Re: [Asterisk-Users] Can't compile cdr_mysql
On Wednesday 13 August 2003 03:24 pm, Jerk Face wrote: > /usr/src/usr/include/mysql/errmsg.h > The version of MySQL that I'm running is 3.23.57-1 What distribution are you running? That's a pretty braindead place to put the mysql header files. You'd think someplace like /usr/include, /usr/local/include, or even /usr/local/mysql/include, but /usr/src/usr/include shouldn't be the correct location on any Linux system. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail2 - auto fill the dialing extension?
Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming extension is, and use it? On our current handsets, the voice mail button goes directly to the password prompt. And if you dial in and get voice mail, # sends you to the "please enter voice mail box please..." (Siemens HiCom). Not a must-have feature, but something I'd like to use myself and for our office. Also, we decided to go with actual extension numbers on the phones instead of usernames per extension. On the Cisco phones, is there a way to change the name/number on the top line (white text on black) to the user's name, while having the extension number next to each presentation (line1, line2, etc)? Thanks, * is a rocking and rolling! Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
On Wed, 2003-08-13 at 18:36, Andy Powell wrote: > On 13/08/2003 at 17:46 Dave Cotton wrote: > > >in the file wcfxo.c the following structure is initialised as below > >which would suggest that FCC is wrong for France or pretty well all of > >Europe. > > errm, > > FCC mode is for the US. CTR21 is for Europe - you even pasted the info > in your message! Exactly, the question really is how do you change it? I've even commented out the FCC bit and it still gives the same in /var/log/messages -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference + E100P + H323
Hello Martin, Yes, I have span configure in zaptel.conf: span=1,0,0,esf,b8zs I dont have a PRI plugged in to the card. Would it be an issue? Reason is I am current only testing the call originating from H323 endpoints. Firewall shouldn't be a issue since the call works fine with ztdummy loaded. I debug the chan_h323 and it uses the right codec G729 from digium. Only cant hear the Meetme prompt. Foong - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 13, 2003 12:28 PM Subject: Re: [Asterisk-Users] Conference + E100P + H323 > On Wed, 13 Aug 2003, Chee Foong wrote: > > > Hello, > > > > I have a E100P card from digium and I try to implement a conference > > bridge in asterisk. > > > > I wonder since I got the E100P card do I still need to load ztdummy > > for caller from h323 endpoints to work with Meetme? > It's not necessary. > > > > > I load the E100P driver but i did not load the ztdummy driver. My h323 > Do you have the span configured in /etc/zaptel.conf ? > > > caller does not hear any voice play by Meetme. Looks like ztdummy is > > required as long as h323 is concern and not depend on whether there is > > a zaptel device. > Check the firewall and codecs. > > regards > Martin > > > > > Foong > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Andy Powell wrote: FCC mode is for the US. CTR21 is for Europe - you even pasted the info in your message! Exactly, the question really is how do you change it? modprobe wcfxo opermode=1 HTH Andy This switch (opermode=1) is redundant with the current X100P cards, as it changes register contents that are specific to the "Global" version of the chipset on the card. The X100P currently out there uses the "USA & Japan" chipset, and thus does not achieve the intended result. The register concerned deals with the impedance presented to the line connected to the card - 600 ohms (US) vs various complex impedances used in other countries. For internationalised FXO cards, see Mark's recent comments, in the thread "Does Wildcard x100p support BT Caller ID in UK?" Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323, Asterisk and DTMF issue
We use rfc2833 for everything and have no trouble. Make sure your 7960 is sending the right indications. Jeremy McNamara Jay Sakata wrote: I have the same problem that Michael describes below does anyone have any recommendations? Jay __ Hi folks, I’m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won’t work with Asterisk’s voicemail system. I’m using the g.729 codec for h323 and Asterisk. I’m told dtmfmode=inband won’t work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summary: dtmfmode = info ; works with h323 not with Asterisk Voicemail dtmfmode = inband; works with h323 (with a flood of warnings) not with Asterisk Voicemail dtmfmode = rfc2833 ; works with Asterisk Voicemail not with h323 Any help would be greatly appreciated. Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open G.729A codec
> I made a mistake of buying it so that I can have a low-bandwidth > well-tested codec for use on an IAX2 link. Then I've caused Digium lots > of unwanted trouble, because hair stood on the back of my neck after > reading the licensing agreement and seeing the .so library. Let's hope > it gets better in the future! Believe it or not, we worked hard to get that license agreement *improved*. I wish they took our concerns (and those of our customers) more seriously. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload
Chee Foong wrote: I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place. Type "help" into the console and read. canopy*CLI> help restart gracefully Usage: restart gracefully Causes Asterisk to stop accepting new calls and exec() itself performing a cold. restart when all active calls have ended. canopy*CLI> help restart when convenient Usage: restart when convenient Causes Asterisk to perform a cold restart when all active calls have ended. canopy*CLI> help reload Usage: reload Reloads configuration files for all modules which support reloading. -- Alastair Maw <[EMAIL PROTECTED]> MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
It can be a bad module. Contact [EMAIL PROTECTED] regards Martin On Wed, 6 Aug 2003, Eduardo Goncalves wrote: > Martin, > > With KFLAGS+=-DNO_CALIBRATION uncommented, the module loads fine. Without the > calibration. > But I have no dial-tone on port 4. All the three other ports works fine. > Could it be a hardware problem? > > Thanks in Advance > Eduardo > > On Fri, 1 Aug 2003 16:11:12 -0500 (CDT) > Martin Pycko <[EMAIL PROTECTED]> wrote: > > > Try to uncomment in zaptel/Makefile > > KFLAGS+=-DNO_CALIBRATION > > > > and "make clean install" > > > > that should help > > > > Martin > > > > On Fri, 1 Aug 2003, Eduardo Goncalves wrote: > > > > > On Fri, 1 Aug 2003 15:34:23 -0500 (CDT) > > > Martin Pycko <[EMAIL PROTECTED]> wrote: > > > > > > > What does 'dmesg' say ? > > > > > > > > > > > > > CSLIP: code copyright 1989 Regents of the University of California > > > PPP generic driver version 2.4.1 > > > Zapata Telephony Interface Registered on major 196 > > > Freshmaker version: 62 > > > Freshmaker passed register test > > > Module 0: Initialized > > > Module 1: Initialized > > > Module 2: Initialized > > > Timeout waiting for calibration of module 3 > > > ProSlic died on Calibration. > > > Module 3: Not installed > > > Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) > > > > > > and then the errors that I mentioned. > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk behind NAT?
This is starting to sound like a feature request, perhaps by using the same method that Cisco phones use (comparison using the Via: header, and re-registering if the Via: header is different than the known IP address.) JT At 11:02 PM -0700 8/12/03, Terence Chan wrote: Wasim: Hi! Thanks a lot for your help. I guess the problem is the REGISTER command. The field is using the local IP address instead of the external IP. I am just wondering if it is possible to tell asterisk what is the external IP so that it can be used for the registration. Thanks again. Terence -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, August 11, 2003 9:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Running Asterisk behind NAT? Yes, you can run Asterisk behind a NAT. NO, you CAN'T (reliably, easily) run SIP behind a NAT. For FWD think about using their behindnat and fwdproxy addresses. Maybe a STUN would help. Also, test your setup infront of NAT also, make sure they work, before you head behind a NAT. -- wasim This mail is confidential & intended solely for the use of the addressee. On Tue, 12 Aug 2003, Terence Chan wrote: I would like to ask if it is possible to run Asterisk behind NAT. I have a linksys router that forwarded the port UDP 5082 to the local IP of my Asterisk box, I got the error 479 when I try to register my Asterisk box with FWD. (see detail below). Have anyone got Asterisk working behind NAT and successfully registers with FWD? Any pointer or information will be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip and One Way Audio
It does, but you have to use IAX2 (or IAX) which is a single-socket, sanely designed protocol which penetrates any NAT/PAT which does not explicitly block outbound UDP connections on port 4569 (or 5036 for old IAX1) Mark On Tue, 12 Aug 2003, Dave Cotton wrote: > On Tue, 2003-08-12 at 15:29, Steve Lane wrote: > > Would the firewall pose a problem? I thought Asterisk had the solution > > for working behind a firewall? > > If is has it's one of the best kept secrets. > -- > Dave Cotton <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension and phone management bestpractices??
On Wed, 13 Aug 2003, Devon Henderson wrote:
[...]
> > > We have agents who work both from home and from the office.
> > Some agents are
> > > always in the office, some are always at home, and some
> > alternate between
> > > the two.
[...]
> I guess my big question is: is it possible to have extensions mapped to
> people, not to phones?
Sure, as mentioned by others, you'll just have to map user extensions to
phone lines when users are logged in.
I'd strongly recommend assigning numbers to to individual phones, too.
(Sometimes you want to call "room No. 123, no matter who's logged in
there", eg if the PC in there suddenly stops responding and you want
somebody to look after that...) Those location based extensions would be
independent from the technology used, so you could exchange MGCP, SIP or
analog phones by just reassigning the phone number to a new channel.
That said, you can quite nicely map user extensions to location extensions
using *'s extension logic and the database application. Let's assume your
inhouse policy says:
Extensions 5XYZ refer to Phone No. Z in room No. Y on Floor X,
Extensions 6XXX refer to user no. XXX.
You would then start by conventionally mapping your phone lines to the
5XYZ extensions in extensions.conf (and the appropriate channel
configuration files).
Next step is adding your users to *'s database, assigning them a password,
eg:
CLI> database add password 001 123
CLI> database add password 002 456
(That assigns user 001 the password "123", whereas user 002 gets "456".)
Now you can add quite simple extension logic for all users in your
database, eg. like this (extensions.conf, again):
exten = _6XXX,1,DBGet(TARGET=location/${EXTEN:1})
;user location was found in DB, go there (assuming _5XXX in "default")
exten = _6XXX,2,Goto(default,${TARGET},1)
;DBGet failed, so user is not logged in. => give VoiceMail
exten = _6XXX,102,Voicemail2(u${EXTEN})
You'll probably want to set outgoing caller ID to the user's extension
instead of the location-based one. That's easy, too:
Put all the phones into a special context that has this:
exten = s,1,DBGet(USER=UID/${EXTEN:1})
exten = s,2,SetCallerID(6${USER})
exten = s,3,Goto(default,s,1)
Finally, you'll need some way for users to register. That could be a web
form that checks the user-provided password (using DBget, maybe via
asterisk -rx "database get password XXX"), and puts the given location
into the database (using DBput, maybe via
asterisk -rx "database put location XXX 5XYZ")
The same script must set the reverse mapping, too, using DBput or like:
asterisk -rx "database put user 5XYZ XXX"
You could also do that via extension logic, eg let people just dial
7XXX to log in, using something like this:
(note that I leave the password check as an exercise to the reader ;-)
; login user if they come from a location (5XXX) extension
exten = _7XXX/_5XXX,1,DBput(location/${EXTEN:1}=${CALLERIDNUM})
exten = _7XXX/_5XXX,2,DBput(user/${CALLERIDNUM}=${EXTEN:1})
exten = _7XXX/_5XXX,3,Playback(login-succcessfull)
exten = _7XXX/_5XXX,4,VoiceMailMain(s6${EXTEN:1})
And to logout, just have people dial 7000:
exten = 7000/_6XXX,1,DBget(LOCATION=location/${CALLERIDNUM:1})
exten = 7000/_6XXX,2,DBdel(user/${LOCATION})
exten = 7000/_6XXX,1,DBdel(location/${CALLERIDNUM:1})
exten = 7000/_6XXX,2,Playback(user)
exten = 7000/_6XXX,3,SayDigits(${CALLERIDNUM:1})
exten = 7000/_6XXX,4,Playback(logged-out)
Phew!
That has become more that I had expected. Anyhow: You should get the idea
that it's possible (and quite useful) to have _both_ locations and users
assigned to constant extensions using Asterisk, and you can do the login
via both a web form (CGI, PHP, whatever) or directly by having people dial
"magic" extensions.
Cheers,
Siggi
___
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Re: [Asterisk-Users] Virtual extension as local modem
Only if you have another FXS port and a real modem connected to that and you bridge the call between FXS and FXO in asterisk. regards Martin On Thu, 14 Aug 2003, Dan wrote: > Hi, > > There is any possibility to define a virtual extension on the asterisk box > to act as a local modem? > This is the scenario I think of: > - call Asterisk from PSTN throug the X100P card. > - answer the call and prompt the user with a menu. > - dial a special extension (long number) where you get a data modem type > answer and connect. > I want to enable a RAS server on te same box, using a single PSTN line. > I don't want to use something like an ATA for this purpose ( I don't even > think that you gen get a decent data connection speed). > The ideal thing will be to emulate a real modem through the serial port > (make the linux box to act as a serial modem for another computer). Then you > can define another computer as a fax/data(RAS) box. > > Do you thik this is something feasable? > Any comments are wellcome. > > Thanks, > Dan > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension and phone management bestpractices??
