloading and distortion, and the signal cannot be decoded. Many people seem to be nearly deaf. They run systems at massive gain with awful distortion, and seem content until they find something like a modem or DTMF detection doesn't work too well. Steve Well Steve, you should be proud of your latest 0.0.3 snapshot code (20061012 ?) - It has solved all of our faxing issues here, even those that have been ongoing for almost a year with really flakey cheap multifunction fax machines... Thank you. Steve ___ --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] Getting started with sample dial plans
Now I'm ready to begin playing with dial plans and am having a difficult time getting started. You may want to read the book : http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 That should help you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 dial plan help please
On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote: As all i do have extensions and trunks configured on it. I want one of my extensions to use a particular outbound route only. I have rightly done all the setup in the extensions_custom.conf. Hence i do want the 55 to be an identifier for his route hence the 55 will be sent from the PAP2 without the user knowing the setup itself. Hope you got what i intend. I don't know if there is any other solution other than this as all user will be using the same number format to dial. Depending on whether this is a *preference* or a *requirement*, imposing it at the ata may not be good enough. Remember the lesson learned from Quake: impose your security at the server; the clients aren't under your control. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] considering purchasing a t1 card, any recommendations?
On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote: Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? (Hopefully this does not oversimplify) Fundamentally, since Digium updated the Hardware E/C on their recent boards, there is little difference between the PRI-side electronics of the Sangoma or the Digium cards (at least, not which would worry the end user) The remaining H/W difference is the PCI interface hardware. Sangoma have far more hardware development resource, to their PCI and now their PCI-e interfaces are more compatible with variations in motherboards. Digium are a software company, so the software support for their cards will be easier to get going as it is built in - On the other hand, Sangoma's build environment has come a long way, and their support IS good. As far as do I need echo cancellation is concerned. I would say that if you can afford it, and it is available, then buy it. It generally saves a lot of heartache in the longrun, regardless of the manufacturer. As you say, the Sangoma single-port cards do not have EC, so that decides the issue :) When using software EC, the MG2 echo canceller in the 1.2.x release is good. We have almost no problems on PRI circuits. I am not as confident of the 1.4.x version of the EC, but time will tell I imagine. Hope that helps. Steve ___ --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] Embedded Asterisk
Tim Panton wrote: On 19 Oct 2006, at 21:10, Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. I've been using an nslu2 (slug) as a lightweight asterisk server. It isn't a broadband router, but it is cheap and works well. I'd be happy to chat with anyone at Astricon about it. I'll be on Booth 118 launching Corraleta - our zero install web softphone (Funny thing, I found a couple of bugs in Corraleta that only showed up when testing against the slug - byte order things if I remember.) Tim Panton www.mexuar.com Tim, How do you use a web softphone on a slug? P.S. - I'm sure we'll have a chance to talk about it. I'm in booth 116/117 - howdy neighbor! -- 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] Embedded Asterisk
Administrator TOOTAI wrote: Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, OpenWRT -running on Linksys WRT- has asterisk packages. [EMAIL PROTECTED]:~# ipkg list | grep asterisk asterisk - 1.0.10-1 - An open source PBX asterisk-chan-mgcp - 1.0.10-1 - a Media Gateway Control Protocol implementation for Asterisk asterisk-chan-skinny - 1.0.10-1 - a Skinny Client Control Protocol implementation for Asterisk asterisk-codec-ilbc - 1.0.10-1 - an Internet Low Bitrate Codec (ILBC) Translator for Asterisk asterisk-codec-lpc10 - 1.0.10-1 - an LPC10 (Linear Predictor Code) 2.4kbps Voice Coder for Asterisk asterisk-codec-speex - 1.0.10-1 - a Speex/PCM16 Codec Translator for Asterisk asterisk-mini - 1.0.10-1 - A minimal open source PBX asterisk-mysql - 1.0.10-1 - MySQL modules for Asterisk asterisk-pbx-dundi - 1.0.10-1 - Distributed Universal Number Discovery (DUNDi) support for Asterisk asterisk-pgsql - 1.0.10-1 - PostgreSQL modules for Asterisk asterisk-res-agi - 1.0.10-1 - Asterisk Gateway Interface module asterisk-sounds - 1.0.10-1 - a sounds collection for Asterisk asterisk-voicemail - 1.0.10-1 - VoiceMail related modules for Asterisk Daniel, Those are ancient! Capouch has MUCH newer packages. -- 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] Polycom boot error
Hi Jesse, Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto Polycom IP500s? Or are they only for the later models? Do you know if you can still use TFTP for these software updates? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Friday, 20 October 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom boot error The correct way to perform a factory reset on the Polycom phone is documented in one of our knowledge-baseArticles, number:KB-001032 http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-13.htm You want to do a factory format to COMPLETELY erase everything on the phone. After this is done you can have the Polycom phone get the latest bootrom (important for this type of error) and firmware from your FTP or TFTP server set up at your office. If you don't have one, nows the time to learn (admin guide and normal IT knowledge), get one and use that to get the latest bootrom (3.2.2) and firmware (2.0.1 or 1.6.7 if you don't want to attempt version 2 yet) on your phones. That should resolve your problem. You can get the bootrom and firmware from your service provider, distributor, or reseller as long as they are Polycom certified. Hope that helps. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.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] voicemail usernames can't begin with j letter?
Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with j letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of Asterisk bug, or am I skipping some configuration? How can I make things work fine? Thanks in advance, Ricardo. ___ --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 to do remote control functions
Has anyone used the TrixBox/AAH builtin facility xPL for facility (including home/office/industrial) automation? On Fri, 2006-10-20 at 05:17 -0700, [EMAIL PROTECTED] wrote: Date: Fri, 20 Oct 2006 11:28:51 +0200 From: Gregory Machin [EMAIL PROTECTED] Subject: [asterisk-users] using asterisk to do remote control functions To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8; format=flowed Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... Many Thanks -- Gregory Machin -- (C) Matthew Rubenstein ___ --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] /dev/zap/channel ownership
Exactly what I was looking for. Thanks for the info. Going to go study now ... -- Mitch Tzafrir Cohen wrote: On Thu, Oct 19, 2006 at 11:08:09PM -0500, Mitch Miller wrote: * is having permission problems accessing /dev/zap/channel. When I look, these devices (everything in /dev/zap) shows root.root for uid and gid. If I start Asterisk from the command line, it runs fine (running as Root). When I start it as a service, I get Oct 19 23:02:55 WARNING[10587] chan_zap.c: Unable to open '/dev/zap/channel': Permission denied Oct 19 23:02:55 ERROR[10587] chan_zap.c: Unable to open channel 1: Permission denied Please see README.udev of zaptel. Basically, those files are generated by udev. You might as well tell udev to chown them to asterisk.asterisk (or root.dialout, the standard on Debian systems) The default permissions.rules file on Debian Etch now contains: SUBSYSTEM==zaptel,GROUP=dialout A more complete rule would be: But you may choose to use: SUBSYSTEM==zaptel, MODE=0660, USER=asterisk, GROUP=asterisk BTW: that line is missing from the udev package in Debian Sarge, leading to a similar problem to the one described here once the uder decides to use udev. ___ --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] #Transfer - Timeout is configurable?
Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison Transfer the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call This is timeout after pressing the first digit isn't it? -- best regards, Marco Mouta ___ --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] Embedded Asterisk
Hi Kristian, http://www.voip-info.org/wiki/view/Mexuar :) see you at the show Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- Tim, How do you use a web softphone on a slug? P.S. - I'm sure we'll have a chance to talk about it. I'm in booth 116/117 - howdy neighbor! -- 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 ___ --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] considering purchasing a t1 card, any recommendations?
Thanks Steve, it was helpful to read your post. Neither Digium or Sangoma single span cards have built in E/C, I´m wrong? To get one with this feature enable is completely out of my budget; since it is no for commercial use. I hope I wont have echo problems or they can be solve by means of software parameters. Thanks a lot. R.R. Libera Steve Davies escribió: On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote: Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? (Hopefully this does not oversimplify) Fundamentally, since Digium updated the Hardware E/C on their recent boards, there is little difference between the PRI-side electronics of the Sangoma or the Digium cards (at least, not which would worry the end user) The remaining H/W difference is the PCI interface hardware. Sangoma have far more hardware development resource, to their PCI and now their PCI-e interfaces are more compatible with variations in motherboards. Digium are a software company, so the software support for their cards will be easier to get going as it is built in - On the other hand, Sangoma's build environment has come a long way, and their support IS good. As far as do I need echo cancellation is concerned. I would say that if you can afford it, and it is available, then buy it. It generally saves a lot of heartache in the longrun, regardless of the manufacturer. As you say, the Sangoma single-port cards do not have EC, so that decides the issue :) When using software EC, the MG2 echo canceller in the 1.2.x release is good. We have almost no problems on PRI circuits. I am not as confident of the 1.4.x version of the EC, but time will tell I imagine. Hope that helps. Steve ___ --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] Clicking Noise on Pure Voip Calls
Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some assistance? Thanks, Taf.. Send instant messages to your online friends http://uk.messenger.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] Linksys PAP2 dial plan help please
On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote: Let me put is clear. I'm using TRIXBOX which many out here feel - its for kids- but i do like it an use it for sometime. As all i do have extensions and trunks configured on it. I want one of my extensions to use a particular outbound route only. In normal Asterisk, you'd solve that by putting that phone in its own context. I don't know if Trixbox cripples Asterisk so much that you can't do that. ___ --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] voicemail usernames can't begin with j letter?
What version of * are you running? I have several j usernames in voicemail.conf under SVN-branch-1.2-r37458M. On 10/20/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:Dear all, I've configured Asterisk Voicemail, but after some tests I realised thatwhen some call is sent to the voicemail of someone which username beginswith j letter,Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry invoicemail config file for 'ohn'(for a called user named john, for example)Is this some kind of Asterisk bug, or am I skipping some configuration? How can I make things work fine?Thanks in advance,Ricardo.___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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
Re: [asterisk-users] Polycom boot error
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote: Hi Jesse, Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto Polycom IP500's? Or are they only for the later models? Do you know if you can still use TFTP for these software updates? They are compatible with the new bootroms and firmwares according to Polycom's release notes, but you cannot use the HTTPS provisioning on the IP500 I believe. -- 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] Embedded Asterisk
Dean Collins wrote: Hi Kristian, http://www.voip-info.org/wiki/view/Mexuar :) see you at the show Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). Arg! I understand what a soft phone is. I know what java is. I also know that neither have anything to do with the slug. The slug cannot support any web browsers that support java. The slug doesn't have any audio interfaces that could support audio for a softphone - so again, what does the slug have to do with a java softphone?!?!? You mentioned that compiling for the slug helped resolve some issues for you. What are you compiling, and how does it run on the slug? -- 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
[asterisk-users] getting DID info..
