I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk - actually, I still get an error message, but it doesn't appear to be fatal. Suggestions? Art O'Dea
On Fri, 2003-02-28 at 19:51, Art O'Dea wrote:> I have an ATA-186 in a SIP configuration (following Shawn Djernes > how-to), but I get the following error at the asterisk console when I > try to call the phone connected to the ATA: > > ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device > Failed to register zone 'United States / North America': No data > available > > Everything works if I remove indications.conf from /etc/asterisk - > actually, I still get an error message, but it doesn't appear to be > fatal.Quick guess, since ZT_* functions have to do with zaptel or zapata, make sure you compiled and installed them, even if you aren't using zapata hardware. -- Steven Critchfield <critch at basesys.com>
Actually, this is the exact same error I get. However, it appears when I pick up an extension connected to a zhone zplex. It is then followed by an error that says it cannot play the dial tone (but I can still dial extensions). Two other problems I'm having are: 1. When calling from one extension to another on the zhone, when I pick up the receiving extension, the 'ringing' sound heard by the calling party can be heard by both parties (over the top of the conversation) 2. When calling from messenger to an extension on the zhone, everything works great. When calling from an extension on the zhone, messenger acknowledges the call and when I click accept, messenger believes that it is in a call, but the calling extension acts like no-one has picked up the other end. Any thoughts would be very much appreciated. Cheers, Kent> -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 9:57 PM > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] Newbie question > > On Fri, 2003-02-28 at 19:51, Art O'Dea wrote: > > I have an ATA-186 in a SIP configuration (following Shawn Djernes > > how-to), but I get the following error at the asterisk console whenI> > try to call the phone connected to the ATA: > > > > ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device > > Failed to register zone 'United States / North America': No data > > available > > > > Everything works if I remove indications.conf from /etc/asterisk - > > actually, I still get an error message, but it doesn't appear to be > > fatal. > > Quick guess, since ZT_* functions have to do with zaptel or zapata,make> sure you compiled and installed them, even if you aren't using zapata > hardware. > > -- > Steven Critchfield <critch at basesys.com> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
I have zaptel and zapata compiled and installed for the X100P in my system (the * box has the x100p, and the ATA-186 and * are on a class-c network, all behind a firewall. Calling in to voicemail works fine, but ringing the ATA gives me the ZT_LOADZONE error, the line continues to ring when the ATA phone picks up. Following your suggestion, I'd guess it would have something to do with zaptel since ZT_LOADZONE is referenced in zaptel.c - but could it be something in the ATA-186 configuration? Art> On Fri, 2003-02-28 at 19:51, Art O'Dea wrote: > > I have an ATA-186 in a SIP configuration (following Shawn Djernes > > how-to), but I get the following error at the asterisk console when I > > try to call the phone connected to the ATA: > > > > ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device > > Failed to register zone 'United States / North America': No data > >available > > > > Everything works if I remove indications.conf from /etc/asterisk - > > actually, I still get an error message, but it doesn't appear to be > > fatal. > > Quick guess, since ZT_* functions have to do with zaptel or zapata, > make > sure you compiled and installed them, even if you aren't using zapata > hardware. > > -- > Steven Critchfield < critch at basesys.com >
Hi! I've installed Asterisk and connected ATA-186. When I press 8500, I listen "voice main menu" and prompt for enter mailbox number. I press "1234", but asterisk not accept number and switch to "demo-instruct". Also Asterisk write warning: NOTICE[77839]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible. Where am I wrong? -- Sincerely yours, Andrey Katkov.
>RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389 >packets. You can turn this off through the ATA 186 web interface. > >It looks as though you need to configure that ATA186 properly - several >people have posted guides on this. > > IainI've used guide from http://www.loligo.com/asterisk/Cisco/ATA-186- guide.v20030628.txt I did all step by step. And one time it worked. When I reload Asterisk it don't work again (and without message about RFC). How can I trace what happen? -- Sincerely yours, Andrey Katkov.