> > I guess my big question is: is it possible to have extensions mapped to > > people, not to phones? > > Yes, you just need to link the user/extension to a technology/channel > when logged in, and to a bogus value when not so that your dial will > fail quickly and fall through to voicemail. Also you will want to make > sure your voicemail goes out in email since you won't be able to get > stutter tone or MWI due to the channel not being assigned a extension. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users It appears to me that this functionality is going to be required more and more often by users of Asterisk, especially in larger organisations where employees roam between home and an office or between offices.. I think I will add a feature request for "Virtual Extensions" to bugs.digium.com becasue I think it would be a valuable addition to Asterisk.. The application would probably be fairly simple for a developper to create but for my limited progamming skills I would take ages to get even close to a working solution.. Even something simple to keep the user/extension mapping static would be a major step in the right direction and then it could be added to to work with the queues and various other components of Askterisk.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension and phone management best practices??
neither does agentlogin, see http://www.digium.com/asterisk_handbook/agentlogin_queues.html remember, you can define members as devices/types (like agents)/local so you can create some pretty wicked setups Brian West wrote: > > Nope.. sure doesn't.. You call the AgentLoginCallback extension from any > phone.. Enter you agent ID.. and Password... then enter the extension your > calls should go to and its done. > > bkw > > On Wed, 13 Aug 2003, Devon Henderson wrote: > > > Being a relative Asterisk newbie, I may be wrong.. but as far as I can tell, > > it doesn't. The standard queue/agent logic requires that you assign an > > extension to a phone. > > > > Someone correct me if I'm wrong, please. :) > > > > - Devon > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Brian West > > Sent: Wednesday, August 13, 2003 10:10 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] Extension and phone management best > > practices?? > > > > > > AgentLoginCallback does this doesn't it? > > > > On Wed, 13 Aug 2003, Steven Critchfield wrote: > > > > > On Wed, 2003-08-13 at 11:28, Devon Henderson wrote: > > > > We're still in the planning stages of our Asterisk implementation, but > > we > > > > have a requirement that the extension be mapped to a user, with the > > phone as > > > > a variable (we have hot seats in our contact center, and we also have > > agents > > > > that work both from remote locations and our contact center). > > > > > > > > So, I am also very interested to see what everyone has to say about > > this. > > > > > > All of this can be done in a very simple dbapp or agi app. BAsically all > > > you need is a way to identify the user and the channel they are on. Then > > > you just consult your data store for the mapping. > > > > > > So your basic app should need a login script, a logout script, and a > > > translation extension logic. > > > > > > login script should look to see if another user had logged in at that > > > phone and disable that login. logout script can make the mapping invalid > > > so dial will jump to voicmail. Your extension logic would just need to > > > be a pattern match on your extensions, and a lookup of the extension in > > > the data store, retrieving the current mapping and attempt to dial the > > > phone listed. Might even want to include the technology in your mapping > > > as it will let you go SIP, H323, IAX, and Zap channels. > > > -- > > > Steven Critchfield <[EMAIL PROTECTED]> > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mixing audio from Music on Hold and IVR
Oh my why do that? Customers/Users will have a hard time hearing and understanding in some cases. bkw On Wed, 13 Aug 2003, Stuart Hirst wrote: > Does anyone know if it would be possible to play music on hold in the > background whilst playing IVR prompts. I am hoping that this would have > the effect of the background music being continuous when moving between > IVR levels. There maybe a break when moving between IVR levels but the > background music would not start again from the beginning at the start > of each IVR prompt. > > Rgds, > > Stuart > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP phone recommendation
Hey thanks. Much appreciated! > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Andy Hester > Sent: Wednesday, August 13, 2003 9:23 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] IP phone recommendation > > > Nathan, > I am using the Pingtel phones at a customer site. I > should be able to give > a report in a couple of days > > Sincerely, > Andy Hester > Consero > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Nathan > > Littlepage > > Sent: Wednesday, August 13, 2003 8:15 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] IP phone recommendation > > > > > > Has anyone had the opportunity to use a PingTel phone with Asterisk? > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile cdr_mysql
On Wed, 2003-08-13 at 15:06, Jerk Face wrote: > I'm trying to compile the cdr_mysql module, but I am receiving error > messages. > I have installed mysql-devel. > > Here is the output of make cdr_mysql: > > cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o > cdr_mysql.o cdr_mysql.c > cdr_mysql.c:30:26: mysql/errmsg.h: No such file or directory > cdr_mysql.c: In function `mysql_log': > cdr_mysql.c:74: `CR_SERVER_GONE_ERROR' undeclared (first use in this > function) > cdr_mysql.c:74: (Each undeclared identifier is reported only once > cdr_mysql.c:74: for each function it appears in.) > make: *** [cdr_mysql.o] Error 1 > > I have looked up CR_SERVER_GONE_ERROR, but I'm not sure how that applies to > compiling the module. If you look just 2 lines farther up the screen you will see your true problem. If it can't find errmsg.h then it should fail. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Handled
On Wed, 13 Aug 2003 17:56:46 -0500 Tilghman Lesher <[EMAIL PROTECTED]> wrote: > > The CNG tones are sent by the sending fax machine, not the receiving > fax machine. Those tones are sent from the moment that the fax > machines dials and continues until either a timeout or the receiving > fax machine sends its synchronization tone. Hum, Thanks for the explanation. > > How is your fax machine connected to the Asterisk machine? |FAX|---|PBX|---|ATA186|SIP---|Asterisk|E1-e&m|PSTN| -- Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't know how to calculate timelen
Maybe get on IRC and try to debug it with IAX2. SHouldn't be any different peering as long as your gateway provider supports it. Mark On Thu, 14 Aug 2003, Dave Wilson wrote: > mark wrote: > > > > Can you try iax2? > > > > We tried that, but couldnt seem to get the peering to work on IAX2. We being > myself and a third party gateway provider in UK > > Dave > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Asterisk user.
Hello, I am a new asterisk user as well. I am very impressed with the functionality and flexibility of the platform. However I can not make out the documentation as it seems kind of vague. I was hoping if some one could kind of guide me through off-line and help me get familiar. It has been a while since I have been on the Linux scene, but I do have plenty of UNIX experience. I have a developer background but haven’t used any of my skills for the last 6 months (long story). Anyway I was hoping if someone with a bit of knowledge and expertise could guide me into becoming an expert as well. I have quite a bit of telephony experience and I would like to share that as well. One other thing I was wondering that would probably make life easier and it’s probably been asked before but… I wanted to know if there is a GUI that is used to manage Asterisk? Not that I have a problem with writing config files or anything but I was just wondering. Thanks in advance. Steve Lane [EMAIL PROTECTED]
Re: [Asterisk-Users] Can't compile cdr_mysql
On Wednesday 13 August 2003 03:06 pm, Jerk Face wrote: > I'm trying to compile the cdr_mysql module, but I am receiving > error messages. > I have installed mysql-devel. > > Here is the output of make cdr_mysql: > > cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o > cdr_mysql.o cdr_mysql.c > cdr_mysql.c:30:26: mysql/errmsg.h: No such file or directory > cdr_mysql.c: In function `mysql_log': > cdr_mysql.c:74: `CR_SERVER_GONE_ERROR' undeclared (first use in > this function) > cdr_mysql.c:74: (Each undeclared identifier is reported only once > cdr_mysql.c:74: for each function it appears in.) > make: *** [cdr_mysql.o] Error 1 > > I have looked up CR_SERVER_GONE_ERROR, but I'm not sure how that > applies to compiling the module. > > Any help is appreciated. Could you tell me where mysql/errmsg.h is located on your distribution? We can update the Makefile to look there for that header. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Handled
It could work if it would be coming over g711 and you'd have
dtmfmode=inband set for that call
regards
Martin
On Thu, 14 Aug 2003, James Golovich wrote:
>
>
> On Thu, 14 Aug 2003, Eduardo Goncalves wrote:
>
> > I'm using G.711alaw.
> > My extensions.conf:
> >
> > ===
> > [globals]
> > TRUNK=Zap/g1
> > [sip]
> > exten => s,1,Background(demo-moreinfo)
> > exten => fax,1,Dial(${TRUNK}/${EXTEN})
> > exten => _0.,1,Dial(${TRUNK}/${EXTEN})
> > exten => _9.,1,Dial(${TRUNK}/${EXTEN})
> >
> > Is this correct?
> >
>
> The last time I looked at the code, fax would only be detected if they
> came in on a Zap channel. So if the fax was coming in on a SIP channel
> then it would not work.
>
> James
>
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>
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Re: [Asterisk-Users] '#' doesn't work for me
Hi Brian, ATA is in SIP mode, and RFC2833 is used. Something else to check? Thanks, Dan - Original Message - From: "Brian West" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 14, 2003 6:16 PM Subject: Re: [Asterisk-Users] '#' doesn't work for me > Accually it will work with any codec if you use rfc2833. G711 is only > needed if you are passing DTMF inband. > > bkw > > On Thu, 14 Aug 2003, Martin Pycko wrote: > > > It works only with G711 (ulaw/alaw) > > > > regards > > Martin > > > > On Thu, 14 Aug 2003, Dan wrote: > > > > > Hi, > > > > > > I cannot use '#' to initiate transfers. > > > I have tried on different phones (7960, ATA, X-Lite). > > > When I press '#' during a call, nothing happen. > > > I have both T and t switches in Dial application. > > > The transfer function works with Flash key on ATA, but in a very strange > > > wayThe final destination is hunged up and then automatically called by > > > the initial caller... This behavior request to put on hook the phone > > > connected to the ATA in order to accept the transfer. During this period the > > > phone is busy for the caller, so I must use some tricks in the dialing macro > > > in order to acomodate this. > > > > > > Any other suggestions to better solve the transfer function? > > > > > > > > > BR, > > > Dan > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CODEC & DTMF
No i don't think so.. - Original Message - From: "George Lin" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 14, 2003 9:33 PM Subject: [Asterisk-Users] CODEC & DTMF > > Dear all, > > I like to know if the DTMF option is related to the codec or not. Can a SIP > phone with g729 codec to access asterisk voicemail2 in case the asterisk > does not have g729 license ?? If yes, what is the DTMF option inband or > outband ??? Is there any successful experience ??? > > Regards, > > George Lin > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
Accually it will work with any codec if you use rfc2833. G711 is only needed if you are passing DTMF inband. bkw On Thu, 14 Aug 2003, Martin Pycko wrote: > It works only with G711 (ulaw/alaw) > > regards > Martin > > On Thu, 14 Aug 2003, Dan wrote: > > > Hi, > > > > I cannot use '#' to initiate transfers. > > I have tried on different phones (7960, ATA, X-Lite). > > When I press '#' during a call, nothing happen. > > I have both T and t switches in Dial application. > > The transfer function works with Flash key on ATA, but in a very strange > > wayThe final destination is hunged up and then automatically called by > > the initial caller... This behavior request to put on hook the phone > > connected to the ATA in order to accept the transfer. During this period the > > phone is busy for the caller, so I must use some tricks in the dialing macro > > in order to acomodate this. > > > > Any other suggestions to better solve the transfer function? > > > > > > BR, > > Dan > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Park and out-going trunk calls.
If you add "t" to you out-going trunk "Dial" lines: exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t) exten => _NXX,2,Congestion so that you can still use park to park a call or transfer the phones, You have a problem of not being able to use "#" on external IVR systems. Is there any solution to this problem? Other then training hundreds of users not to try and park calls that you originate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
On Wed, 2003-08-13 at 09:46, Dave Cotton wrote: > I've had a few problems with my system holding the line after a call has > been made, just now I rebooted and noticed the following in > /var/log/messages When you say "holding the line", do you mean that asterisk still believes a channel is in use even after you hang up? If so, I've seen the same thing happen several times with the X100P. If I do "show channels" it will show one of my SIP phones connected to one of the outside lines, but if I check that SIP phone, it is not in use, and there is no way to re-activate the channel from the SIP phone. Running "soft hangup " will hangup the channel (you don't need to reboot). I'm not entirely sure what causes it. So far, I've only seen it happen from 2 of our 9 SIP phones, but they're the ones most often on the phone. It always involves an outside line, so I believe the X100P is the problem, but I can't be sure. What other information can I gather to pinpoint the problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Hi,
I exactly got the same problem on the Belgian network. I have tried to recompile the
wcfx0 driver with the FCC line
commented and I have created a zone for Belgium in zonedata.c (see below with the
values I know. I'm not sure of
call wait, dial recall and record tone). Everything works fine but I still got the
same problem. The system does
not detect a external line hangup.
extract of zonedate.c
[...]
{ 8, "be", "Belgium", { 1000, 3000 },
{
{ ZT_TONE_DIALTONE, "450" },
{ ZT_TONE_BUSY, "450/150,0/150" },
{ ZT_TONE_RINGTONE, "450/1000,0/3000" },
/* XXX I'm making up the congestion tone XXX */
{ ZT_TONE_CONGESTION, "450/250,0/250" },
/* XXX I'm making up the call wait tone too XXX */
{ ZT_TONE_CALLWAIT, "450/300,0/1" },
/* XXX I'm making up dial recall XXX */
{ ZT_TONE_DIALRECALL,
"!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440" }
,
/* XXX I'm making up the record tone XXX */
{ ZT_TONE_RECORDTONE, "1400/500,0/15000" },
{ ZT_TONE_INFO, "!950/330,!1400/330,!1800/330,0" },
{ ZT_TONE_STUTTER,
"!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,
!0/100,!440/100,!0/100,440" } },
[...]
/etc/zaptel.conf
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
fxsks=1
loadzone=be
defaultzone=be
DAA mode is CTR21 as I can see in /var/log/messages
Aug 13 20:31:41 ws183 kernel: wcfxo: DAA mode is 'CTR21'
Any other ideas on how to solve this issue?
Regards,
Dave Cotton wrote:
>
> I've had a few problems with my system holding the line after a call has
> been made, just now I rebooted and noticed the following in
> /var/log/messages
>
> Aug 13 17:23:15 Sheriff kernel: wcfxo: DAA mode is 'FCC'
>
> in the file wcfxo.c the following structure is initialised as below
> which would suggest that FCC is wrong for France or pretty well all of
> Europe.
>
> static struct fxo_mode {
> char *name;
> int ohs;
> int act;
> int dct;
> int rz;
> int rt;
> int lim;
> int vol;
> } fxo_modes[] =
> {
> { "FCC", 0, 0, 2, 0, 0, 0, 0 }, /* US */
> { "CTR21", 0, 0, 3, 0, 0, 3, 0 }, /* Austria, Belgium,
> Denmark, Finland, France, Germany,
>
> Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
>
> Norway, Portugal, Spain, Sweden, Switzerland, and UK */
>
> Any thoughts as to if this is the root of the problem.