This might be a newbie question... I'm using a SIP trunk and trying
to get DID line information on an incoming call. All I hear is a
nice lady saying 'Zero' - then the call continues... Any suggestions?
thanks
Todd
exten = s,n,Set(DIDID=(${FROM_DID}))
exten = s,n,SayNumber(DIDID)
or
exten = s,n,Set(FROM_DID=${EXTEN})
exten = s,n,SayNumber(FROM_DID)
and a third try.. (I'm not sure what 's' is, but saw it somewhere..)
exten = s,n,Set(FROM_DID=s)
exten = s,n,Wait(1)
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Re: [asterisk-users] Embedded Asterisk
Kristian Kielhofner wrote: Administrator TOOTAI wrote: Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, OpenWRT -running on Linksys WRT- has asterisk packages. [EMAIL PROTECTED]:~# ipkg list | grep asterisk asterisk - 1.0.10-1 - An open source PBX asterisk-chan-mgcp - 1.0.10-1 - a Media Gateway Control Protocol implementation for Asterisk asterisk-chan-skinny - 1.0.10-1 - a Skinny Client Control Protocol implementation for Asterisk asterisk-codec-ilbc - 1.0.10-1 - an Internet Low Bitrate Codec (ILBC) Translator for Asterisk asterisk-codec-lpc10 - 1.0.10-1 - an LPC10 (Linear Predictor Code) 2.4kbps Voice Coder for Asterisk asterisk-codec-speex - 1.0.10-1 - a Speex/PCM16 Codec Translator for Asterisk asterisk-mini - 1.0.10-1 - A minimal open source PBX asterisk-mysql - 1.0.10-1 - MySQL modules for Asterisk asterisk-pbx-dundi - 1.0.10-1 - Distributed Universal Number Discovery (DUNDi) support for Asterisk asterisk-pgsql - 1.0.10-1 - PostgreSQL modules for Asterisk asterisk-res-agi - 1.0.10-1 - Asterisk Gateway Interface module asterisk-sounds - 1.0.10-1 - a sounds collection for Asterisk asterisk-voicemail - 1.0.10-1 - VoiceMail related modules for Asterisk Daniel, Those are ancient! Capouch has MUCH newer packages. -- Mine probably aren't managed as well. I'm a one-person operation with too many irons in the fire!! Has anybody out there, on non-FPU embedded platorms, made any good use of things like ilbc and Speex? I downloaded those packages a while back but they were dramatically unusuable on either the WGT or the WRT models. Maybe I'm missing something? 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] SIP_HEADER function; what names are available?
Any news on this thread? I also need to know the way to get the R-URI
from sip INVITE messages received by Asterisk, through ${SIP_HEADER()}.
Thanks in advance,
Ricardo.
kjcsb wrote:
I have read the wiki about the SIP_HEADER function (http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header). Where can I
get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}. What are the
others?
I would guess that you can check the RFC. Easier is to turn on SIP
debug and see the INVITE packet yourself and
check the headers that you have with your equipment.
/Olle
Thanks but I don't know how to get the actual INVITE details (the
request URI?). For example I want to get sip:[EMAIL PROTECTED]
SIP/2.0 from the following dialogue:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
I can get Record-Route, Via, From, To etc but don't know how to get
the bit after the INVITE. Interestingly only the first Via is returned
by ${SIP_HEADER(VIA)}.
I've tried R-URI, RURI, URI, ALL, *, blank.
Any advice appreciated.
Cameron
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Re: [asterisk-users] RIP plainvoip - down ???
Apparently it is down for GOOD :-/ (RIP) http://www.voip-info.org/wiki/view/RIP+VOIP -- #Joseph On Fri, 2006-10-20 at 08:20 -0400, Mailing List wrote: - Original Message - From: J. Oquendo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 19, 2006 3:22 PM Subject: Re: [asterisk-users] plainvoip - down ??? Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails I can get there just fine. Your routes might be toasted [EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com PING plainvoip.com (66.199.240.2) 56(84) bytes of data. ... --- plainvoip.com ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9013ms rtt min/avg/max/mdev = 75.531/78.550/80.349/1.418 ms, pipe 2 Depending on your location thought, there are issues with GBLX possibly due to a fiber cut either in VA or DC. No, the service is down. If you turn asterisk off, isn't your box still pingable? ___ --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] PRI boards with g729 capable DSPs
I'm currently running 1.4b3 with a Digium card and 23 g.729 licenses. Is there a way I can get the g.729 codec work off the CPU and onto a DSP? Any T1/PRI cards with onboard codec DSPs? -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.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] How to get the agent id in the recording filename
Just a wild idea:Store the filename in a variable before the call enters the queue - say RECFILENAME - and then once you know which agent has taken the call, execute an mv operation (using the system command) something like
system(mv ${RECFILENAME} ${RECFILENAME}-${AGENTNAME})i don't remember the exact syntax but something like this should work. rajeevOn 10/19/06,
David Gagnon [EMAIL PROTECTED] wrote:
Hi,
I'm sure
some else has been facing this problem. I want to record all the call coming in
my queue. I want this format: MMDD-HHMMSS-AgentID-CallerId - UniqueID. I'm
using the monitor feature inside the queue.conf. I can't use the
agents.conf monitor features because I'm using dynamic agent
(addqueuemember)
The problem I'm facing is that I
can change the filename before the call enters the queue but at this step, I
don't know which agent will get the call.
Curent dialplan :
exten =
s,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID})
exten =
s,n,Playback(recording)
exten = s,n,Queue(myJavaClub,t,,,300)
Anyone could help?