> Message: 1 > From: "Dan" <dtoma@fx.ro> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Newbie question > Date: Thu, 3 Jul 2003 13:14:44 +0300 > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I have the same error only whan I start Asterisk with asterisk -vvvvvc > If I start it with safe_asterisk and then enter to the console with > asterisk -vvvvvvr this message does not appear. > Any type of calls I made from the SIP phone I get the same NOTICE type > message. > Anyway, it seems to work except that I must enter the DTMF codes several > times in order to be correctly interpreted. > The used SIP phone is a Cisco 7960 >I've found the problem. In example said: Step 4: Set up our sip.conf file for Asterisk A typical configuration for an ATA-186 would look like this: [2299] type=friend username=2299 secret=lordwhorfin canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=2299 nat=1 But, demo voicemail doesn't accept dtmf dialing. I've changed string "dtmfmode" to "inband" and demo start work ... -- Sincerely yours, Andrey Katkov.
On Fri, 29 Aug 2003, Timothy Soos wrote:> -----BEGIN PGP SIGNED MESSAGE----- > I have a very newbie question: I recently bought an Asterisk Developer's > Kit (TDM). Which of the following can I use with an TDM400P card and expect > it to work properly: > - - A $5 analog telephone I got from Wal Mart into the TDM400P card and expect > it to work properly, or can I only use an analog phone specificaly designed > for use in a PBXa) please don't post PGP onto a mailing list b) the standard $5 phone should work c) specific phones for PBX MAY or MAY NOT work, depends on how specific they are to a specific PBX :) welome to * -wasim
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello All, (apologies to all for the previous version of this message sent in error) I have a very newbie question: I recently bought an Asterisk Developer's Kit (TDM). Which of the following can I use with an TDM400P card and expect it to work properly?: - - A $5 analog telephone I got from Wal Mart - - An analog phone specifically designed for use in a PBX - -- Thanks, Timothy Soos XQL, LLC -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/T2UnnG4mv6BHya0RAuPqAJ4uMVqd+BeR2S9xL7Uug2tIiWwFHQCfVq3n T3dwCPXgWJceU9LWgcfvzRw=s0oZ -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have the following questions: 1.) Can Asterisk interface with the OpenSwitch12 board? I've read some postings that say yes and some that said no. 2.) If I use a soft phone do I still need to purchase a board like the open switch or TDM400P to handle the internal extensions? General details of our PBX upgrade: Currently we have a comdial PBX system and we are going to be branching our company into two locations. I have found that it will be cheaper to setup two * systems then to upgrade our current system. (Currently we have 32 internal extension system and 16 incoming lines.) I already know that I am going to use the Wildcard T100P with a PRI from the phone company. My only concern are how I handle the extensions at the users desks. Thanks for the help Chris Mader -----BEGIN PGP SIGNATURE----- Version: PGP 8.0 iQA/AwUBP4RMl/v5JE8nnIbeEQLd6gCeJyC7riwMx5gHfrflCO1be3wbzEYAnRi7 rbIEcxkkVSh+Sm7VG+xPEMhd =l/E5 -----END PGP SIGNATURE-----
Hey All, We are using Asterisks as a voicemail only application, and so far all is great. (Excellent product!) However, I do have one question that I am hoping you might be able to help me with. In our asterisk application. When our customers call *55 (our dialplan code to check voicemail) then they are sent directly to voicemail (asterisk). Asterisk then gives a voice prompt asking the customer to enter their extension number (entire 10 digit telephone number in our case). My question is. Is there a way to make asterisk aware of the calling-from (callerID) number so that it will automatically detect the number and then go directly to asking them to input their password. If so, where would I make the config changes for this in the asterisk config files, and does anyone have an example of a similar config? Thanks! Darren Nay VOIP Network Developer Ionosphere, Inc dnay@ionosphere.net <mailto:dnay@ionosphere.net> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/80a275f9/attachment.htm
If you change the extension to the follow
exten => *55,1,VoiceMailMain(${CALLERIDNUM})
The voicemail will now user their caller id for the mailbox
>
On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:> My question is.? Is there a way to make asterisk aware of the > calling-from (callerID) number so that it will automatically detect > the number and then go directly to asking them to input their > password. > > ?From "show application VoicemailMain" try: exten => 1001,1,VoiceMailMain(${CALLERIDNUM}) We use: exten => 1001,1,VoiceMailMain(s${CALLERIDNUM}) to skip password Change extensions accordingly. Jeb