>
> --
> Dave Cotton
> Directeur
> Linux Autrement
> 193 rue Marcel Cerdan
> 84270 Vedene
> 04 90 23 30 81
> "Internet Sheriff Technology" revendeur en France
>
> IAX 17004902330
>
> ___
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Re: [Asterisk-Users] FXO mode
On Thu, 2003-08-14 at 12:24, Andy Powell wrote: > Can't find the message in a search.. but below is a msg retreved from my > archive.. > > this is what Mark sent a little while ago > I have no idea if it actually does anything to the card, but on a modprobe I > do get a msg saying it's using CTR21 > > Andy > > > > >I'm in Paris right now and can't test this change, but I've been > >researching the DAA and there are a few international settings I can > >change, so I've changed the driver in CVS so that you can specify > >the operational mode. Try "modprobe wcfxo opermode=1" if you're in most > >of Europe and that should switch to CTR21 mode which slightly modifies a > >few of the electrical characteristics of the DAA. > > > >As we add modes you'll be able to see them with "modprobe wcfxo > >opermode=-1" and then doing a dmesg. > > > >Anyway all you folks that had some trouble like this try it out and let me > >know if it makes any difference. > > > >Mark > > > >___ Thanks Andy. The stage I'm at at the moment is that I've removed the code for the US and so dmesg will show CTR21 without the modprobe option. But I don't think that is the whole problem. Last night I posted showing that the problem is repeatable and only occurs in one certain circumstance. I think it is within voicemail.c. If the caller exits voicemail by pressing # the line is dropped correctly, if they just hang up voicemail continues to record. I put some debugging statements into voicemail.c and I think that a condition statement is never reached so the line is held up. The routine in question is 14 pages long, so it reminds me of my Cobol days when we used to lay the printouts along the corridor to debug them. As far as the option to modprobe is concerned couldn't zaptel.conf be used for this as it would be more obvious, I only heard about the option from your post last night. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
Hi Martin, I use ATA-186 (G.711) with two analog phones. I can transfer using Flash, but nothing happen when press on '#'... There is something else I have to check? Thanks, Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 14, 2003 5:56 PM Subject: Re: [Asterisk-Users] '#' doesn't work for me > It works only with G711 (ulaw/alaw) > > regards > Martin > > On Thu, 14 Aug 2003, Dan wrote: > > > Hi, > > > > I cannot use '#' to initiate transfers. > > I have tried on different phones (7960, ATA, X-Lite). > > When I press '#' during a call, nothing happen. > > I have both T and t switches in Dial application. > > The transfer function works with Flash key on ATA, but in a very strange > > wayThe final destination is hunged up and then automatically called by > > the initial caller... This behavior request to put on hook the phone > > connected to the ATA in order to accept the transfer. During this period the > > phone is busy for the caller, so I must use some tricks in the dialing macro > > in order to acomodate this. > > > > Any other suggestions to better solve the transfer function? > > > > > > BR, > > Dan > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for pricing on complete setup...
Chris, Your project sounds right up our alley! Your price constraints are quite tight, but we may be able to work something out. I have a few questions I need clarified before we can provide you with a quote. I'm assuming we're getting you 4 phones instead of 8. - Does the $2500-3000 price include the 4 phones? - Does each person need their own fax machine (Personal Fax for each VM box)? - Would you be amicable to other phones than the SNOM-200 or Cisco 7940 (a way to reduce over all costs)? Thanks, -- Ashley Jones [EMAIL PROTECTED] http://vergeworks.com 510.531.3739 x202 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Hale Sent: Tuesday, August 12, 2003 10:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Looking for pricing on complete setup... All - If I supply the box and the OS, what's a price for someone to set it all up for us and include 4 SNOM-200 or Cisco 7940 phones? We'd like to handle an 8x16 system (from what I've read, we can use the Zhone for this) I'd like to see what kind of pricing I can get for this setup, as well as support for the first 3 months... Questions and pricing off-line, please. We'd be looking for the following options: 1. Voicemail 2. 8x16 capacity (maybe just 4x8 now if that's possible) 3. Caller ID on phones 4. 3-way calling 5. Local conferencing 6. Remote message retrieval 7. VM notification to e-mail (or even attaching VM to an e-mail). 8. Personal Fax for each VM box 9. What else am I missing?? :) As you can probably tell, I'm pretty new to all this. I'm trying to avoid purchasing a small KSU for a new office we have, and wanted to see what a completely outsourced system would cost. I know we can get a KSU with VM for about $2500-$3000, so if we're to go with *, it would need to be under this $$. Thanks for the replies... Chris Hale [EMAIL PROTECTED] http://www.peaknetworks.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which version of MySQL are you running?
I am trying to compile the cdr_mysql module but I am getting errors. I have MySQL version 4.0.11a installed on my box (Mandrake 9.1). As far as MySQL packages, I have installed: MySQL-4.0 MySQL-client MySQL-devel MySQL-common libmysql I have the latest CVS source for Asterisk. When I run make cdr_mysql, I get a bunch of "undefined reference to" statements. Does anybody know why this may be happening? Also, does anybody know of any web-based email sites which have the "In-Reply-To" feature? It would really help out with the Asterisk mailing list. Thanks _ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk behind NAT?
Is this not just a case of a new entry in sip.conf EXTERNIP = with the code for the contact header modified to use it (if present). Then the external firewall is set to forward the rtp and 5060 to * .. I know many people either have sip aware firewalls (as i do) or their * box has a real IP, but the number of people requesting this feature seems to be growing by the hour. I'm trying to get this working for quite a number of FWD users, at the moment I'm trying to fudge it with partysip... it's not very pretty and requires a linux iptables based firewall it's not big, it's not clever and it's certainly not funny Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:19 Dave Cotton wrote: >On Wed, 2003-08-13 at 10:59, John Todd wrote: >> This is starting to sound like a feature request, > >Absolutely, that would be the real icing and cherry on the cake all in >one go. > >Totally seamless coms behind a NAT firewall. >IAX, Analog, ISDN, SIP and H323, etc..., if Pavlov could see me now. > >-- >Dave Cotton <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h extension seems to wipe variables?
Brian West wrote: Correct me if i'm wrong but doesn't the cdr modules log the call duration? If you look at the last sentence of my post: Storing stuff using the cdr isn't really an option. This is because I want to add other things to my call log that CDR doesn't support (for custom IVR apps and the like), and I'd rather not have to write scripts to pull stuff from the CDR database and sync it with a table in another database, which would be really ugly. -- Alastair Maw <[EMAIL PROTECTED]> MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode <2147483647.1060797007@[192.168.1.210]> <1060794258.27544.62.camel@RobinHood.LinuxAutrement.com>
On Wed, 2003-08-13 at 11:13, Emmanuel Bergmans wrote: > In order to test CTR21, I was forced to comment the line in the source file as I did > not find a define or a > zaptel.conf directive. It's really bad but... In my case this change has not solved > the problem (see previous > posting) Well, I'm in the US, and I still have the problem, so I'm assuming the problem isn't some European-only problem. Mine is sporadic, however - if you're getting the same thing consistently, then maybe your problem is worse. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile cdr_mysql
I'm trying to compile the cdr_mysql module, but I am receiving error messages. I have installed mysql-devel. Here is the output of make cdr_mysql: cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o cdr_mysql.o cdr_mysql.c cdr_mysql.c:30:26: mysql/errmsg.h: No such file or directory cdr_mysql.c: In function `mysql_log': cdr_mysql.c:74: `CR_SERVER_GONE_ERROR' undeclared (first use in this function) cdr_mysql.c:74: (Each undeclared identifier is reported only once cdr_mysql.c:74: for each function it appears in.) make: *** [cdr_mysql.o] Error 1 I have looked up CR_SERVER_GONE_ERROR, but I'm not sure how that applies to compiling the module. Any help is appreciated. _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile cdr_mysql
I am running Mandrake 9.1, and MySQL 3.23.57-1; and yes, I would think that /usr/src/usr/include/mysql is not the right place for errmsg.h. What can I do to get around this? I changed the cdr_mysql.c file: #include Changed to #include But I get the following error: [root cdr]# make cdr_mysql cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o cdr_mysql.o cdr_mysql.c cdr_mysql.c:40: parse error before "mysql_lock" cdr_mysql.c:40: warning: excess elements in scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: excess elements in scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: excess elements in scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: braces around scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: excess elements in scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: excess elements in scalar initializer cdr_mysql.c:40: warning: (near initialization for `mysql_lock') cdr_mysql.c:40: warning: data definition has no type or storage class make: *** [cdr_mysql.o] Error 1 Any help is always appreciated. On Wednesday 13 August 2003 03:24 pm, Jerk Face wrote: /usr/src/usr/include/mysql/errmsg.h The version of MySQL that I'm running is 3.23.57-1 What distribution are you running? That's a pretty braindead place to put the mysql header files. You'd think someplace like /usr/include, /usr/local/include, or even /usr/local/mysql/include, but /usr/src/usr/include shouldn't be the correct location on any Linux system. -Tilghman _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP NAT question
Hello all, I am sorry to bring the old question to the community. But I cannot find any answer in the google. I want to deploy multiple SIPs phone in our office. And we have shutdown the firewall at our office router(with ip 211.x.x.x). we have deployed the asterisk with IP 218.x.x.x. All SIP phones have 192.x.x.x. When the SIP phone is power on, they are registered in the asterisk. we can check at asterisk side by issueing "sip show peers", and all the phones are associated with 211.x.x.x:port-number. PRoblem: Now some times the sip can receive call, and some time it cannot recieve call. When we dumping the sip log, and see that asterisk tried to INVITE the specified SIP phone with the 211.x.x.x:port-number, and was failed after 5 times. But the call orginated from SIP phone is always OK. Questions are: 1. Does asterisk remember the mapping between 192.x.x.x AND 211.x.x.x:port-number ? 2. When a call to a sip phone, is it asterisk responsiblility to map the 211.x.x.x:port-number to the 192.x.x.x, and send to the office router ? OR it is the office router to remeber all the mapping between each sip phone 192.x.x.x and 211.x.x.x:port-number, and asterisk juts sends the 211.x.x.x:port-number to the office router ?? 3. If it is the office router's responsiblity, what should we configure the office router even there is no firewall??? Please advise , and thanks alot. George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
In order to test CTR21, I was forced to comment the line in the source file as I did
not find a define or a
zaptel.conf directive. It's really bad but... In my case this change has not solved
the problem (see previous
posting)
in wcfx0.c
[...]
fxo_modes[] =
{
/* { "FCC", 0, 0, 2, 0, 0, 0, 0 }, */
/* US */
{ "CTR21", 0, 0, 3, 0, 0, 3, 0 }, /* Austria, Belgium, Denmark, Finland,
France, Germany,
Greece, Iceland, Ireland, Italy,
Luxembourg, Netherlands,
Norway, Portugal, Spain, Sweden,
Switzerland, and UK */
};
[...]
Dave Cotton wrote:
>
> On Wed, 2003-08-13 at 18:50, Iain Stevenson wrote:
> > Assuming this is on incoming calls,
>
> Yes
>
> > the most usual source of the problem is
> > that the telco exchange either doesn't send a disconnect pulse or the wcfxo
> > driver doesn't recognise the format used.
>
> Exactly, if * is stuck in FCC mode CTR21 will be a foreign language.
>
> Somewhere in the code I should be able to either #define what I want or
> zaptel.conf should be able to set it
> --
> Dave Cotton <[EMAIL PROTECTED]>
>
> ___
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be subject to data corruption, interception, unauthorised amendment,
viruses and unforeseen delays, for which we accept no liability.
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[Asterisk-Users] chan_capi in the US
For those that are using chan_capi in the US, how do you have your line provisioned (ordering code)? Are you using CACH EKTS? thanks! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Errm, no... does that mean you'll personally check to see if my line is busy or not ;P will try it now... Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:58 Martin Pycko wrote: >Did you try BUSYDETECT_MARTIN in asterisk/Makefile ? > >regards >Martin > >On Thu, 14 Aug 2003, Andy Powell wrote: > >> Hi Dave, >> >> I have a similar problem, I tried using busydetect and busycount but >calls kept being dropped >> at random intervals. It didn't seem to matter what i set the busycount >to. I guess it's a case >> of deciding which is more important... You can also limit the length of >the voicemails using >> >> ; Maximum length of a voicemail message >> maxmessage=180 >> >> in voicemail.conf >> >> which cuts down the length of the recorded dial tone... >> >> Andy >> >> >> >> >> >> >Thanks Andy. The stage I'm at at the moment is that I've removed the >> >code for the US and so dmesg will show CTR21 without the modprobe >> >option. But I don't think that is the whole problem. >> > >> >Last night I posted showing that the problem is repeatable and only >> >occurs in one certain circumstance. I think it is within voicemail.c. If >> >the caller exits voicemail by pressing # the line is dropped correctly, >> >if they just hang up voicemail continues to record. I put some debugging >> >statements into voicemail.c and I think that a condition statement is >> >never reached so the line is held up. The routine in question is 14 >> >pages long, so it reminds me of my Cobol days when we used to lay the >> >printouts along the corridor to debug them. >> > >> >As far as the option to modprobe is concerned couldn't zaptel.conf be >> >used for this as it would be more obvious, I only heard about the option >> >from your post last night. >> > >> >-- >> >Dave Cotton <[EMAIL PROTECTED]> >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
>> FCC mode is for the US. CTR21 is for Europe - you even pasted the info >> in your message! > >Exactly, the question really is how do you change it? > modprobe wcfxo opermode=1 HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP phone recommendation
At 08:14 13-8-2003 -0500, you wrote: Has anyone had the opportunity to use a PingTel phone with Asterisk? No, I have used the Pingtel softclient though, and it's supposed to be very similar. Works pretty well, although I seem to remember something about DTMF modes... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 problems
I'm getting the following message when I start Asterisk: WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 Did I mess up the registration key or is something else wrong? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gatekeeper
Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for microsoft operating system? Does Asterisk have a built in gatekeeper? Many thanks Wayne
RE: [Asterisk-Users] [OT] unsubscribe
On Thu, 2003-08-07 at 13:27, Andy Powell wrote: > I have to say that some listserv's do allow this .. at least > he didn't reply to 20 messages with > > REMOVE > > in them True. I've seen that. I guess I'm just not on the right lists. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing question
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten => _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call as if
the user didn't dial 9? I try this (see below) and it failed:
exten => _9NX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just missing
something...
Regards,
M. Gamble
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Re: [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk?