David
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Re: [asterisk-users] voicemail usernames can't begin with j letter?
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone enters
for example john's voicemail, Asterisk thinks that j letter is jump
flag to n+1 priority. How can I disable, (if possible) this erroneous
interpretation that Asterisk does?
Regards,
Ricardo.
Bruce Reeves wrote:
What version of * are you running? I have several j usernames in
voicemail.conf under SVN-branch-1.2-r37458M.
On 10/20/06, *Ricardo Carvalho* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Dear all,
I've configured Asterisk Voicemail, but after some tests I
realised that
when some call is sent to the voicemail of someone which username
begins
with j letter, Asterisk gives me the error:
WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in
voicemail config file for 'ohn'
(for a called user named john, for example)
Is this some kind of Asterisk bug, or am I skipping some
configuration?
How can I make things work fine?
Thanks in advance,
Ricardo.
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Bruce
Nortex Networks
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RE: [asterisk-users] Polycom boot error
Subject: Re: [asterisk-users] Polycom boot error On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote: Hi Jesse, Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto Polycom IP500's? Or are they only for the later models? Do you know if you can still use TFTP for these software updates? They are compatible with the new bootroms and firmwares according to Polycom's release notes, but you cannot use the HTTPS provisioning on the IP500 I believe. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ Sweet thanks for that, is there any reason not to go to version 2.0.1 now? I know people were concerned initially because you cant go back but is there a reason to go back if I have a few Polycom IP 500's? Cheers, Dean ___ --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] Embedded Asterisk
On Fri, Oct 20, 2006 at 11:54:45AM -0400, Kristian Kielhofner wrote: I understand what a soft phone is. I know what java is. I also know that neither have anything to do with the slug. They do if Asterisk is runnin on the slug. What he meant was perfectly clear to *me*, Kristian; I'm not sure why you didn't understand him. The slug cannot support any web browsers that support java. The shipped interface may not utilize Java, but your assertion means that *no* current-day browser could operate a slug, which seems like a misstatement to me... The slug doesn't have any audio interfaces that could support audio for a softphone - so again, what does the slug have to do with a java softphone?!?!? You mentioned that compiling for the slug helped resolve some issues for you. What are you compiling, and how does it run on the slug? He's compiling his softphone, so that it does not have endianness problems which are exposed by connecting it to an Asterisk instance running on a slug, which is opposite-endian from most PC's, IIRC. Of course, if Asterisk *exposes* underlying endianness issues, it's broken... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] voicemail usernames can't begin with j letter?
Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone enters
for example john's voicemail, Asterisk thinks that j letter is jump
flag to n+1 priority. How can I disable, (if possible) this erroneous
interpretation that Asterisk does?
Have you tried exten = _[a-z].,3,VoiceMail(u${EXTEN})
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Re: [asterisk-users] Polycom boot error
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote: Sweet thanks for that, is there any reason not to go to version 2.0.1 now? I know people were concerned initially because you cant go back but is there a reason to go back if I have a few Polycom IP 500's? We've got clients running 501's on 2.0.1 with a good amount of success. Of those that we've upgraded to date, we didn't really have a reason to rollback with any of 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] PRI boards with g729 capable DSPs
On Friday 20 October 2006 13:01, Matthew Crocker wrote: I'm currently running 1.4b3 with a Digium card and 23 g.729 licenses. Is there a way I can get the g.729 codec work off the CPU and onto a DSP? Any T1/PRI cards with onboard codec DSPs? Digium's got their transcoder card. Are you running into CPU issues that this is becoming an issue? -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] #Transfer - Timeout is configurable?
Dear Marco, Take a look at featuredigittimeout, that might help :) Regards. --- M. Shokuie Nia From: Marco Mouta [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] #Transfer - Timeout is configurable? Date: Fri, 20 Oct 2006 15:54:40 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc7-f8.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 20 Oct 2006 08:33:38 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id B24CE2FCACB;Fri, 20 Oct 2006 07:54:46 -0700 (MST) Received: from psmtp.com (exprod8mx31.postini.com [64.18.3.131])by lists.digium.com (Postfix) with SMTP id 043422FC908for asterisk-users@lists.digium.com;Fri, 20 Oct 2006 07:54:31 -0700 (MST) Received: from source ([64.233.182.189]) by exprod8mx31.postini.com([64.18.7.10]) with SMTP; Fri, 20 Oct 2006 07:54:49 PDT Received: by nf-out-0910.google.com with SMTP id a25so1647923nfcfor asterisk-users@lists.digium.com;Fri, 20 Oct 2006 07:54:48 -0700 (PDT) Received: by 10.49.80.12 with SMTP id h12mr270922nfl;Fri, 20 Oct 2006 07:54:40 -0700 (PDT) Received: by 10.49.61.4 with HTTP; Fri, 20 Oct 2006 07:54:40 -0700 (PDT) X-Message-Info: txF49lGdW40NbEIHbS9sAQsIwf1mUc3x4qkWMdWhkFg= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:mime-version:content-type:content-transfer-encoding:content-disposition;b=EsdeGoRq2HU+414uKhc1nlvrB+1F3Yul63Gn1RtE0AVBAzJsVuy8H9SAgcOSScDviu4gAzCKkpbOHE3ie+84kH5l4oqdtnSkUoFsS+koelEdh6UgBH27DoYK56dzdZcLfvX6xXaLt5l6HuuYD9K0xB5vK3RpA/3/gCfP4nROYNA= X-pstn-levels: (S:30.80636/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 20 Oct 2006 15:33:38.0955 (UTC) FILETIME=[1D6E6DB0:01C6F45D] Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison Transfer the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call This is timeout after pressing the first digit isn't it? -- best regards, Marco Mouta ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] voicemail usernames can't begin with j letter?