Good Day List, I am interested in the Asterisk product and have seamed to be able to answer most of my questions on the wiki board. However, there is a single question that continues to haunt me and so here i am asking the Users list. Does Asterisk support users logging into asterisk from the phone? I know that it supports ACD Group agents logging in to present themselves as available for a call, but my question is in regards to standard users. Lets say i have 4 phones, one in each office. I have 8 employees who share those offices If Sam is set up as extension number 1000 and Bob is Extension 1001 and Sally is 1002 etc...... If today Bob wants to use the phone in office A can he pick up the phone in that office and login so asterisk knows that Bob (extension 1001) should be routed to the phone in office A. Then tomorrow Bob works in office B) Can he again pickup the phone in office B and login such that the Asterisk server routes all calls to BOB (Extension1001) to the phone in office B. Etc........ Finally if the answer to the above is a resounding "Yes you greenhorn it most obviously does :-P" Can I have multiple people log into the same phone simultaneously for example lets say Bob (extension 1001), Sally (extension 1002) and Dave (extension 1000) are all working in the conference room. Can they each log into the phone such that the Asterisk server sends all calls destined to extension 1000, 1001, 1002 to the phone in the conference room? Lastly, Again if the Answer is a resounding "Yes stupid you should RTFM (read the F!@# Manual) then could someone please assist me by providing a link to this functionality. Thanks for your time and patience in this matter. ~peaceoutman _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
Hello: First, I'm really new to asterisk and I'm testing it in order to improve my first steps... Recently I installed * asterisk on a FreeBSD Box (5.2.1) I got it working on my internal LAN (it works fine !). I was trying to connect my * box through FWD using SIP but it is not working an I'm very confused about *, in fact I can't call from my * client (X-Lite) to a FWD number, but bettwen my * sip authenticated clients yes... Please somebody can help me or guide me to the right direction ? Any kind of help will be appreciated and excuseme by my english :o( Here is attached my config files -------------- next part -------------- A non-text attachment was scrubbed... Name: extensions.conf Type: application/octet-stream Size: 315 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041112/967a592e/extensions.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.conf Type: application/octet-stream Size: 395 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041112/967a592e/sip.obj
> First, I''m really new to asterisk and I''m testing it in order > to improve my first steps... > > Recently I installed * asterisk on a FreeBSD Box (5.2.1) > I got it working on my internal LAN (it works fine !). > > I was trying to connect my * box through FWD using SIP > but it is not working an I''m very confused about *, in fact > I can''t call from my * client (X-Lite) to a FWD number, > but bettwen my * sip authenticated clients yes... > > Please somebody can help me or guide me to the right direction ? > > Any kind of help will be appreciated and excuseme by my english :o( > > Here is attached my config filesThere really isn''t enough info in those config files to answer your question. If the bsd box is behind a nat box, then your missing the sip.conf statements to support that. If you are, consider using iax2 for the link (instructions are on fwd''s site). Your exten => _7., statement doesn''t look right either. Here''s what works on my system (substituting your userid/secret and using iax): exten => _7.,1,Dial(IAX2/500460:cuco99@iax2.fwdnet.net/${EXTEN:1},60,r) Modify as necessary for sip. Also, in your register statement: register => 500460:cuco99@fwd.pulver.com/500460 the /500460 at the end tells fwd to send that extn number when they dial your * box. So you will need something in your extensions.conf file that looks something like: exten => 500460,1,Dial(what ever your want to do) Without that, incoming calls from fwd have no where to go. An alternate way to accomplish that is to drop the /500460 from that register statement, and then have an inbound-fwd context something like: [fwd] exten => s,1,Dial(what ever you want to do) To help troubleshoot your config, break the process down into diagnosing outbound calling first followed by diagnosing inbound calls. Use the CLI ''sip debug'' to identify whether your register statement is actually working. Once that is successful, then do the same for an outbound call to one of the fwd test numbers. Once that is successful, then use the fwd web site option to initiate a test call from their site to your * system. Remember, the only thing the register statement does is to tell fwd where to reach your machine (IP address). Your Dial statement is used to send calls to fwd, but you still need an incoming ''context'' for inbound fwd