Vonage got Cisco to include a "password protect the config" in the latest version of the firmware, and as far as I know now all the Vonage ATAs are forever destined to be used with Vonage and only Vonage. Cell providers do the same, but they help you unlock the phone after a set period - one or two years. I would hope that Vonage change their policy to the same - vendor lock in for a limited period of time. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This is how to set ATA186 for different standards of CallerID format
This is what I have found on the Cisco web site: CallerIdMethod (a parameter from the web interface) Description This 32-bit parameter specifies the signal format to use for both FXS ports for generating Caller ID format. Possible values are: Bits 0-1 (method)-0=Bellcore (FSK), 1=DTMF, values 2 and 3 are reserved. If method=0, set the following bits: Bit 2-Reserved. Bit 3 to 8-Maximum number of digits in phone number (valid values are 1 to 20; default is 12) Bit 9 to 14-Maximum number of characters in name (valid values are 1 to 20; default is 15) Bit 15-If this bit is enabled (it is by default), send special character O (out of area) to CID device if the phone number is unknown. Bit 16-If this bit is enabled (it is by default), send special character P (private) to CID device if the phone number is restricted. Bits 17 to 27-Reserved. If method=1, set the following bits: Bits 3-6-Start digit for known numbers (valid values are 12 for "A," 13 for "B," 14 for "C," and 15 for "D.") Bits 7-10-End digit for known numbers (valid values are 11 for "#," 12 for "A," 13 for "B," 14 for "C," and 15 for "D.") Bits 11-Polarity reversal before and after Caller ID signal (value of 0/1 disables/enables polarity reversal) Bits 12-16-Maximum number of digits in phone number (valid values are 1 to 20) Bits 17 to 19-Start digit for unknown or restricted numbers (valid values are 4 for "A," 5 for "B," 6 for "C," and 7 for "D.") Bits 20 to 22-End digit for unknown or restricted numbers (valid values are 3 for "#," 4 for "A," 5 for "B," 6 for "C," and 7 for "D.") Bits 23 to 24-Code to send to the CID device if the number is unknown (valid values are 0 for "00," 1 for "00," and 2 for "2." 3 is reserved and should not be used. Bits 25 to 26-Code to send to the CID device if the number is restricted (valid values are 0 for "10," and 1 for "1." 2 and 3 are reserved and should not be used. Bits 27 to 31-Reserved. Examples The following examples are recommended values for the CallerID Method parameter: USA=0x19e60 Sweden=0x0ff61 or 0x006aff61 Denmark=0x0fde1 or 0x033efde1 Value Type Bitmap Default 0x00019e60 (USA) BR, Dan P.S. Tested with a C&W DECT model CDW2500 it works with the default values (USA). I still try different values to make it work with my Philips DECT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zhone Zplex 10 units
OK. Thanks - I think :-) I'll go trolling on Ebay, see what comes up. Given that most of my projects take 6 months or so to get off the ground, I hate to put a bunch of money into this anyway. So, for <$1,000, I can put a 6 x 18 unit in my office & play with it to see if this is a product line I want to get into. Steven Critchfield wrote: On Tue, 2003-08-05 at 02:45, Kent Williams wrote: Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). I didn't say they didn't work. Mine has been in production use for over a year now with only a couple hiccups related to sync source and timings. Specifically after a major power failure, it was possible for the Zhone to sync to asterisk and not to the Telco until I had unplugged asterisk. Now we no longer have the telco plugged into the Zhone and have no problems with it. The sync problem is not why we changed our setup. -Original Message- From: John Schmerold [mailto:[EMAIL PROTECTED] Sent: Tuesday, 5 August 2003 12:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zhone Zplex 10 units Thanks for the Zplex heads up. Steven Critchfield wrote: On Mon, 2003-08-04 at 13:13, John Schmerold wrote: I frequently see Zhone Zplex 10 units on Ebay - cheap. What's the story on these? Are they flaky? search the archives. Tough to configure? tough, no, pain in the ***, yes Any other issues that come to mind? search the archive, that is why it is there. I don't see them listed on Zhone's website (except in support), so I suspect they've discontinued the product, but if it's a good product I could use it to learn Asterisk. Thats funny since they don't really even act like they want to support them. -- John Schmerold Katy Computer Systems, Inc 20 Meramec Station Rd Valley Park MO 63088 314-316-9000 v 775-227-6947 f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- John Schmerold Katy Computer Systems, Inc 20 Meramec Station Rd Valley Park MO 63088 314-316-9000 v 775-227-6947 f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax for hiding caller ID but stillpassing ANI?
Lorenzo -
I've submitted a feature request with this patch
(http://bugs.digium.com/bug_view_page.php?bug_id=052). Your
patch isn't completely descriptive, since I still don't know how you
set the hidecallerid value from within a dialplan. Can you explain a
bit more, please? Have you submitted a disclaimer to Digium so this
patch might be added if it's seen as a useful addition?
Linus -
Thanks for the specifications. Did you have a patch or comments on
how this might be implemented in the code?
JT
We did something like this in chan_zap at pri_call() time:
case SIG_PRI:
[...]
if (ast->callerid) {
strncpy(callerid, ast->callerid, sizeof(callerid)-1);
ast_callerid_parse(callerid, &n, &l);
if (l) {
ast_shrink_phone_number(l);
if (!ast_isphonenumber(l))
l = NULL;
}
}
[...]
if (l) {
pres = ast->hidecallerid ?
PRES_PROHIB_USER_NUMBER_NOT_SCREENED :
PRES_ALLOWED_USER
} else
pres = PRES_NUMBER_NOT_AVAILABLE;
if (pri_call(p->pri->pri, p->call,
p->digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEEC
p->prioffset, p->pri->nodetype == PRI_NETWORK ? 0 : 1, 1, l,
p->pri->dialplan - 1,
c + p->stripmsd, p->pri->dialplan - 1,
((p->law == ZT_LAW_ALAW) ?PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW)))
{
[...]
where hidecallerid is a new field we added in ast_channel structure and it's
set by our apps...
As far as we can understand this should be more compliant to the q931 specs.
(and it works for us in Italy ;-)
my 2 cents,
Lorenzo
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 04, 2003 8:34 PM
Subject: Re: [Asterisk-Users] Syntax for hiding caller ID but still passing
ANI?
l is set a couple of lines above. Basically l carries the nubmer so if
there is no callerid in 'l' then we send this other flag 'callerid not
available'.
You need to choose one of these flags:
/* Presentation */
#define PRES_ALLOWED_USER_NUMBER_NOT_SCREENED 0x00
#define PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN 0x01
#define PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN 0x02
#define PRES_ALLOWED_NETWORK_NUMBER 0x03
#define PRES_PROHIB_USER_NUMBER_NOT_SCREENED0x20
#define PRES_PROHIB_USER_NUMBER_PASSED_SCREEN 0x21
#define PRES_PROHIB_USER_NUMBER_FAILED_SCREEN 0x22
#define PRES_PROHIB_NETWORK_NUMBER 0x23
#define PRES_NUMBER_NOT_AVAILABLE 0x43
I think it might be PROHIBPASSED_SCREEN but not sure. Check q931
specs.
Martin
On Mon, 4 Aug 2003, John Todd wrote:
>
> I have a question regarding the flags for hiding caller ID presentation:
>
> My customer has a requirement that they are able to specify if
> outbound calls (on a T100P) will have the caller ID displayed or not.
> This could be easily solved, of course, by not setting a caller ID
> when creating the outbound call. However, the PRI to which this
> T100P is connected _must_ see a valid caller ID, and the caller ID is
> used for billing purposes.
>
> I know that there is the ability to hide caller ID within the Zaptel
> libraries, using the presentation flags. If set correctly, the
> > expected behavior would be that the ANI would be sent to the switch,
> > but with a flag that would tell the remote switch to suppress the
> > caller ID from being transmitted to the end user.
>
> How does one activate that presentation switch from within a dialplan?
>
> Searching the archives gives me some hints, but no answers.
> Searching the code, I see in channels/chan_zap.c around line 1399
> that the PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN and
> PRES_NUMBER_NOT_AVAILABLE are referenced, but I'm not clear on where
> "l" is set, or even if that is a trigger. Can someone give me a hand
> > on syntax on this?
>
> JT
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RE: [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?
Hi, Have someone tried to use the same trick with the PCPhone application (soft SIP phone) from iConectHere which can fully support Actiontec's Iinternet Phone Wizard USB phone interface? I have tried without any success. Thanks, Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 06, 2003 11:26 PM Subject: Re: [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ? > One could imagine that you could just use the ATA Vonage box creating some > fake routes to your system ... so that if the box tries to contact > gateway.com you could point that to asterisk and make it work using the > current vonage setup. I would also recommend adding a rule to firewall > that your ATA box cannot contact outside world only the asterisk > server. > > regards > Martin > > On Wed, 6 Aug 2003, Steve Haehnichen wrote: > > > -=> On Wed, 06 Aug 2003 14:39:53 -0500, John Schmerold <[EMAIL PROTECTED]> said: > > > > > I've canceled my Vonage service because of the requirement to prefix > > > every call with a 1. > > > > (I was more bothered by the lack of CallerID Name, but the 11-digit > > dialing was indeed annoying for local calls.) > > > > The Vonage ATA-186 units shipped from Vonage before July arrived > > 'unlocked' either with no password, or a firmware version old enough > > to always allow #FACTRESET. As far as I know, the ones from > > Amazon.com arrived locked because no contract was yet in place. > > > > Sometime around mid-July, Vonage pushed out v3.16 and password-locked > > *all* the ATAs. > > > > If your box has been off-line since June, you should still be able to > > reset it with the process you described. If it has recently been > > connected, then it is not going to work. > > > > You can find out by attempting the reset. If it says "PASSWD", then > > it won't let you change any config, reset the box, or upload new > > firmware without the password. If instead it asks you to confirm with > > "*", then you're golden -- you can reset it and then connect to > > http:///dev to reconfigure. > > > > Vonage used to always charge the cancellation and leave you with the > > box. The option to return the box is new, and should address some of > > the 'green' concerns. > > > > If anyone has a method for 'unlocking' password-locked unsubscribed > > Vonage ATA's, there are a lot of interested folks out there, myself > > included. > > > > -Steve > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Trunk config
And to answer Wade's question: to limit outbound calls on a
particular path, you'd use a local db set routine. In other words,
every time a call is created to that particular SIP peer, you'd add 1
to the counter, and every time a call was hung up out of that pool,
you'd subtract one.
JT
At 3:30 PM -0400 8/7/03, Patrick wrote:
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200; Qualify peer is no more than 200ms away
defaultip=192.168.0.4
incominglimit=20 ; set limit to 20 voice channels
setting the limit to 0 (incominglimit=0) is unlimited.
to view the current lines in use --- sip show inuse from the cli.
Patrick
I've also run into the "how many lines" problem.
Possibly something similar to incominglimit= and outgoinglimit= in
h323.conf
could be implemented in sip.conf?
-wade
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Hindmarsh
Sent: Thursday, August 07, 2003 12:19 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sip Trunk config
Thanks for that,
I was looking at the extensions.conf, particularly the line in the
general
section which is
TRUNK=SIP/???
Using this method would be easier.
How do you tell asterisk how many lines are available at the gateway
Dave
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 07, 2003 12:34 PM
Subject: Re: [Asterisk-Users] Sip Trunk config
> exten => _9X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
>
> regards
> Martin
>
> On Thu, 7 Aug 2003, David Hindmarsh wrote:
>
> > Hi
> >
> > Is it possible to use a sip gateway as a trunk.
> >
> > If so, how would I do this
> >
> > David Hindmarsh
> >
> > - Original Message -
> > From: "Jamie Carl" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Thursday, August 07, 2003 12:14 PM
> > Subject: Re: [Asterisk-Users] X-Lite <-> Snom200
> >
> >
> > > Yes, over a LAN. It does it with both g.711 and GSM which
> > > both used to work. Havn't had a chance to have a REAL
> > > good look into it though.
> > >
> > > J
> > >
> > > On Wed, 06 Aug 2003 14:33:47 +
> > > "WipeOut ." <[EMAIL PROTECTED]> wrote:
> > > >*This message was transferred with a trial version of
> > > >CommuniGate(tm) Pro*
> > > >> *This message was transferred with a trial version of
> > > >>CommuniGate(tm) Pro*
> > > >> Dunno what I'm doing wrong here but I just did an
> > > >>upgrade to the latest
> > > >> version and now I get no audio at all!
> > > >> I havn't changed a single thing. Is there anything
> > > > >>special I need to do
> > > > >> to get this to work again?
> > > > >>
> > > > >> I get a quick 'chirp' of audio, which you can tell is
> > > > >>what I'm
> > > > >> connecting to, (ie MOH), but then nothing.
> > > >>
> > > >>
> > > >> Regards,
> > > >>
> > > >> Jamie Carl
> > > >> Email: [EMAIL PROTECTED]
> > > >> Phone: +61 414 365 466
> > > >> Jabber: [EMAIL PROTECTED]
> > > >>
> > > >
> > > >Are you connecting to * over a LAN?? I have experienced
> > > >the "chirp" when the phone was trying to use G.711 over a
> > > >dial up link so there was not enough bandwidth..
> > > >
> > > >
> > > >--
> > > >__
> > > >http://www.linuxmail.org/
> > > >Now with e-mail forwarding for only US$5.95/yr
> > > >
> > > >Powered by Outblaze
> > > > >___
> > > >Asterisk-Users mailing list
> > > >[EMAIL PROTECTED]
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > Regards,
> > >
> > > Jamie Carl
> > > Jazz Inc.