I playing a bit with this, it seems that if you use the new syntax it
works:
exten = _[a-z].,3,VoiceMail(${EXTEN}|u)
You can, of course, also use the b, j, s, and g flags.
Even using the VoiceMail(u${EXTEN}) still elides the 'j'.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric ManxPower Wieling
Sent: Friday, October 20, 2006 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] voicemail usernames can't begin
with j letter?
Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with
version 1.2.4
and got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN}) exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone
enters for example john's voicemail, Asterisk thinks that
j letter
is jump flag to n+1 priority. How can I disable, (if possible) this
erroneous interpretation that Asterisk does?
Have you tried exten = _[a-z].,3,VoiceMail(u${EXTEN})
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Re: [asterisk-users] voicemail usernames can't begin with j letter?
On Fri, 2006-10-20 at 18:08 +0100, Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone enters
for example john's voicemail, Asterisk thinks that j letter is jump
flag to n+1 priority. How can I disable, (if possible) this erroneous
interpretation that Asterisk does?
Regards,
Ricardo.
If VoiceMail() has only one argument it falls back to old-style option
parsing (i.e., options at the beginning of the single argument, see
vm_exec() in apps/app_voicemail.c). If you use
exten = _[a-z].,3,VoiceMail(${EXTEN}|)
it should use the new-style (i.e., options as second argument).
Aside from that: Are you sure that it is a wise idea to use symbolic
mailbox names (instead of only numeric)? You will not be able to enter a
mailbox on the phone when asked for it (i.e., VoiceMailMain() without an
argument).
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europe tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/
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[asterisk-users] Escape from Voicemail
I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason ___ --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] Embedded Asterisk
Brian Capouch wrote: Kristian Kielhofner wrote: Administrator TOOTAI wrote: Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, OpenWRT -running on Linksys WRT- has asterisk packages. [EMAIL PROTECTED]:~# ipkg list | grep asterisk asterisk - 1.0.10-1 - An open source PBX asterisk-chan-mgcp - 1.0.10-1 - a Media Gateway Control Protocol implementation for Asterisk asterisk-chan-skinny - 1.0.10-1 - a Skinny Client Control Protocol implementation for Asterisk asterisk-codec-ilbc - 1.0.10-1 - an Internet Low Bitrate Codec (ILBC) Translator for Asterisk asterisk-codec-lpc10 - 1.0.10-1 - an LPC10 (Linear Predictor Code) 2.4kbps Voice Coder for Asterisk asterisk-codec-speex - 1.0.10-1 - a Speex/PCM16 Codec Translator for Asterisk asterisk-mini - 1.0.10-1 - A minimal open source PBX asterisk-mysql - 1.0.10-1 - MySQL modules for Asterisk asterisk-pbx-dundi - 1.0.10-1 - Distributed Universal Number Discovery (DUNDi) support for Asterisk asterisk-pgsql - 1.0.10-1 - PostgreSQL modules for Asterisk asterisk-res-agi - 1.0.10-1 - Asterisk Gateway Interface module asterisk-sounds - 1.0.10-1 - a sounds collection for Asterisk asterisk-voicemail - 1.0.10-1 - VoiceMail related modules for Asterisk Daniel, Those are ancient! Capouch has MUCH newer packages. -- Mine probably aren't managed as well. I'm a one-person operation with too many irons in the fire!! Has anybody out there, on non-FPU embedded platorms, made any good use of things like ilbc and Speex? I downloaded those packages a while back but they were dramatically unusuable on either the WGT or the WRT models. Maybe I'm missing something? B. B, If you went to GlobalSound, you might be able to get a fixed point ilbc implementation. I think they sell it. Of course I'm mostly joking. That would probably be a pretty large undertaking. It would still be awesome if someone did it, though. If someone does take up this cause - why not whip up a fixed point g729 implementation while you are at it? I'm pretty sure that there is an easy to get fixed point implementation of speex. I don't know if that will solve all of your issues, but it is a start. P.S. - We'll have to work on your code to get it in a buildroot. Maybe even some packages! Hopefully we'll have some time at Astricon. -- 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] Embedded Asterisk
Jay R. Ashworth wrote: On Fri, Oct 20, 2006 at 11:54:45AM -0400, Kristian Kielhofner wrote: I understand what a soft phone is. I know what java is. I also know that neither have anything to do with the slug. They do if Asterisk is runnin on the slug. What he meant was perfectly clear to *me*, Kristian; I'm not sure why you didn't understand him. The slug cannot support any web browsers that support java. The shipped interface may not utilize Java, but your assertion means that *no* current-day browser could operate a slug, which seems like a misstatement to me... The slug doesn't have any audio interfaces that could support audio for a softphone - so again, what does the slug have to do with a java softphone?!?!? You mentioned that compiling for the slug helped resolve some issues for you. What are you compiling, and how does it run on the slug? He's compiling his softphone, so that it does not have endianness problems which are exposed by connecting it to an Asterisk instance running on a slug, which is opposite-endian from most PC's, IIRC. Of course, if Asterisk *exposes* underlying endianness issues, it's broken... Cheers, -- jra jra, He implied that he made code tweaks to his web based softphone (which is written in java) as a result of running it on the slug. I was trying to figure out exactly what he was talking about, and how he did it. Slug audio device? Which java virtual machine? How did the interface work? These were the kinds of questions that I had. He answered all of them off-list. So it seems I wasn't as far off as you seem to think. It actually had *nothing* to do with Asterisk running on the slug, so it seems that you might be even more confused than I am :). He confirmed off-list that the scenario he described did not involve running Asterisk on the slug. It involved running the softphone in a java vm on the slug, as a softphone with audio, etc. Check out the website. The shipped product appears to be %100 Java. That's pretty clear. As far as my misstatement, I think it was pretty clear too: the slug cannot support any browsers that support java *that support java* - this is key Which is completely true. links, lynx, etc. These are all browsers that run perfectly well on the slug. None of them support java (even if they did, what would be the point)? Does the slug support browsers - yes. Do those browsers support java - no. Although you claim to fully understand him, your post indicates just the opposite. Did you read the whole thread? P.S. - FYI, Asterisk does run on the slug. Quite well in fact. Just like it runs on mipsel, xscale, ppc, x86_64, and x86. -- 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
[asterisk-users] some transfers dropped.