sip calls. Might take a look in the wiki and fwd''s web site for * config examples. http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Hi Mike -> its my first time to post here, im in the process of building asterisk > based telephone system (just small). i already installed asterisk > server, i just wanted to test 2 sip softphones to get working before i > move on, is it possible to have 2 softphones talk to each other > without > any cards?? (i.e. digium, just for testing purposes) > im using kphone, and i follow the instruction on how to configure it > for > asterisk, but there is a problem when i register the kphone, is there > any configuration must follow before i make a calls using asterisk?? > pls > help.Yes, asterisk can route calls without Digium hardware. Many people using asterisk do this (connect via TCP/IP to VoIP providers for outside lines). If you do this, you will need to set up ztdummy for proper timing. See this on the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy A good source for a setup very similar to what you describe is an old article from O'Reilly. It tells you exactly what to do to set up asterisk with just two sip soft phones: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Thanks, Noah
Hello! When the oprator transfers calls to internal extensions to unavailable or busy extensions, how can I prevent these calls from going to voicemail, and route them back to the oprator? But other calls, ie internal between extensions, and calls coming in via DID should get voicemail if extensions are busy / unavailable? Any help be appreciated. TIA! Tim
Hi all, Just a quick question from someone who is reasonably new to the Asterisk server. We have ordered the hardware for a test environment, and plan on setting it up at the start of next week. At the moment, we have a couple of VOIP handsets, a Digium TE110P card, an E1 line and the basic network infrastructure to make it all happen. Can anyone tell me from experience how long it might take to get it up and running so that we can make some basic test calls ? We have an experienced unix / linux admin and a person who deals with traditional hardware PBX / PABX solutions here. I know that this kind of thing is often relative to experience and hardware, but can anyone tell me from experience how long it took them to get it up and running ? Thanks, Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050303/0e8c2d95/attachment.htm
As someone who started out using * when I was just slightly educated on Linux, I can say that you are probably many steps ahead of the average new * user. You have people that know Linux so that is a major plus. The actual configuration of the services you reference is very easy and you could probably have a full system installed in a day. What takes time is doing your dial plan and making sure you have your confs right. That can take a day or two at least for someone who gets it. It took me about 2 weeks of tinkering and Wiki reading to get all the concepts down but I had only been a tinkerer with Linux at that point. If you follow the Getting Started With Asterisk guide from the link at digium.com, you can have most of the setup done lickity split. Then do the install of the sample confs and you will see how it is all stitched together. Alternatively, you can look for Asterisk@home which has a lot of tools integrated for a real turnkey Asterisk install but you will miss the benefit of knowing the innards. Best of luck! I think you will do well with the resources you have. Cheers, Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Callum McGillivray Sent: Thu 3/3/2005 8:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newbie Question Thanks for the quick reply Kevin (and Dean!)... It's the kind of answer I was hoping for. The E1 has been used on a Hardware based PBX until a week or so ago, so I can't see there being an issue there. We are looking at replacing our existing PBX with an Asterisk Machine and the E1 live for another month before it gets switched off (we just switched providers and now have an additional E1 kicking around doing nothing..) Another question... and I know that it's all relative to experience, complexity etc, but how long should I take into account to get some basic voicemail, messages on hold, and basic IVR type stuff up and running? We have several programmers / linux admins in house and they are pretty good at what they do, and have been canvassing the lists to work out the details... but we lack experience at the moment. We have also been looking at various GUI's for Asterisk... (Asterisk@home being one)... can anyone recommend one that would be ideal for a business user in a basic small / medium office environment? Thanks again, Callum -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, 4 March 2005 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question Callum McGillivray wrote:> Can anyone tell me from experience how long it might take to get it up > and running so that we can make some basic test calls ?If the hardware is functional and the E1 is provisioned properly, a decent admin should be able to have Asterisk running and making test calls in a couple of hours. That's a long way from 'done', though :-) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050304/a071ba03/attachment.htm