> > > Email: [EMAIL PROTECTED]
> > > Web:www.jazz-inc.net
> > > Phone: +61-414-365-466
> > > Jabber: [EMAIL PROTECTED]
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
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> >
>
> ___
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Re: [Asterisk-Users] chan_capi: Hanging channels - again
Hi Roy, always use latest chan_capi. the bug is fixed in 0.2.4a. today 0.2.4b is online which fixes some issues with sending dtmf and a small enhancement to capiECT. capi on! regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Die, 2003-08-05 um 15.41 schrieb Roy Sigurd Karlsbakk: > hi all > > sad news > > those channels are hanging again. > > Anything more I can do to troubleshoot this? > > *CLI> show channels > Channel (ContextExtensionPri ) State Appl. Data > CAPI[contr2/22545079]/57 (ola22545079 1 ) Up Bridged Call > SIP/ola-ac25 >SIP/ola-ac25 (default51676840 1 ) Up Dial > CAPI/22545079:b51676840|300|T > CAPI[contr2/22545066]/55 (torgeir22545066 1 )Down (None) > (None) > CAPI[contr2/22545066]/54 (torgeir22545066 1 )Down (None) > (None) > 4 active channel(s) > *CLI> > > -- > Roy Sigurd Karlsbakk, Datavaktmester > ProntoTV AS - http://www.pronto.tv/ > Tel: +47 2254 5070 (work) > +47 9801 3356 (mobile) > > Computers are like air conditioners. > They stop working when you open Windows. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie just starting out with *
Chris: Try not to be so worried about sound card, analog (FXO/FXS), digital (ISDN, BRI, PRI) and what is available by connecting device. The channel drivers take care of making the devices available to Asterisk. In turn Asterisk makes all the features such as voice mail, call parking, and conference bridges available to the channels. It is a beautiful and flexible design. many thanks to Mark! With a few exceptions most features will be available to all connection methods. Yes you can upgrade the TDM400P. My thinking was to get at least two ports on it when I purchased it originally so I could call phone-to-phone internally without using our only external phone line. That way I could learn to configure and use asterisk with out annoying my friends in family trying to call in. Zapateller does not stop telemarketers it stops the predictive dialers they use. Ever received a call and answered hello two or three times before you get a person? That is a predictive dialer loaded with a list of numbers dialing all of their phone lines as quickly as possible. It will do it more efficiently than a group of agents with a phone number list in hand. When you answer the dialer takes a moment to diagnose the fluctuations in the voice. The dialer makes a determination if someone even answered and if so if you are a person or an answering machine. If you turn out to be what it thinks is a real person it must find an available agent. That is what causes the pregnant pause. It has to find someone since you turned out to be a real person. Now what Zapateller can do is answer the phone and play the SIT (special information tone). When the dialer hears this it thinks your number is no longer in service and hopefully removes your number from that companies list. The other thing it can do is just play the SIT tones to any incoming call not providing caller id. Just take the plunge, buy the equipment, play around and come back here when you get stuck, Matt -Original Message-From: Chris Hirsch [mailto:[EMAIL PROTECTED]Sent: Tuesday, August 05, 2003 12:13To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Newbie just starting out with * So its sounds like I do have a clue then...can analog devices have their own extension and do call parking, and paging and all that? I assume the caller id gets passed from the POTS CO to the internal phones? So from my understanding I can get a TDM400P with one port now and upgrade to additional ports? Thats what comes with the dev kit right?Tell me more about Zapateller..is that the script that I've seen a description of that gives the fake line-disconnected tone?McAughan, Matt wrote: Chris: I started using Asterisk for very much the same reason. To blast those telemarketers and to improve my knowledge of PBX and telco. You have got a good start for a newbie. Yes the Wildcard X100P will terminate the POTS CO in to your Asterisk Linux Box. Then you have to figure out how to get everything "internal" connected up to it. I have a TDM400P. It can be purchased with up to 4 ports. I purchased three to save money. That gives me three "internal" extensions to plug analog phones in to. I just use three cordless phones with the base stations plugged in near the Asterisk computer. We leave one phone in the kitchen, one in the garage, and one in the bedroom. They can call each other when we are too lazy to go get one another or too far to scream at each other. They can all also share the one "external" line. Asterisk has been wonderful using Zapateller to blast those damn predictive dialers. The Asterisk voice mail has been wonderful too as it sends the recorded message to me and my wife at work as an attachment to an email. Best of luck, Matt -Original Message- From: Chris Hirsch [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 05, 2003 11:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie just starting out with * Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay. That will get me two analog devices on the system. If I have four analog devices (2 normal phones, 1 fax and one 4 phone cordless system) is this the best setup? Do I need the TDM10B with the Asterisk TDM Dev Kit or does that just let me do one analog phone into the system? When converting from analog to VOIP do I get all the same features that I would if I got a TDM400P (4 ports of analog devices)? As I said I'm new and I would
Re: [Asterisk-Users] AgentCallbackLogin
Beautiful. Thanks! Jim Friedeck TC wrote: Jim I added a patch that mark got into cvs last night use ackcall=no in agent.conf -Original Message- From: Jim Friedeck <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Date: August 6, 2003 1:46 PM Subject: Re: [Asterisk-Users] AgentCallbackLogin Thanks! I don't know why that works. Where is that behaviour documented? Also is there any way to turn off the '#' to confirm so the agent can just pick up the phone to answer the call? I'm running CVS as of about an hour ago. Thanks again. Jim Friedeck - The Traveller wrote: Hey Jim, On Wed, Aug 06, 2003 at 15:12:50 -0500, Jim Friedeck wrote: I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: [...] extensions.conf: [c_in_1];internal lines (up to 48 agents and admins) exten => 400,1,AgentCallbackLogin(|c_in_1) [...] I don't understand where the default context comes from in the message 'No such extension/context [EMAIL PROTECTED]'. Where do I tell the queue app the proper context? Any ideas? Try adding an "@" in front of the context in the argument to "AgentCallbackLogin", like so: exten => 400,1,AgentCallbackLogin(|@c_in_1) Also make sure you're running the latest CVS-version, as the functionality has just been added yesterday or so. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
Dave, did you ever figure out why the last digit was dropping? I am having the same
problem right now even with the fixes suggested to you..
I am trying to add the following to my extensions.conf:
exten => _1650XXX,1,Dial,Zap/1/${EXTEN}
with debug on for the Zap/2-1 line (the developer's kit) I see all digits dialed but
the last.
-- John
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[Asterisk-Users] chan_OH323
Hello, I downloaded the chan_oh323. I experience few problems: When I dial from console I get all the object creation and deletion message, and when a call get connected it gives me the following output. Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! !Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! this message keep outputed to the console untill I end the call. When I dial in to asterisk, I get WARNING[524312]: File chan_oh323.c, Line 948 (oh323_read): H323:7160: Invalid size for G.729 (2 bytes). then i got disconnected. I am using Digium g.729 codec. In oh323.conf i set codec=G729 Any idea? Foong
Re: Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACKINFO]
Try to set the "frames" option in section [codecs] to a reasonable value, say 20 for G711, 2 for G7231, 4 for GSM. Also, do you get segfaults when you try the same with just one codec enabled? Michael. Sip Rtp wrote: Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #4 0x47784af3 in RTP_DataFrame::SetPayloadSize(int) () from /home/sip/openh323/lib/libh323_linux_x86_r.so.1.12 #5 0x4776ea76 in H323_RTPChannel::Transmit() () from /home/sip/openh323/lib/libh323_linux_x86_r.so.1.12 #6 0x4776ba84 in H323LogicalChannelThread::Main() () from /home/sip/openh323/lib/libh323_linux_x86_r.so.1.12 #7 0x47c756f1 in PThread::PX_ThreadStart(void*) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #8 0x4002e332 in start_thread () from /lib/tls/libpthread.so.0 Rgds Sip Rtp - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 08, 2003 3:56 PM Subject: Re: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk Sip Rtp wrote: Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. More information is needed. You should provide a backtrace of the core file, the screen log of Asterisk (generated when executed with "asterisk -vvvcdg"), your oh323.conf and the important sections of extensions.conf. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in the * ). But now problem comes in the outbound as when i use a extension like exten=>12,1,Dial(OH323/12) Then the call goes through but i don't hear any voice. So my two problems are 1.Why asterisk gives seg. fault when i dial exten on 711 codec from ATA 2.Why can't i hear voice from * to ATA when i use 723 in ATA. for 2nd i think that there is mismatch between the codecs so can we change the priority order of the codecs used in the * or Oh323 and if yes, then how? Please ask if any further Input is required. Rgds Manoj K Gupta Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] .:. .: .. .Stottering audio ??
Installed Asterisk on Redhat 9.0 - and not channeled to PSTN/PLMN networks (no XP100 or special hardware) yet When I use * with a softphone (SIP) - Asterisk answers the call but voicemail or other playbacks are STOTTERING for the first 30 secs (approx.)It happens more often when I start Asterisk with -vvvgc (less in -c). Following message displayed (loads of them!) NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Does anyone know where the problem is - what am I doing wrong or is it a known bug? Thanks in advance! Jeroen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk
Hello Michael, Here is the information which you asked for. Please look into it..If you need more info tell. I am using the following call scenerio.. I am dialing to PBX from openphone by dialing a PSTN number connected to * through development kit of digium. then i press 12 as the extension to dial for ATA connected to GNUGK. Thanks for the time Rgds Sip Rtp Written by Mark Spencer <[EMAIL PROTECTED]> = DEBUG[1074447520]: File config.c, Line 712 (__ast_load): Parsing /etc/asterisk/logger.conf Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount Asterisk Management interface listening on port 5038 == RTP Allocating from port range 1 -> 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SetAccount] == Registered application 'SetAccount' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' Asterisk Dynamic Loader Starting: [chan_modem.so] => (Generic Voice Modem Driver) == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] => (Call Parking Resource) [res_parking.so] => (Call Parking Resource) == Registered application 'ParkedCall' [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' [res_indications.so] => (Indications Configuration) -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAXpeers == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 5036 [chan_sip.so] => (Session Initiation Protocol (SIP)) == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 [skipping chan_oss.so] [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == MGCP Listening on 0.0.0.0:2427 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers WARNING[1074447520]: File chan_iax2.c, Line 5061 (set_config): Ignoring port for now == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 [chan_local.so] => (Local Proxy Channel) [chan_phone.so] => (Linux Telephony API Support) [chan_zap.so] => (Zapata Telephony w/PRI) -- Registered channel 1, FXS Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling -- Registered channel 3, FXS Kewlstart signalling -- Registered channel 4
[Asterisk-Users] X100P delivery
I live in Japan and last Sunday I bought my first X100P to see if it really works for my H323 application. How long time it should take to be delivered? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ansering machine/voicemail detect?
Please learn to start new thread properly. On Thu, 2003-08-07 at 12:17, Garry Adkins wrote: > I've been playing with an asterisk box for about 6 months, (bought an > FXO card, etc.)... > > I was thinking about having the system "deliver" my voicemail from the > asterisk machine to me at work... I haven't found anything in the > documentation to help. > > > It would work something like: > > > Voicemail comes into the asterisk machine, > * Calls me at work > Plays message for me to enter PIN for voicemail > Retrieve Voicemail > Hangup. > > > However, if it got my voicemail at work (due to being on the phone or > out of the office), I'd like it to do something like: > Voicemail in * > * Calls me at work > Notices that it's voicemail (Possibly due to no pause at the > beginning, just continuous talking?) > Just plays a message that I have voicemail at home. > Hangs up. Fairly easy, write an app that watches over your INBOX every so often and then drops a /var/spool/asterisk/outgoing call in there to call you at work with your mailbox specified. The only tricky part here is knowing you have sent the call and not try and do it again just because you haven't listened to it yet. Of course you could just get an email copy of the file. Makes this a lot easier to deal with now doesn't it? > Possible? How could it tell that it got an answering device? currently it doesn't -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing -- an opinion
Jan Rychter wrote: Please try to find a better solution. The DSP Group owns G.729. There is nothing anyone can do, they have us by the family jewels. We use iLBC and found it to be very acceptable in quality and bandwidth usage and its free. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
http://www.bkw.org/~brian/ata.html Pay attention to connectmode and audiomode Its how I set it up and it works. bkw On Thu, 14 Aug 2003, Dan wrote: > Hi Brian, > > ATA is in SIP mode, and RFC2833 is used. > Something else to check? > > Thanks, > Dan > > > - Original Message - > From: "Brian West" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, August 14, 2003 6:16 PM > Subject: Re: [Asterisk-Users] '#' doesn't work for me > > > > Accually it will work with any codec if you use rfc2833. G711 is only > > needed if you are passing DTMF inband. > > > > bkw > > > > On Thu, 14 Aug 2003, Martin Pycko wrote: > > > > > It works only with G711 (ulaw/alaw) > > > > > > regards > > > Martin > > > > > > On Thu, 14 Aug 2003, Dan wrote: > > > > > > > Hi, > > > > > > > > I cannot use '#' to initiate transfers. > > > > I have tried on different phones (7960, ATA, X-Lite). > > > > When I press '#' during a call, nothing happen. > > > > I have both T and t switches in Dial application. > > > > The transfer function works with Flash key on ATA, but in a very > strange > > > > wayThe final destination is hunged up and then automatically > called by > > > > the initial caller... This behavior request to put on hook the phone > > > > connected to the ATA in order to accept the transfer. During this > period the > > > > phone is busy for the caller, so I must use some tricks in the dialing > macro > > > > in order to acomodate this. > > > > > > > > Any other suggestions to better solve the transfer function? > > > > > > > > > > > > BR, > > > > Dan > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wierd Message
Can you send a trace from your screen after you turn of the debug in /etc/asterisk/logger.conf console => blabla,debug regards Martin On Tue, 5 Aug 2003, Ricardo Villa wrote: > Is it possible to know what application? The extension I'm daling is very > simple: > exten => 1001,1,Dial(SIP/1001,15) > exten => 1001,2,Voicemail2(u1001) > > As soon as the Voicemail picks up the NOTICE line appears multiple times on > the console. > > Thanks, > Ricardo > > - Original Message - > From: "Martin Pycko" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, August 05, 2003 3:57 PM > Subject: Re: [Asterisk-Users] Wierd Message > > > > It means that some application scheduled an execution of some routine in > > the past, eg: it will never be executed since it's way in the past ... > > > > regards > > Martin > > > > On Tue, 5 Aug 2003, Ricardo Villa wrote: > > > > > Hi, > > > > > > Whenever someone leaves a Voicemail in our system we get this message on > the > > > console: > > > > > > NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to > schedule > > > in the past?!?! > > > > > > Does anybody know what it means? > > > > > > Ricardo. > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi-newbie question "Softswitch" and Asterisk- Is there a difference?