We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is deferent in the logs above the CANCEL request for successful or failed transfers. So, I am not sure why the CANCEL is being sent. I can not discern what may be different when it fails. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ref: from SIP Phone (I think these are the second invite after it is hung up) 2006-OCT-20 17:49:52 GMT +++ Current Timestamp +++ 2006-OCT-20 17:19:47 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:56:37 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:50:00 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:45:38 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:11:28 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:10:58 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 14:59:26 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 12:45:30 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 19:53:25 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:40:52 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:03:45 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 17:55:55 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:09:13 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:04:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:52:12 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:34:35 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:20:17 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 13:45:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER ref. from asterisk 1.2.10 logs: Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Requested transfer capability: 0x00 - SPEECH Oct 20 13:19:45 DEBUG[8159] channel.c: Avoiding initial deadlock for 'Zap/25-1' Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Called g2/5155 Oct 20 13:19:45 VERBOSE[10652] logger.c: Transmitting (no NAT) to 172.16.8.200:5065: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200 From: From Desksip:[EMAIL PROTECTED];tag=2425948795 To: sip:[EMAIL PROTECTED];tag=as279eb184 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Oct 20 13:19:45 DEBUG[10658] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 2 (In use) Oct 20 13:19:45 DEBUG[10659] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8167] chan_zap.c: Enabled echo cancellation on channel 25 Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Zap/25-1 is ringing Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 6 (Ringing) Oct 20 13:19:45 DEBUG[10660] app_queue.c: Device 'Zap/25' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8171] chan_sip.c: Header 0: (0) Oct 20 13:19:46 VERBOSE[8171] logger.c: -- SIP read from 172.16.8.200:5065: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 To: sip:[EMAIL PROTECTED] From: From Desksip:[EMAIL PROTECTED];tag=2425948795 Call-Id: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 2 CANCEL Content-Length: 0 Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 (36) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 (65) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To: sip:[EMAIL PROTECTED] (27) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 3: From: From Desksip:[EMAIL PROTECTED];tag=2425948795 (55) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 4: Call-Id: [EMAIL PROTECTED] (43) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 5: Max-Forwards: 70 (16) Oct 20 13:19:46
Re: [Asterisk-Users] rxfax problem
Hi Steve, As a matter of fact, you've done a greate job in writting this library, no doubts. I really dont know rxgain = 12 makes that much distortion but I'm curios to know if I pass through the incoming fax to an analog fax machine on another fxs line, the machine wouldn't receive the fax too? Anyways, let me take the most benefit as im sure you'd read this post, i have problem with the size of received page which is shrinked, can u give me a hint about this problem too :) Thanks. --- M. Shokuie Nia From: Steve Underwood [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] rxfax problem Date: Fri, 20 Oct 2006 20:20:18 +0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc6-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 20 Oct 2006 05:42:01 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id EFF0B2FC87C;Fri, 20 Oct 2006 05:20:37 -0700 (MST) Received: from psmtp.com (exprod8mx13.postini.com [64.18.3.113])by lists.digium.com (Postfix) with SMTP id B67A62FC82Ffor asterisk-users@lists.digium.com;Fri, 20 Oct 2006 05:20:05 -0700 (MST) Received: from source ([202.14.67.92]) by exprod8mx13.postini.com([64.18.7.10]) with SMTP; Fri, 20 Oct 2006 05:20:20 PDT Received: from [192.168.2.50] (229.166.17.210.dyn.pacific.net.hk[210.17.166.229]) by cwb.pacific.net.hk with ESMTPid k9KCKIfs013165 for asterisk-users@lists.digium.com;Fri, 20 Oct 2006 20:20:19 +0800 X-Message-Info: txF49lGdW43chsCTszkrRosGSMI+inUm7kbzJdpspc0= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com User-Agent: Mozilla Thunderbird 1.0.8-1.1.fc4 (X11/20060501) X-Accept-Language: en-us, en References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 20 Oct 2006 12:42:02.0256 (UTC) FILETIME=[241ED900:01C6F445] M. Shokuie Nia wrote: Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldnt hand shake with other side well. rxfax isn't sensitive to noise at all. At a gain of 12 you've caused overloading and distortion, and the signal cannot be decoded. Many people seem to be nearly deaf. They run systems at massive gain with awful distortion, and seem content until they find something like a modem or DTMF detection doesn't work too well. Steve ___ --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 _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --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] getting DID info..