We learnt as we went, and to be honest it went pretty well. One of our contractor took the job on in order to learn her way around asterisk and add it to her list of skills. Later, PaulH Melbourne Australia -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Callum McGillivray Sent: Friday, 4 March 2005 2:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newbie Question Thanks for the quick reply Kevin (and Dean!)... It's the kind of answer I was hoping for. The E1 has been used on a Hardware based PBX until a week or so ago, so I can't see there being an issue there. We are looking at replacing our existing PBX with an Asterisk Machine and the E1 live for another month before it gets switched off (we just switched providers and now have an additional E1 kicking around doing nothing..) Another question... and I know that it's all relative to experience, complexity etc, but how long should I take into account to get some basic voicemail, messages on hold, and basic IVR type stuff up and running? We have several programmers / linux admins in house and they are pretty good at what they do, and have been canvassing the lists to work out the details... but we lack experience at the moment. We have also been looking at various GUI's for Asterisk... (Asterisk@home being one)... can anyone recommend one that would be ideal for a business user in a basic small / medium office environment? Thanks again, Callum -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, 4 March 2005 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question Callum McGillivray wrote:> Can anyone tell me from experience how long it might take to get it up > and running so that we can make some basic test calls ?If the hardware is functional and the E1 is provisioned properly, a decent admin should be able to have Asterisk running and making test calls in a couple of hours. That's a long way from 'done', though :-) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you.
I installed an Asterisk@home machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys
Contact me offlist and I will gice you some info.... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of bram Sent: Friday, March 18, 2005 10:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie question I installed an Asterisk@home machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I guess the first time it didn't get through... I didn't see it appear in the list, that is... I installed an Asterisk@home machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys
bram kortleven Wrote>"Message: 6 >Date: Sat, 19 Mar 2005 22:16:39 +0100 >From: bram kortleven <bram@antwerpen.be> >Subject: [Asterisk-Users] newbie question >To: asterisk-users@lists.digium.com >Message-ID: <1111266999.7391.0.camel@athlon> >Content-Type: text/plain>I guess the first time it didn't get through... I didn't see it appearin >the list, that is...>I installed an Asterisk@home machineand configured a few SIP accountson >it. They seem to run fine inside my network, so that's OK. Now, I want to >start using a X100P to connect it to my phone line, to make call routing >between internal SIP phones/softphones, my local phoneline and an external >SIP server. How do I enable and configure the X100P?>I ran the configuration tool locally on the machine (the genzaptelconf >thing) and it added a line to the config. >Now using the number it gave me, in the trunk config in AMP, I stillcannot >get an outside line (connected it to a simple analogue pbx>system) and call outside the *-server.. >Could anyone help me with this? >Thanks guys"You need to go into the Zapta.conf and remove the semi colon ; channel => 1 Jeff
Hi folks, I have an asuscom ISDN BRI card in my server and was wanting to know whether this would be good enough to use with Asterisk. I am VERY new to this, so have no idea how to config the software, etc. But I am very eager to learn. Cheers H
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hamish Whittal wrote:> Hi folks, > > I have an asuscom ISDN BRI card in my server and was wanting to know > whether this would be good enough to use with Asterisk. I am VERY new to > this, so have no idea how to config the software, etc. But I am very > eager to learn. > > Cheers > HThis should work fine with ISDN4Linux as it uses the Hisax chipset. Get it working first as a normal ISDN card, then add Asterisk. Look at /etc/asterisk/modem.conf, it will need to contain something like this: [interfaces] context=inbound driver=i4l language=en type=autodetect stripmsd=0 dialtype=tone mode=immediate msn=yourMSNhere group=9 dtmfmode=asterisk incomingmsn=* device => /dev/ttyI0 device => /dev/ttyI1 You _WILL_ need to set your MSN (change yourMSNhere to your full MSN, usually without the area code). HTH - -- Ron Wellsted http://www.wellsted.org.uk ron@wellsted.org.uk FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQpOLWEtP/KMNOfRbAQKyMwf5AWyv903Zk04g9g/lNGIErfDdQHlo7Vr7 lThVhfMScoW5PlFI3UIVjjzikwZYsqYUBL0aEevFa64hV40u84ZzQ0WbgLzTekgB qPOaFXwzI//D7GQUOcKpA9ydlWYv7y16BQ2lgj93DP/xttImu6qf+HwF2vzhYMmE TxC2CUkU3zeMnFYy3QGBHPyVJVH+jRD5C3jgnnWfGeUVYBjQhVA3PQHZ+nLw/Ip+ 3Iv5R4wcDrtM1F08ZLkH1NJ2D07NvmkasglEyaJlCDBh6LGyBexuiSjUAn8cxH3Z 3cfmDfKtWjAFZV060LyAnQQYAmhrnDc88/HRbIzHTVJJ6/+uPduHMQ==9O75 -----END PGP SIGNATURE-----