Q: What's the difference between Asterisk and a softswitch A: About $100,000 Soft switch - Hard to afford! Regards, Steve Bruce Ferrell wrote: I've been working in the VoIP industry for just a bit over a year now... Mostly taking care of the underlying systems. I've now reached the point where I'm being drawn more and more into the call processing side of things. My background is in computer and "classic" telephony systems (DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor modules and analog mux etc). I've seen one commercial product recently based on openh323proxy called a transit softswitch and now I'm looking at asterisk. Is there a difference between what asterisk is and a softswitch? Can someone explain it in small words and phrases for me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
well if you ask me, the leastrecent part would work if you reversed the logic on the metric. my other last_used mod would do a time_t on that agent the last time it was 'tried' (ast_request'd) then (i was using arrays) qsort so that (new agents) '0' would be on top, and the agent that got the most recent attempt would be on the bottom '1057174447' (below is an example) -- sorted agent array: 317 last_used: 0 -- sorted agent array: 318 last_used: 0 -- sorted agent array: 319 last_used: 0 -- sorted agent array: 300 last_used: 1057174447 that way, (for leastrecent anyway), you are always working with a full stack of agents. Brian West wrote: First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into roundrobin if that agent didn't answer. Next new caller would ring fewestcalls agent first then start roundrobin. What do you think should happen in fewestcalls? Right now it just rings the agent with the fewestcalls over and over with current app_queue logic. leastrecent from what I have been looking at will ring the agent that has least recently take a call first then if they don't answer go into roundrobin. Then the next new call coming from queue would first go to the leastrecent first then try every agent in roundrobin till answered then starting over again. New caller from queue hits leastrecent agent first. Same thing happens in leastrecent strategy. The leastrecent agent will ring over and over with current app_queue logic. Now some of you might recommend autologoff options. But that also might need some work. I don't want to log off an agent for not answering the phone only once. So here is how I would like to see autologoff work. Example: queue timeout = 20 agent autologoff = 60 The agent would have to not answer their phone 3 times in a row to get logged off. As it stands now they did not answer just once and get logged off. Thus allow for an employee to use the excuse for not working when they should be logged in and taking calls. Unless i'm wrong here. Please post your input on these options and how you would like them to see them function function. Thanks, Brian CWIS Internet Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Center RFP
I have an opportunity for a 50 seat call center requiring outbound dialling, inbound call queuing, agent management, call recording, call/skill matching, call list management, reporting, IVR, management call whisper, etc. Are there any * resellers on this list who are capable of handling a sophisticated installation such as this? If so, please contact me off list. Regards, Ray Burkholder 519 570 0689 x2002 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 support
Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM Here is the last mail that I recall seeing on the subject: From: Steve Underwood <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E1 R2 on Asterisk Reply-To: [EMAIL PROTECTED] Date: Fri, 18 Jul 2003 09:51:45 +0800 John Todd wrote: LQ (Asterisk) wrote: Dear fellows, I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina. I know that the E100P don't support it right now. Correct Does anybody developing R2 drivers? Yes. Interestingly terse reply; perhaps you can be more specific? I have an interest in the same drivers, and there was some discussion a week ago (two weeks?) on the topic, specifically about how a driver might be written, but I heard no confirmation that there was progress or any timeframes. Anyone have any encouraging updates for those of us waiting for R2? I've been more specific in the past. This was just a brief recap, since its the same question each time (OK, the country varies, but not much else). The work was held up by SARS, as the testing has been done in China. Now the SARS threat has abated, I hope we will see a polished China R2 soon. Every other country requires modifications, as no two countries implement R2 in quite the same way. However, the software has been written with all the variants in mind, and completing support for other countries should be pretty straightforward, once it is a proven platform in China. You will find some elements of R2ness in CVS. That is work I did over a year ago, then left unfinished. The DSP part of that is OK, although I have improved it recently. The rest was a lash-up, which has now been totally replaced by a solid implementation. I have a new channel driver for Asterisk, which supports R2 and PRI with the Zaptel drivers. My intention is to add other protocols, so this becomes a replacement for chan_zap. The way chan_zap works had some flexibility issues for me, so I have tried to steer everything through a generalised signalling API. On one side are plug in protocols - currently PRI and R2. On the other side is Asterisk. Hopefully, this will allow new protocols to plug in with little or no change to anything in Asterisk. This is somewhat like Dialogic's GlobalCall, but hopefully without so much of their clunkiness. :-) FAQ: Can I help with testing, and be an early adopter? No. I have all the testers I need right now. Until I have it well proven, with E1's full of calls pumping through it, I am not interesting in having other testers involved. I expect the assistance of other testers will be needed later, to deal with issues arising from the local variants of R2. Help at that time will be most appreciated. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list proposal
Perhaps there is another way to cut down on increased traffic... Specifically, I would go back to the suggestion of a collaborative website for documentation. Collecting info and organizing into Howto's would reduce the number of times people ask the same questions. Also, the documentation could grow as quickly as the project. Unfortunately, I don't have a place to host it currently. Ideally, the list would just be for issues that aren't already addressed. Any one else interested in this? Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Friday, August 08, 2003 1:25 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] list proposal > > > With the increased traffic as of late, I'm wondering if it is time to > split the list again. Specifically I am wondering if it should be split > along the various VoIP protocols and zap hardware, then leave a general > list that does configuration other than VoIP related? > > The hope is that those asking SIP or H323 questions could get help from > the various supporters while the main list can deal with transport > neutral content like extension logic and voicemail configs. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] unsubscribe
On Thursday 07 August 2003 11:10 am, Steve Meyers wrote: > On Thu, 2003-08-07 at 10:01, Justin Carlson wrote: > > unsubscribe > > Has anyone ever been on a mailing list where you could unsubscribe > simply by sending a message with "unsubscribe" in it to the mailing > list? I swear, every list I've been on, people try to do that, but > it doesn't work on any of them. Actually, all of the NetCentral lists do that. Nashville Linux Users Group uses their service, and all the KDE lists used to be hosted with them. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
If I am understanding correctly your setup looks like this..
{Asterisk}--[NAT]--Internet--[NAT]--{X-Lite}
If this is correct then you are going to have major problems getting it to work.. Your
RTP traffic is going to get very confused..
You need to get Asterisk onto a Public IP address..
I have seen many try and get the double NAT setup to work but I haven't yet heard of
anyone getting it right..
> Hi,
> thanks for that.
>
> after implementing yours and "wipeout's" suggestions (thank you both),
> x-lite changed its default codecs to G711a. which is great... and a way
> forward.
>
> but it still does not play sound when the "1000" is dialed.
>
> my * is behind nat. and my test pc is as well.
> Here are my settings:
>
> sip.conf
> [senad]
> type=friend
> secret=blah
> host=dynamic
> dtmfmode=inband
>
> thanks
>
> senad
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Nathan
> Littlepage
> Sent: 08 August 2003 19:50
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
>
>
> Change the allow=all in sip.conf to allow=alaw and see if that works.
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Senad Jordanovic
> > Sent: Friday, August 08, 2003 1:14 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue
> >
> >
> > Hi,
> >
> > X-Lite logs into * with no problems. I dial "1000" and *
> > plays greeting, but
> > i can not hear it.
> > Tried many times with the same result.
> >
> > After quite few tries * complains about:
> > -
> > WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt):
> > Maximum retries
> > exceeded on call
> > [EMAIL PROTECTED] for
> > seqno 43 (Response)
> > WARNING[81926]: File chan_sip.c, Line 2002
> > (__transmit_response): Unable to
> > determine sequence number from ''
> > -
> >
> >
> >
> > Has anyone had the same problem?
> >
> > Senad
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
__
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Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
No need for the pri debug span, the problem is the duration of the tones when using dtmfmode=rfc2833. It is way to short. A lot of IVR's just don't get enough of the tone to work. The info method still has the correct duration. Simple to test just deal another phone and hit keys, you will see what I mean. Martin Pycko wrote: type on your asterisk CLI "pri debug span " and send the trace of a broken call regards Martin On Mon, 4 Aug 2003, Stefano Finetti wrote: - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, August 03, 2003 5:52 PM Subject: Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN Are you experiencing it over PRI ? Can you send the "pri debug span trace ?Is your asterisk/libpri code very recent ? I'm experiencing this both over a PRI line (E1), with july CVS, and over a normal ISDN BRI line, with latest CVS sources (taken about a week ago). I'v tried to debug both SIP and using messages (/var/log/asterisk/messages) but i found no useful informations. It's quite important to solve this problem 'cause i'm not able to call some *very* important number used for my job (Telecom HelpDesk, and so on). Thanks, -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ansering machine/voicemail detect?
> Garry, > > yes this is possible although it would end up being quite convoluted. No, simpler than that... > >Voicemail comes into the asterisk machine, > >* Calls me at work > >Plays message for me to enter PIN for voicemail > >Retrieve Voicemail > >Hangup. > > > >However, if it got my voicemail at work (due to being on the phone or out > >of the office), I'd like it to do something like: > >Voicemail in * > >* Calls me at work > >Notices that it's voicemail (Possibly due to no pause at the beginning, > >just continuous talking?) > >Just plays a message that I have voicemail at home. > >Hangs up. 1) Some cron job runs 2) Script checks if any file in the voicemail box is newer than it's last run date/time 3) updates the date/time in a file/DB 4) drops call file into the spool dir to initiate the call from a specific context 5) Call starts and follows this process: a) Ask for PIN# b) If receive PIN, then GOTO voicemail c) Timeout (no PIN received) d) Wait long enough for your voicemail to have ansered, played the message and the beep (just time it) e) Playback(youhaveamessage) It's a bit of a hack, but should work. I have asterisk divert to my mobile, but with a timeout so I don't end up diverting the caller to my mobile's voicemail Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "Out of area" displayed as caller-id
When connecting an analog phone (Siemens Gigaset) to * via a WX100USB, the phone displays "Out of area" first, and then the caller id. The two displays alternate, making the caller-id hard to see. Is there any way I can tell the phone to just display the caller id? Out of area is a flag that gets sent during caller-id transmission, right? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zhone Zplex 10 units
Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). > -Original Message- > From: John Schmerold [mailto:[EMAIL PROTECTED] > Sent: Tuesday, 5 August 2003 12:37 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Zhone Zplex 10 units > > Thanks for the Zplex heads up. > > Steven Critchfield wrote: > > On Mon, 2003-08-04 at 13:13, John Schmerold wrote: > > > >>I frequently see Zhone Zplex 10 units on Ebay - cheap. > >> > >>What's the story on these? > >> > >>Are they flaky? > > > > > > search the archives. > > > > > >>Tough to configure? > > > > > > tough, no, pain in the ***, yes > > > > > >>Any other issues that come to mind? > > > > > > search the archive, that is why it is there. > > > > > >>I don't see them listed on Zhone's website (except in support), so I > >>suspect they've discontinued the product, but if it's a good product I > >>could use it to learn Asterisk. > > > > > > Thats funny since they don't really even act like they want to support > > them. > > -- > John Schmerold > Katy Computer Systems, Inc > 20 Meramec Station Rd > Valley Park MO 63088 > 314-316-9000 v > 775-227-6947 f > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does Wildcard x100p support Caller ID outside the US?
The x100p does get the CID in France. It is now a question of how to break it down. I changed callerid.c line 278 to :- ast_log(LOG_NOTICE, "Got this:- %s\n", cid->rawdata); and the result on August 8 at 10:06 from 0490233081 was:- File callerid.c, Line 278 (callerid_feed): Got this:- 080810060490233081 OK, Now what do I do? -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why are FXO so expensive?
> For smaller systems, you'd have to go a NetJet ISDN BRI card ($150? for two lines) Try BT Speedway BRI ISDN, ~20$ on ebay Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server(fwd)
Hi again. On Mon, 11 Aug 2003, Rainer Jochem wrote: > I've played around a little bit and discovered the following: > > with > services_url: > "http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234&pin=1234"; > > the phone tried to get > "GET /xmlservices/vm/index.php?user=1234?pin=1234&name=..." > > > changing the services_url to: > "http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php&user=1234&pin=1234"; > > the phone requests > "GET /xmlservices/vm/index.php?user=3841&pin=3841&name=..." Whew! These phones seem to do some quite strange stuff... > if I enter this URL in my Browser, it works. But the phone still says > "CMXML Error" Okay, you already told me (in pm) that your phone does this whenever it gets any SoftKey definitions. I guess it's time for Cisco support to solve this mystery... > So I guess there's something in the xml-files the phone doesn't understand. If you want to check more: I've finally put some test scripts on http://swt.uni-stuttgart.de/~langausd/asterisk > Greetings from too sunny Saarbruecken Bah! It's at least that sunny here in Stuttgart. Plus we do not have any significant wind, being encircled by mountains. The City has had max. temperatures of >40°C during the past 4 days. *sweat* Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_mysql uncompress
Hey, Have i done something wrong or is there something wrong with latest CVS and cdr_mysql, cause after checking out latest CVS today, I got warning: [cdr_mysql.so]WARNING[1074424544]: File loader.c, Line 226 (ast_load_resource): /usr/lib/asterisk/modules/cdr_mysql.so: undefined symbol: uncompress WARNING[1074424544]: File loader.c, Line 345 (load_modules): Loading module cdr_mysql.so failed! ...And Asterisk won't start. Help needed! -Johanna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Libpri and asterisk fail to install
Martin Pycko wrote: well should be ok if you cvs update now. Many Thanks ! Martin On Wed, 6 Aug 2003, Rhys Hopkins wrote: Martin Pycko wrote: You're looking for libncurses-dev and in libpri you can remove -Werror from libpri/Makefile or cvs update libpri (it should be fixed) Thanks for that - I installed ncurses-devel and asterisk now builds ok, but libpri still gives the following error: rhys2:/usr/src/libpri # make clean rm -f *.o *.so *.lo rm -f testpri libpri.a libpri.so.1.0 rm -f pritest pridump rhys2:/usr/src/libpri # make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c q931.c: In function `q931_handle_ie': q931.c:1394: warning: comparison between signed and unsigned make: *** [q931.o] Error 1 rhys2:/usr/src/libpri # Regards, Rhys. regards Martin On Wed, 6 Aug 2003, Rhys Hopkins wrote: Hi, I am having trouble building and installing libpri and asterisk on my system. Zaptel seemed to install OK. I am running SuSE 8.2 ( Linux 2.4.20-4GB ) I have made sure I have the prerequisites ( rpm versions shown below ) rhys2:/usr/src/libpri # uname -a Linux rhys2 2.4.20-4GB #1 Fri Jul 11 07:33:18 UTC 2003 i686 unknown unknown GNU/Linux rhys2:/opt/libpri-0.3.2 # rpm -q openssl-devel openssl-devel-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q readline-devel readline-devel-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q readline readline-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q openssl openssl-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q kernel-source kernel-source-2.4.20.SuSE-62 rhys2:/opt/libpri-0.3.2 # rpm -q termcap termcap-2.0.8-674 This is the output from make clean ; make install for libpri: rhys2:/usr/src/libpri # make clean rm -f *.o *.so *.lo rm -f testpri libpri.a libpri.so.1.0 rm -f pritest pridump rhys2:/usr/src/libpri # make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c q931.c: In function `ie2str': q931.c:1210: warning: comparison between signed and unsigned q931.c: In function `msg2str': q931.c:1227: warning: comparison between signed and unsigned q931.c: In function `q931_dumpie': q931.c:1251: warning: comparison between signed and unsigned q931.c: In function `add_ie': q931.c:1334: warning: comparison between signed and unsigned q931.c: In function `q931_handle_ie': q931.c:1394: warning: comparison between signed and unsigned make: *** [q931.o] Error 1 This is the end of the output from make install for asterisk ( full output attached as "asterisk_errors" ) loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/config.h] Error 1 As can be seen from the rpm queries above, I have termcap installed. Any help would be much appreciated. Regards, Rhys Hopkins Systems Administrator Culver Technologies Limited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 + dtmf
Hello Michael My extensio.conf are as follows: I have try it with H323 phone, it works ok all digits detected. Only when call is coming from pstn cause the problem Also, the console output when digit is press is: Invalid extension '1 ' in context...' There is a space after the 1, I believe its a # key. It could possible be the problem? Any idea to fix it? [conference] ; ; conference: Conference Call ; exten => s,1,Ringing exten => s,2,DigitTimeout,10; Set Digit Timeout to 5 seconds exten => s,3,ResponseTimeout,10; Set Response Timeout to 10 seconds exten => s,4,Answer exten => s,5,Background(conf-getconfno) exten => t,6,Goto(s,5) exten => 1234,1,Meetme,1234|ps|9888 - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 06, 2003 4:40 PM Subject: Re: [Asterisk-Users] chan_oh323 + dtmf > Chee Foong wrote: > > Hello all, > > > > I have a cisco AS5300 which is register with a gatekeeper and a Asterisk > > server also register with the gatekeeper. > > > > PSTN >AS5300 >Gatekeeper >Asterisk > > > > I set up a conference room on the Asterisk sever (Room No 1234). > > I try to call from PSTN to AS5300, The AS5300 will call the Asterisk > > server through the gatekeeper. > > I manage to get to the start of the conference where the 'Please key > > in conference number' is played. > > But when I press the room no (1234), Asterisk only get the first digit > > which is 1 and play 'Invalid conference number' right a way. > > What are the contents of your extensions.conf at the point that you are > trying to enter the conference number? > > After the H.323 channel has been answered, the DTMFs are handled > by the application connected to the channel (conferencing here). > > > > > I am using chan_oh323, I am close to get this thing to work (having > > sorted the correct codec), just the dtmf issue now. I am using digium's > > g729. > > > > By the way how many variation of g729 are there. I know g729a, g729b, > > but there seem to be others. > > > > Please help. > > > > Thanks > > > > Foong > > > Michael. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers
> hi .. > > I have an asterisk system with three TDM100P (single port FXO) cards > and 10 Grandstream 100 phones connected to it .. The TDMx00P cards are FXS cards.. :) > > 1st question: > when i phone out > or receive a call from one of the SIP phones onto the PSTN, there is > a LOT of local echo in the handset .. the PSTN end of the call does not > here this echo, but it's VERY annoying on the SIP end of things .. > the echo seems to be about 0.3 seconds delayed to the speech .. > there is no echo on incoming voice, just an echo of my own voice > as I speak. What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L, Chan_Capi > > 2nd question: > using a grandstream phone & asterisk, if I hear another phone ringing, > how can answer it from the phone infront of me? eg. if extension 6003 > is ringing, and i have phone number 6004, how can I answer it ? You need to setup call groups, search through the archives cos I rememeber a thread on this a short while ago.. > > 3rd question: > can someone give me some "starter hints" to configure call parking ? > I haven't managed to find a direct way to transfer a call from phone > to phone except using blind transfer and I want the person initiating > the transfer to speak to the receiving person before actually passing > the call. As far as I know there is no facility to do a consultative transfer on the GS phones.. Only a blind transfer.. Maybe it will come later.. > > can anybody help please ? > > cheers > Dave A Caruana > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi: Hanging channels - again
http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.4b.tar.gz the downloads dir is browseable, but i probably should update the website a bit regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Don, 2003-08-07 um 12.42 schrieb Armand A. Verstappen: > Hi Klaus-Peter, > > On Wed, 2003-08-06 at 12:33, Klaus-Peter Junghanns wrote: > > always use latest chan_capi. the bug is fixed in 0.2.4a. > > today 0.2.4b is online which fixes some issues with sending > > dtmf and a small enhancement to capiECT. > > I checked the site, but can't find the 0.2.4b version. The sidebar menu > offers 0.2.4a, and http://www.junghanns.net/asterisk/page1.html offers > 0.2.2 (under download lates version _here_). > > wkr, > > -- > Envida http://www.envida.net/ > Armand A. Verstappen Graadt van Roggenweg 328 > [EMAIL PROTECTED] 3531 AH Utrecht > tel: +31 (0)30 298 2255Postbus 19127 > fax: +31 (0)30 298 21113501 DC Utrecht ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
See answers in-line.