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers
control to the context defined in the IAX.conf -where I have the
trunk line defined...
exten = h,1,Hangup
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,SayAlpha(${EXTEN})
It is my impression that the EXTEN variable is used as the internal
extension - not the incoming DID number, but you seem pretty
confident so I must be wrong. What Im looking to do is a FOP pop-up
with the DID number and caller ID number in it... I'll tie that
into a web-based database...
Here's my full log file..
Oct 20 14:23:42 VERBOSE[5387] logger.c: -- Accepting
AUTHENTICATED call from 204.11.194.34:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, LOOPCOUNT=0) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, __DIR-CONTEXT=default) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Answer
(IAX2/204.11.194.34:4569-4, ) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Wait
(IAX2/204.11.194.34:4569-4, 1) in new stack
Oct 20 14:23:43 DEBUG[5387] chan_iax2.c: Ooh, voice format changed to 4
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Executing SayAlpha
(IAX2/204.11.194.34:4569-4, s) in new stack
Oct 20 14:23:43 DEBUG[5862] channel.c: Scheduling timer at 160 sample
intervals
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Playing 'letters/
s' (language 'en')
DID is the inbound call number.
The is notation for CallerID name, that won't help.
s is the start extension. setting it to FROM_DID makes no sense.
(This is the extention that starts in this context; it is a
default, if
the context is started without an extension. (eg batphone or called
from another
context))
FROM_DID=${EXTEN} gets you the right number.
However, SayNumber is looking for a SINGLE digit. Your
000-000- style number is overflow, and hence zero.
You have to parse the number to do this right.
If you aren't sure how, let me know, I might have a macro to do it.
Thanks,
J.
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Re: [asterisk-users] Embedded Asterisk
On 20 Oct 2006, at 19:05, Kristian Kielhofner wrote: It actually had *nothing* to do with Asterisk running on the slug, so it seems that you might be even more confused than I am :). He confirmed off-list that the scenario he described did not involve running Asterisk on the slug. It involved running the softphone in a java vm on the slug, as a softphone with audio, etc. Well, just to fill in a datapoint, there was an issue, but not in asterisk. I was doing something wrong in the javacode with the way I built with the ipaddress Info Element in IAX. It turned out that the byte swapped slug didn't like my reply (which was wrong) in a way that failed, rather than the i386 arch systems which just silently ignored the my error. My point (if I had one) was that testing against 'odd' architectures is _very_ informative. Tim Panton www.mexuar.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] centos or rhel and txfax with libtiff
I am attempting to use txfax on centos 4.4 the libtiff is: libtiff-devel-3.6.1-12 libtiff-3.6.1-12 Is this OK or do I have to download the libtiff stuff and install it also. I am not having much luck faxing yet. I receive 1/3 pages or 2/3 pages etc... I have yet to receive 3/3 pages. THanks, Jerry ___ --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] Embedded Asterisk
Tim Panton wrote: On 20 Oct 2006, at 19:05, Kristian Kielhofner wrote: It actually had *nothing* to do with Asterisk running on the slug, so it seems that you might be even more confused than I am :). He confirmed off-list that the scenario he described did not involve running Asterisk on the slug. It involved running the softphone in a java vm on the slug, as a softphone with audio, etc. Well, just to fill in a datapoint, there was an issue, but not in asterisk. I was doing something wrong in the javacode with the way I built with the ipaddress Info Element in IAX. It turned out that the byte swapped slug didn't like my reply (which was wrong) in a way that failed, rather than the i386 arch systems which just silently ignored the my error. My point (if I had one) was that testing against 'odd' architectures is _very_ informative. Tim Panton www.mexuar.com Tim, Thanks for the full story! -- 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] Asterisk hangs up on incoming analog calls after a while
On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] wrote:Date: Thu, 19 Oct 2006 09:30:38 -0500 From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? No. They are not set. i.e. default___ --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] Snom 320, Queues and Transfer not working as expected with * 1.2.12.1
Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded to 1.12.1 and the ability to transfer out of the queue went away. Something must have changed. I tried to implement the call transfer mechanism in features.conf. I selected #1 for blind transfer and #2 for attended transfer. The * on the Snoms has an internal use. This seems to work well with other phones than the Snoms. On a Snome one have to press the check button to send the #2. This seems to work occasionally, but not reliably. The transfer call doesn't seem to work at all. I have to confess, that I only have user reports on this. Does anyone have a tip how to get this to work? The users would love to have the old functionality back - to use the specific Snom keys directly. But if we get #2 or something similar to work well, its good enough. Btw, transfer using the normal Snom buttons work well for regular calls, just not in Queues. Problem B - Quick Dial Buttons: I have used the programmable function keys together with the hint system in * to monitor local lines. It works very well, impressive! But people like to use these buttons as quick dial buttons for external numbers too. They program the buttons the same as the internal lines with hints using the Destination feature for the button. This seems to be wrong, as * gets upset over not having a hint for these requests. Trying to read the Snom manual I don't get a clear answer how to program this. Or I don't read well enough... Does anyone have a tip how to solve this?? I could set up a hint for all external numbers people like to program, but it does not scale well and would not be very meaningful. Problem C - A not Snom related transfer problem: When I use #2 to transfer a call with *s internal feature system, I need a way to go back and force between the callee and the goal of the transfer. I can't find a way to do this, either documented or elsewhere. Anyone a tip??? Thanks for your effort! Regards:Håkan pgp5qixJqNoG6.pgp Description: 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] modprobe Ztdummy is not working
Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point modprobe ztdummy it's failing. I'm getting these messages: [EMAIL PROTECTED]:~# modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.17.6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy After doing dmesg I'm getting this: [...] Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.7 Echo Canceller: KB1 ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control I have Kernel 2.6.17.6, and zaptel 1.2.10. What do you thing about this? Thanks J-E ___ --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] getting DID info..