Greetings, I have my first asterisk installation up and running, thanks to a lot of reading. Could anyone point me in the direction of things to read on automated outbound dialing? NOT predictive dialing - I will not have agents handling the calls. These calls are reminders for appointments, etc. Thanks! Charles
read in voip-info.org about Asterisk Call Manager API, and may be an easier soultion are the .call files that you can pleace in /var/spool/asterisk/outgoing/ these files have a description of the type of call you wanna make, in the very moment that you place the file there, a call will be Originated automagically. With Asterisk Call Manager API you can do something similar ( or equal ) using the "Originate" action. best regards On 6/8/05, Charles Austin <ceaustin@gmail.com> wrote:> Greetings, > > I have my first asterisk installation up and running, thanks to a lot > of reading. Could anyone point me in the direction of things to read > on automated outbound dialing? NOT predictive dialing - I will not > have agents handling the calls. These calls are reminders for > appointments, etc. > > Thanks! > Charles > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. vivek@staff.ownmail.com Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends.
I dont need to configure zaptel device, you dont use it :) 2005/11/30, vivek@staff.ownmail.com <vivek@staff.ownmail.com>:> Hello friends, > I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? > > I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? > > Kindly help me solving my doubt. > > > With warm regards. > > Vivek J. Joshi. > > vivek@staff.ownmail.com > Trikon electronics Pvt. Ltd. > > --Truth springs from argument amongst friends. > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Giovanni Miano
Thanks Mr.Miano Thanks a lot. Now I think I wont have to bother about balming all my problems to zapata. I have also succeeded quite a bit and installed a basic PBX system without it. Thanks a lot again. With warm regards. Vivek J. Joshi. vivek@staff.ownmail.com Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Giovanni Miano wrote:> I dont need to configure zaptel device, you dont use it :) > > 2005/11/30, vivek@staff.ownmail.com <vivek@staff.ownmail.com>: > > Hello friends, > > I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? > > > > I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? > > > > Kindly help me solving my doubt. > > > > > > With warm regards. > > > > Vivek J. Joshi. > > > > vivek@staff.ownmail.com > > Trikon electronics Pvt. Ltd. > > > > --Truth springs from argument amongst friends. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > Giovanni Miano
Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller --------------------------------- Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060215/a784ad5a/attachment.htm
On Wed, 15 Feb 2006 08:59:22 -0800 (PST) housi mueller <melicf2005@yahoo.com> wrote:> Hi there, > > I would like to connect an Aasterisk Server with a >Panasonic PBX (has E1extension). > I only need 4 Lines. So I thought I could use an >Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 >card which is more expensive. > > I dont now which card to take. > > Please tell me what you think about. I appreciate all >suggestions. > > Thanks in advance > > Housi Mueller > >My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box. Spend a little more now and save in the future. Just my $.02 Robert
That is a good argument. But I am not sure yet. Do you know if there are big voice quality differences between the Digital and the Analog card? Housi Robert Webb <asterisk@ropeguru.com> wrote: On Wed, 15 Feb 2006 08:59:22 -0800 (PST) housi mueller wrote:> Hi there, > > I would like to connect an Aasterisk Server with a >Panasonic PBX (has E1extension). > I only need 4 Lines. So I thought I could use an >Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 >card which is more expensive. > > I dont now which card to take. > > Please tell me what you think about. I appreciate all >suggestions. > > Thanks in advance > > Housi Mueller > >My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box. Spend a little more now and save in the future. Just my $.02 Robert --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060215/6c221803/attachment.htm