At 4:14 PM -0400 8/7/03, Wade Weppler wrote:
From: "Wade Weppler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
Reply-To: [EMAIL PROTECTED]
Date: Thu, 7 Aug 2003 16:14:51 -0400
Ah, good idea! I assume even a global variable could be used instead of
using db routines...
Where this doesn't work so well is when trying to implement
least-cost-routing using local calling areas spread over satellite offices.
Here's an example:
Office A has 4 Telco lines.
Office B has 4 Telco lines.
Office A and Office B have 8 station sets each.
Office A and Office B both have Asterisk boxes.
Office A and Office B are long distance calls away from each other, so they
use IAX for interoffice calls, and would also like to utilize VoIP to extend
their local calling area.
If an employee from Office A wants to make a call to someone in Office B's
local calling area, the system will need to follow the following logic:
1) Is there a telco line available in Office B?
2) No? Use a local line in Office A and make a long distance call.
3) Yes? Place the call through a local line in Office B.
4) Worst case, all lines are busy. Let the user know.
Bottom line, the call has to go through without any intervention from the
user, but try the cheapest method first.
We're already written an AGI module to handle the call routing (ie. which
numbers are locally available from each Office), but I'd like to be able to
handle line availability as well.
Why use an AGI? This seems to be easily done with the dialplan,
unless I'm missing some additional sophistication that you're not
mentioning.
Any idea how this could be done?
TRIP (RFC2871 and RFC3219) Not implemented in Asterisk yet -
looking for programmers. See my posts to the -dev list last month.
This is probably overkill for a two office situation, but imagine you
have three hundred offices...
JT
-wade
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, August 07, 2003 3:51 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sip Trunk config
And to answer Wade's question: to limit outbound calls on a
particular path, you'd use a local db set routine. In other words,
every time a call is created to that particular SIP peer, you'd add 1
to the counter, and every time a call was hung up out of that pool,
you'd subtract one.
JT
At 3:30 PM -0400 8/7/03, Patrick wrote:
>
>incominglimit is already implemented for SIP. Just specify under the
>endpoint how many incoming connections are allowed.
>
>For example,
>
>[cisco]
>type=friend
>username=cisco
>secret=blah
>nat=yes; This phone may be natted
>host=dynamic
>canreinvite=no ; Cisco poops on reinvite sometimes
>qualify=200; Qualify peer is no more than 200ms away
>defaultip=192.168.0.4
>incominglimit=20 ; set limit to 20 voice channels
>
>
>setting the limit to 0 (incominglimit=0) is unlimited.
>
>to view the current lines in use --- sip show inuse from the cli.
>
>
>Patrick
>
>
>> I've also run into the "how many lines" problem.
>
>> Possibly something similar to incominglimit= and outgoinglimit= in
> >> h323.conf
> >> could be implemented in sip.conf?
> >
> >> -wade
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of David Hindmarsh
>> Sent: Thursday, August 07, 2003 12:19 AM
>> To: [EMAIL PROTECTED]
>> Subject: Re: [Asterisk-Users] Sip Trunk config
>>
>> Thanks for that,
>>
>> I was looking at the extensions.conf, particularly the line in the
> >> general
>> section which is
>>
>> TRUNK=SIP/???
>>
>> Using this method would be easier.
>>
>> How do you tell asterisk how many lines are available at the gateway
>>
>>
>> Dave
>> - Original Message -
>> From: "Martin Pycko" <[EMAIL PROTECTED]>
>> To: <[EMAIL PROTECTED]>
>> Sent: Thursday, August 07, 2003 12:34 PM
>> Subject: Re: [Asterisk-Users] Sip Trunk config
>>
>>
>> > exten => _9X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
>> >
>> > regards
>> > Martin
>> >
>> > On Thu, 7 Aug 2003, David Hindmarsh wrote:
>> >
>> > > Hi
>> > >
>> > > Is it possible to use a sip gateway as a trunk.
>> > >
>> > > If so, how would I do this
>> > >
>> > > David Hindmarsh
>> > >
>> > > - Original Message -
>> > > From: "Jamie Carl" <[EMAIL PROTECTED]>
>> > > To: <[EMAIL PROTECTED]>
>> > > Sent: Thursday, August 07, 2003 12:14 PM
>> > > Subject: Re: [Asterisk-Users] X-Lite <-> Snom200
>> > >
>> > >
>> > > > Yes, over a LAN. It does it with both g.711 and GSM which
>> > > > both used to work. Havn't had a chance to have a REAL
>> > > > good look into it though.
>> > > >
>> > > > J
>> > > >
>> > > > O
[Asterisk-Users] SIP Lines
Instead of using a PCI card is it possible to use an outside SIP service for “CO” lines?
Re: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
It just rings the fewestcalls or leastrecent over and over.. it doesn't hunt one bit right now. Thats why I posted to the list so Mark could get an idea of what people would like to see before he fixes fewestcalls and leastrecent logic. bkw On Mon, 11 Aug 2003, Jim Friedeck wrote: > In our original spec for Digium, leastrecent was specifically 'agent who > answered a call longest ago for that queue'. (not a direct quote) It > would then go to the next agent in order of 'longest go'. Has this > changed? Does it immediatly go roundrobin by agent number or agent load > order? Thanks. > > Jim Friedeck > > -- > > Brian West wrote: > > >Ok just had my boss point something out: > > > >"I'd think dumping calls on most-idle would be fairly straightforward, but > >could be skewed if agentA is on a 40 minute call, agentB has a bunch of 5 > >minute calls" > > > >So total call time should be counted in the logic somewhere. > > > >bkw > > > >On Sun, 10 Aug 2003, Brian West wrote: > > > > > > > >>I think we are starting to see what type of logic people are wanting in > >>fewestcalls and leastrecent strategy. > >> > >>bkw > >> > >>On Sun, 10 Aug 2003, Richard Lyman wrote: > >> > >> > >> > >>>i disagree, instead of thinking 'fallback' how about 'order' the agents > >>>(by effecting the 'metric') so you 'target' the agent you want first > >>>then if fail they go right to the next one in the 'ordered' list. > >>> > >>>Brian West wrote: > >>> > >>> > >>> > leastrecent suffers the same fait as fewestcalls onlying ringing the > leastrecent agent over and over endlessly. It should have a fallback > option. > > roundrobin with leastrecent first > roundrobin with fewestcalls first > > I would like to see a roundrobin with leastbusy first option. > (just because you have taken less call or leastrecent doesn't mean you > haven't been a busy agent!) > > I'm sure better autologoff logic as per my first email would be a great > idea also. > > bkw > > On Sun, 10 Aug 2003, Richard Lyman wrote: > > > > > > >well if you ask me, the leastrecent part would work if you reversed the > >logic on the metric. > > > >my other last_used mod would do a time_t on that agent the last time it > >was 'tried' (ast_request'd) then (i was using arrays) qsort so that (new > >agents) '0' would be on top, and the agent that got the most recent > >attempt would be on the bottom '1057174447' (below is an example) > > > > -- sorted agent array: 317 last_used: 0 > > -- sorted agent array: 318 last_used: 0 > > -- sorted agent array: 319 last_used: 0 > > -- sorted agent array: 300 last_used: 1057174447 > > > >that way, (for leastrecent anyway), you are always working with a full stack of > >agents. > > > > > > > >Brian West wrote: > > > > > > > > > > > >>First of all I would like to thank Mark for getting roundrobin to go > >>roundrobin. Good job. > >> > >>Now we have some options here for leastrecent and fewestcalls strategy. It > >>needs some work on the logic and Mark recommend that I ask the list and > >>get some input before he makes any changes to it. > >> > >>fewestcalls from what I have seen would always ring the agent with the > >>fewestcalls first then go into roundrobin if that agent didn't answer. > >> > >>Next new caller would ring fewestcalls agent first then start roundrobin. > >> > >>What do you think should happen in fewestcalls? Right now it just rings > >>the agent with the fewestcalls over and over with current app_queue logic. > >> > >>leastrecent from what I have been looking at will ring the agent that has > >>least recently take a call first then if they don't answer go into > >>roundrobin. Then the next new call coming from queue would first go to > >>the leastrecent first then try every agent in roundrobin till answered > >>then starting over again. New caller from queue hits leastrecent agent > >>first. > >> > >>Same thing happens in leastrecent strategy. The leastrecent agent will > >>ring over and over with current app_queue logic. > >> > >>Now some of you might recommend autologoff options. But that also might > >>need some work. I don't want to log off an agent for not answering the > >>phone only once. So here is how I would like to see autologoff work. > >> > >>Example: > >>queue timeout = 20 > >>agent autologoff = 60 > >> > >>The agent would have to not answer their phone 3 times in a row to get > >>logged off. As it stands now they did not answer just once and get logged > >>off. Thus allow for an employee to use the excuse for not working when > >>they should be logged in and taking calls. > >> > >>Unless i'm wrong here. > >
RE: [Asterisk-Users] queue / agent documentation
Title: RE: [Asterisk-Users] queue / agent documentation
My configuration is with a X100P (incoming) and TDM400P w/ 2 ports (agents) and the calls will distribue just as perscribed with ringall and leastrecent. Those are the only two I have used thus far. CVS was a check out from last night.
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]]
Sent: Friday, August 08, 2003 03:16
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] queue / agent documentation
> And you should take a look at queues.conf for some comments detailing the
> various queue distribution algorithms, ringall, roundrobin, leastrecent so
> on so forth.
I wanna see if anyone else has seen this result?