This might be a newbie question...
You're right, part ofit is. I don't mean to sound rude, but you really need to go do some research first to get the basics down. First place is to read the book, Asterisk: The Future of Telephony (available for free, there's this site called
google.com that can give you loads of information, including where to download the book), the next stop would be www.voip-info.org.
By basics, I'm referring to your numbering. It should be:
exten = s,1,
exten = s,n,
Now, to answer your question, which looks to be a good question. On a PRI, the variable ${DNID} contains the dialed number (I believe, or at least it did for me in testing). On a SIP connection, see:
http://www.voip-info.org/wiki/view/DNID. It appears you may need to use something else for a SIP connection. You'll have to see for yourself. In my case, it does not return the number dialed, but rather the SIP username dialed.
BTW, the FROM_DID variable is something from FreePBX, not standard Asterisk. Are you using standard Asterisk, FreePBX, Trixbox, or [EMAIL PROTECTED]?
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[asterisk-users] Re: Snom 320, Queues and Transfer not working as expected with * 1.2.12.1
HK == Håkan Källberg [EMAIL PROTECTED] writes: HK Problem B - Quick Dial Buttons: HK I have used the programmable function keys together with the hint HK system in * to monitor local lines. It works very well, HK impressive! But people like to use these buttons as quick dial HK buttons for external numbers too. They program the buttons the HK same as the internal lines with hints using the Destination HK feature for the button. This seems to be wrong, as * gets upset HK over not having a hint for these requests. Trying to read the Snom HK manual I don't get a clear answer how to program this. Or I don't HK read well enough... Upgrade to 5.x or 6.x, then you can make quickdial keys which won't query asterisk for status. I don't recall exactly which setting you need to pick in the drop down menu, but it should be reasonably easy to find. Otherwise ask again and I'll find it on Monday. /Benny ___ --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] Escape from Voicemail
Here you go, from the voip-info.org wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail Also. during the prompt if the caller presses: '*' - the call jumps to extension 'a' in the current voicemail context. This needs an example '#' - the greeting and/or instructions are stopped and recording starts immediately. On 10/20/06, Jason Walker [EMAIL PROTECTED] wrote: I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] modprobe Ztdummy is not working
Top posting since this is simple. Your kernel does not have a RTC compiled in or as a module... Do you build your own kernels? If so add RTC as a builtin or a module. If you use a distro kernel, you might be able to modprobe the RTC module. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Etienne Kelly Sent: Friday, October 20, 2006 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] modprobe Ztdummy is not working Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point modprobe ztdummy it's failing. I'm getting these messages: [EMAIL PROTECTED]:~# modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.17.6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy After doing dmesg I'm getting this: [...] Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.7 Echo Canceller: KB1 ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control I have Kernel 2.6.17.6, and zaptel 1.2.10. What do you thing about this? Thanks J-E ___ --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] Embedded Asterisk
Has anybody out there, on non-FPU embedded platorms, made any good use of things like ilbc and Speex? The exisiting implementations of both run very poorly on a non-fpu cpu's, especialy if clock speed 400 Mhz I have run asterisk (and still do) on mips,ixp and powerpc (all without fpu's) and i think that without modifications the codecs are not so usable There are 3 options 1) Get a faster fp library - Been looking into the GoFast fp lib, no definate results yet 2) Convert codecs to fixed point - Although i know a G729 fixed point implementation exists haven't tested and i am not sure that a speex or ilbc implementation exists. 4) Get a cpu with fpu :) - There are mips and powerpc cpu's (i am talking the types used in embedded dev's) that have an fpu I will be also at Astricon and brinking with me a powerpc based embedded asterisk appliance which has support for zaptel also. Maybe we could exchange some ideas on the matter. Stelios ___ --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] getting DID info..
Todd- Asterisk wrote:
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers control
to the context defined in the IAX.conf -where I have the trunk line
defined...
exten = h,1,Hangup
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,SayAlpha(${EXTEN})
It is my impression that the EXTEN variable is used as the internal
extension - not the incoming DID number, but you seem pretty confident
so I must be wrong. What Im looking to do is a FOP pop-up with the DID
number and caller ID number in it... I'll tie that into a web-based
database...
There is no difference between an extension and a DID as far as Asterisk
is concerned. You must have typoed the above example as you do not have
an exten = s,1
When you do a exten = s,n,SayAlpha(${EXTEN}) the extension IS s. If
it was not s then it would never have gotten to that extension.
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Re: [asterisk-users] modprobe Ztdummy is not working
On Fri, Oct 20, 2006 at 04:51:47PM -0400, Jean-Etienne Kelly wrote: Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point modprobe ztdummy it's failing. I'm getting these messages: [EMAIL PROTECTED]:~# modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.17.6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy After doing dmesg I'm getting this: [...] Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.7 Echo Canceller: KB1 ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control I have Kernel 2.6.17.6, and zaptel 1.2.10. What do you thing about this? Thanks What Linux distribution? Self-complied kernel? Does it have RTC enabled? To answer the latter, please provide the output of: grep RTC path/to/kernel_config -- 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