All except of which do not work. The only method I can get working is
ringall.
queues.conf:
[techsupport]
music = default
strategy = roundrobin
context = demo
timeout = 15
retry = 15
maxlen = 0
extensions.conf:
[agentlogin]
exten => 800,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM})
exten => 800,2,Playback(agent-loginok)
exten => 800,3,Hangup
exten => 801,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM})
exten => 801,2,Playback(agent-loggedoff)
exten => 801,3,Hangup
[callqueue]
; sales
exten => 900,1,Answer
exten => 900,2,Queue(techsupport|TtH)
exten => 900,3,WaitMusicOnHold(20)
exten => 900,4,Voicemail2(u900)
exten => 900,5,Playback(vm-goodbye)
exten => 900,6,Hangup
Here is what I see when doing a roundrobin with two agents added via
AddQueueMemeber: It never sends a call to agent 1234 at all... and will
only ring 1236 over and over even during a call. I setup iaxclient to
load the queue with 4 calls. Then setup two xten sip phones for the
agents. During my testing even other routing methods deliver similar
results. Bug report is open.
http://bugs.digium.com/bug_view_page.php?bug_id=045
asterisk*CLI> sip show peers
Name/username Host Mask Port Status
1236/1236 65.38.28.149 (D) 255.255.255.255 5060 Unmonitored
1235/1235 (Unspecified) (D) 255.255.255.255 0 Unmonitored
1234/1234 65.38.28.150 (D) 255.255.255.255 5060 Unmonitored
asterisk*CLI> show queues
techsupport has 4 calls (max unlimited) in 'roundrobin' strategy
Members:
SIP/1234 has taken no calls yet
SIP/1236 has taken no calls yet
Callers:
1. [EMAIL PROTECTED]:5036]/24 (wait: 0:18)
2. [EMAIL PROTECTED]:5036]/25 (wait: 0:17)
3. [EMAIL PROTECTED]:5036]/26 (wait: 0:16)
4. [EMAIL PROTECTED]:5036]/27 (wait: 0:14)
default has 0 calls (max unlimited) in 'ringall' strategy
No Members
No Callers
-- Called 1236
-- SIP/1236-7fc3 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-df37 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-c56b is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-e9a7 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-b4dd is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-029d is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-f0aa is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-c264 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-bd12 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-197e is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-2056 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-fa07 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-994b is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-220d is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-997a is ringing
-- SIP/1236-997a answered [EMAIL PROTECTED]:5036]/24
-- Stopped music on hold on [EMAIL PROTECTED]:5036]/24
-- Called 1236
-- SIP/1236-e4ab is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-9ed1 is ringing
-- Nobody picked up in 15000 ms
-- Called 1236
-- SIP/1236-b26c is ringing
-- Nobody picked up in 15000 ms
== Spawn extension (demo, 900, 2) exited non-zero on
'[EMAIL PROTECTED]:5036]/24'
-- Hungup '[EMAIL PROTECTED]:5036]/24'
-- Called 1236
-- SIP/1236-bfd6 is ringing
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Libpri and asterisk fail to install
Martin Pycko wrote: You're looking for libncurses-dev and in libpri you can remove -Werror from libpri/Makefile or cvs update libpri (it should be fixed) Thanks for that - I installed ncurses-devel and asterisk now builds ok, but libpri still gives the following error: rhys2:/usr/src/libpri # make clean rm -f *.o *.so *.lo rm -f testpri libpri.a libpri.so.1.0 rm -f pritest pridump rhys2:/usr/src/libpri # make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c q931.c: In function `q931_handle_ie': q931.c:1394: warning: comparison between signed and unsigned make: *** [q931.o] Error 1 rhys2:/usr/src/libpri # Regards, Rhys. regards Martin On Wed, 6 Aug 2003, Rhys Hopkins wrote: Hi, I am having trouble building and installing libpri and asterisk on my system. Zaptel seemed to install OK. I am running SuSE 8.2 ( Linux 2.4.20-4GB ) I have made sure I have the prerequisites ( rpm versions shown below ) rhys2:/usr/src/libpri # uname -a Linux rhys2 2.4.20-4GB #1 Fri Jul 11 07:33:18 UTC 2003 i686 unknown unknown GNU/Linux rhys2:/opt/libpri-0.3.2 # rpm -q openssl-devel openssl-devel-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q readline-devel readline-devel-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q readline readline-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q openssl openssl-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q kernel-source kernel-source-2.4.20.SuSE-62 rhys2:/opt/libpri-0.3.2 # rpm -q termcap termcap-2.0.8-674 This is the output from make clean ; make install for libpri: rhys2:/usr/src/libpri # make clean rm -f *.o *.so *.lo rm -f testpri libpri.a libpri.so.1.0 rm -f pritest pridump rhys2:/usr/src/libpri # make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c q931.c: In function `ie2str': q931.c:1210: warning: comparison between signed and unsigned q931.c: In function `msg2str': q931.c:1227: warning: comparison between signed and unsigned q931.c: In function `q931_dumpie': q931.c:1251: warning: comparison between signed and unsigned q931.c: In function `add_ie': q931.c:1334: warning: comparison between signed and unsigned q931.c: In function `q931_handle_ie': q931.c:1394: warning: comparison between signed and unsigned make: *** [q931.o] Error 1 This is the end of the output from make install for asterisk ( full output attached as "asterisk_errors" ) loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/config.h] Error 1 As can be seen from the rpm queries above, I have termcap installed. Any help would be much appreciated. Regards, Rhys Hopkins Systems Administrator Culver Technologies Limited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk
Sip Rtp wrote: Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. More information is needed. You should provide a backtrace of the core file, the screen log of Asterisk (generated when executed with "asterisk -vvvcdg"), your oh323.conf and the important sections of extensions.conf. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in the * ). But now problem comes in the outbound as when i use a extension like exten=>12,1,Dial(OH323/12) Then the call goes through but i don't hear any voice. So my two problems are 1.Why asterisk gives seg. fault when i dial exten on 711 codec from ATA 2.Why can't i hear voice from * to ATA when i use 723 in ATA. for 2nd i think that there is mismatch between the codecs so can we change the priority order of the codecs used in the * or Oh323 and if yes, then how? Please ask if any further Input is required. Rgds Manoj K Gupta Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
> Why use an AGI? This seems to be easily done with the dialplan, > unless I'm missing some additional sophistication that you're not > mentioning. Our local area (Toronto) has some extreme overlapping areacode problems that require some logic to decipher. I've been able to pull exchange data into a MySQL database (NPA/NXX codes) to help with the logic, and it's worked quite well. For instance, I'm currently in areacode 905, but not all 905 areacodes are a local call for me. Our satellite office is in areacode 416, where additional 905 areacodes are a local call that we can then use for our other office. The AGI works quite well. I may even post it somewhere if anyone wants it. Here's the basic logic: 1) Offices are assigned an "exchange" number that includes all of the NPA/NXX codes that are local to that exchange (Toronto has over 1300 combinations). NPA, NXX, and exchange are all stored in a MySQL database. 2) The AGI script is passed the local exchange number and the dialed LD number. If the dialed number appears in any other exchange in our database, the call is passed through the proper office. If it doesn't appear, the LD call is made over the local telco. Exchanges are modified constantly, so keeping up with NPA/NXX changes can be a fulltime job! > TRIP (RFC2871 and RFC3219) Not implemented in Asterisk yet - > looking for programmers. See my posts to the -dev list last month. I'll have to do some digging on this. I'm assuming this exposes a lot more information at each office (ie. line availability) than anything we can do in Asterisk today? -wade ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Libpri and asterisk fail to install
Hi, I am having trouble building and installing libpri and asterisk on my system. Zaptel seemed to install OK. I am running SuSE 8.2 ( Linux 2.4.20-4GB ) I have made sure I have the prerequisites ( rpm versions shown below ) rhys2:/usr/src/libpri # uname -a Linux rhys2 2.4.20-4GB #1 Fri Jul 11 07:33:18 UTC 2003 i686 unknown unknown GNU/Linux rhys2:/opt/libpri-0.3.2 # rpm -q openssl-devel openssl-devel-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q readline-devel readline-devel-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q readline readline-4.3-105 rhys2:/opt/libpri-0.3.2 # rpm -q openssl openssl-0.9.6i-12 rhys2:/opt/libpri-0.3.2 # rpm -q kernel-source kernel-source-2.4.20.SuSE-62 rhys2:/opt/libpri-0.3.2 # rpm -q termcap termcap-2.0.8-674 This is the output from make clean ; make install for libpri: rhys2:/usr/src/libpri # make clean rm -f *.o *.so *.lo rm -f testpri libpri.a libpri.so.1.0 rm -f pritest pridump rhys2:/usr/src/libpri # make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c q931.c: In function `ie2str': q931.c:1210: warning: comparison between signed and unsigned q931.c: In function `msg2str': q931.c:1227: warning: comparison between signed and unsigned q931.c: In function `q931_dumpie': q931.c:1251: warning: comparison between signed and unsigned q931.c: In function `add_ie': q931.c:1334: warning: comparison between signed and unsigned q931.c: In function `q931_handle_ie': q931.c:1394: warning: comparison between signed and unsigned make: *** [q931.o] Error 1 This is the end of the output from make install for asterisk ( full output attached as "asterisk_errors" ) loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/config.h] Error 1 As can be seen from the rpm queries above, I have termcap installed. Any help would be much appreciated. Regards, Rhys Hopkins Systems Administrator Culver Technologies Limited. rhys2:/usr/src/asterisk # make clean for x in res channels pbx apps codecs formats agi cdr astman; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' rm -f *.so *.o .depend rm -f busy.h ringtone.h gentone gentone-ulaw make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/pbx' make[1]: Entering directory `/usr/src/asterisk/apps' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/apps' make[1]: Entering directory `/usr/src/asterisk/codecs' rm -f *.so *.o .depend ! [ -d g723.1 ] || make -C g723.1 clean ! [ -d g723.1b ] || make -C g723.1b clean make -C gsm clean make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' rm -f */*.o\ ./tst/lin2cod ./tst/lin2txt \ ./tst/cod2lin ./tst/cod2txt \ ./tst/gsm2cod \ ./tst/*.*.* find . \( -name core -o -name foo \) \ -print | xargs rm -f rm -f ./lib/libgsm.a ./add-test/add \ ./bin/toast ./bin/tcat ./bin/untoast\ ./gsm-1.0.tar.Z make[2]: Leaving directory `/usr/src/asterisk/codecs/gsm' make -C mp3 clean make[2]: Entering directory `/usr/src/asterisk/codecs/mp3' rm -f src/cdct.o src/cupl3.o src/hwin.o src/iup.o src/l3init.o src/msis.o src/wavep.o src/csbt.o src/cwinm.o src/icdct.o src/mdct.o src/uph.o src/cup.o src/dec8.o src/isbt.o src/l3dq.o src/mhead.o src/upsf.o src/iwinm.o rm -f libmp3.a make[2]: Leaving directory `/usr/src/asterisk/codecs/mp3' make -C lpc10 clean make[2]: Entering directory `/usr/src/asterisk/codecs/lpc10' rm -f *.o ./liblpc10.a make[2]: Leaving directory `/usr/src/asterisk/codecs/lpc10' make -C ilbc clean make[2]: Entering directory `/usr/src/asterisk/codecs/ilbc' rm -f lib
Re: [Asterisk-Users] X100P delivery
for me it takes between 2 and 3 weeksstandard delivery. BR, Dan - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 07, 2003 6:23 PM Subject: [Asterisk-Users] X100P delivery > > > I live in Japan and last Sunday I bought my first X100P to see > if it really works for my H323 application. > > How long time it should take to be delivered? > > > Isamar > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
> First of all I would like to thank Mark for getting roundrobin to go > roundrobin. Good job. Exactly, the whole queue system seems significantly better than it was not so long ago. Thank very much! > Now we have some options here for leastrecent and fewestcalls strategy. It > needs some work on the logic and Mark recommend that I ask the list and > get some input before he makes any changes to it. IMHO, leastrecent and fewestcalls should order the 'queue' based on those parameters. The members of the queue should be called in that order. (Of course, the priority parameter should be respected first though). Now, roundrobin should work by always ordering the members in the same order, but the next interface it tries is always one after the previous one it tried. We should also have a type of 'ordered' or similar, which is like roundrobin but always starts from the same spot. (eg, always call the receptionist first, then overflow to the accounts person, and then the marketing person etc... > Now some of you might recommend autologoff options. But that also might > need some work. I don't want to log off an agent for not answering the > phone only once. So here is how I would like to see autologoff work. > > Example: > queue timeout = 20 > agent autologoff = 60 > > The agent would have to not answer their phone 3 times in a row to get > logged off. As it stands now they did not answer just once and get logged > off. Thus allow for an employee to use the excuse for not working when > they should be logged in and taking calls. > > Unless i'm wrong here. This sounds very nice, but perhaps there should also be some sort of indicator to the user that they have been logged off. If you forget to logoff before you go to the toilet, you come back, and start doing 'stuff' while you wait for some calls to arrive and don't realise you have been logged off. Some options: a) Send them an email when they are auto-logged off (a lot of 'agents' may not have an email account) b) Send them a voicemail when they are auto-logged off (might have MWI which provides a visual indicator that you have been logged off) c) Don't actually log them off, just ignore their extension for a configurable amount of time. This might even re-act similar to qmail's logarithmic back-off retry times. ie, suspend for 10 minutes, then retry, suspend for 30 minutes, retry, suspend for 1 hour, etc... (need a way to log back on tho) d) Should also log to cdr when someone is auto-logged off, this way reporting will show that agent 232 is lazy and never answers the phone, resulting in always being logged off Other people with more experience in a call centre might have other ideas on notification methods... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and Caller ID (again and again...)
Hi Martin, Together with another list member we try to find a solution now. We'll keep you in touch if something will be solved. Thanks, Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 06, 2003 5:54 PM Subject: Re: [Asterisk-Users] X100P and Caller ID (again and again...) > asterisk/callerid.c > > On Wed, 6 Aug 2003, Dan wrote: > > > Hi, > > > > I have the X100P card and the CallerID service activated on my PSTN line. > > When I receive a call, the CallerID presented through X100P card represent > > my own PSTN number. > > I have the following lines in zapata.conf. > > > > usecallerid=yes > > callerid=asreceived > > > > It seems that it can pick-up the caller id, but not the right part of it. > > My own PSTN number is included in the CallerID information too. > > > > Where I can modify the part of the received information which is considered > > as the CallerID (number) from all the received data? > > > > Thanks, > > Dan > > P.S. I am located in Romania and the PSTN is Siemens based. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New SIP Phone
> It doesn't make much sense to me, but it appears Robertson intends to > make money just selling pre-configured phone hardware. The sample > units from Grandstream were $60 a while ago, and $75 MSRP. Doesn't > seem like much markup, so I'm curious to see how this plays out. I would assume they are jumping on the bandwagon that Vonage has created in using the Cisco ATA devises to broadband internet users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stable versions of Asterisk (Was: Re: Fair comparison (JohnTodd))
Hi, It's really a problem for new Asterisk users. I am new to Asterisk and do not know * history, which applications are stable, which are in development, and who do what? It's really hard for new users to keep the pace with CVS. So can you recommend more stable Asterisk versions, which are suitable for production environments? My needs is simple: standard switch (Call transfer, Parking...), Voicemail, Voice menu (Autoattendand). If someone have running Asrerisk on production system, please share the yours version info. best regards, Nguyen At 03:49 PM 8/12/2003 -0500, you wrote: As far as Asterisk's stability goes: new features tend to be less stable than older features, just like any software. If your user base isn't requesting all the bells and whistles, then adequate testing will normally reveal problem spots before you stumble across them in production. Despite what some others on the list may claim, running the absolute latest CVS on production systems without testing is probably unwise. :) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ip phones and intercom/paging
I'm a bit interested in an intercom system as well. I'm using asterisk with analog phones. Is there any way I can do this? AJ On Fri, 8 Aug 2003, cwitte wrote: > There was a thread a few months ago that tossed around some ideas for > using a cisco phone for intercom or paging. I don't have any ip > phones, and wondered if anyone had any luck getting intercom or paging > to work on the cisco units. > > Do any of the (cheaper) ip phones have a way to support intercom or > paging? > I presume that it's not part of the SIP or IAX protocols. > > Chris. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
