Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not support SS7, so is there a problem for the call id ?) Thanks in advance. Konstantinos. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030621/3925779c/attachment.htm
On Sat, 2003-06-21 at 11:02, CSTe wrote:> Hi..... > > I am new to this software, and I want to implement a client (SIP or > IAX) with PHP or at least to pass the main functions (connection,call, > transfer, hangup, call id etc) to a CRM. > Does anyone know if I could achive a project like that with AGI ? Any > example using AGI with PHP ? > Do I have all the functionality with AGI ?You couldn't implement a SIP or IAX client in php unless you where using php as a cgi and possibly used it with the gtk extensions. Anyways, this probably wouldn't get you what you think it would. What you really want is something that hooks into the manager interface to provide those functions. And a webapp is not the most useful for that. Your going to need bidirectional communication with the clients. If it was done as a webapp you would probably have to set a quick refresh on the page to keep it up to date with what was going on, and all those connect, query, display. disconnects are going to be a stress on your pbx that you don't want to put it through.> What about call id ? What is depend on ? (As I know * does not support > SS7, so is there a problem for the call id ?) > Thanks in advance.CallerID can come on an analog connection with FSK tones, it can come on PRI via tha facility information plus calling number information. I've had at least 1 telco not beable to provide callerid on a channelized T1. -- Steven Critchfield <critch@basesys.com>
hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks.
I'll apologize right away for asking stupid questions. :-) System Setup: SER = Proxy Asterisk = Voicemail All sip based setup. 1. What Is required to make asterisk -NOT- accept inbound calls/signaling from an unknown host? I tried the peers in sip.conf but it still allows unknown hosts to send it calls. Does anyone have a suggestion or maybe some sample configs? 2. I'm trying to extensions.conf dynamic. Is there any other alternative to the DynamicDB program to do something like that at this time? I'm trying to avoid having to restart * every time we make a change/addition. 3. I'm going to be rolling out a fairly large installation of Asterisk. What is the best way to have them all have the same configs/be synchronized? 4. Does anyone have any good tips/advice on SER+Asterisk integration? I appreciate it. - Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040404/80ca370a/attachment.htm
Hi, I am new to asterisk. And I have some newbie questions :-) I like to use asterisk I our office (around 20 phones) but we need to see if a user is at the moment using his phone so he can not get a second call from someone. Sometimes this is called a telephonecenter where I can see the used "lines". Is it possible to configure the frontend to autoaccept internal incoming calls? Our employees often use this feature to search each other .... Yours, Nicolai ---- Diese Nachricht wurde auf Viren und andere gefaehrliche Inhalte sowie Spam untersucht.
Hi everyone, I'm going to be helping to set * up for the company I work for, and in doing all my research about it, have found it to be a very viable solution for my SOHO side business at home. I do however have a few questions, forgive me if they're stupid but I'm new to all of this. 1. I want to be able to handle 3 analogue phone lines, with a regular bell telephone line coming into the house. So am I to assume that I want one PCI card for a P300mhz or above with three FXS ports and one FXO port? (the TDM31B). Am I correct in the card that I want? 2. Relating to my first question, say I instead get a card with only one FXS port and one FXO port, can I 'chain' my phones together from the one FXS port and still get the same functionality? (what i mean is one phone line coming out, with a splitter going to my three telephones)? 3. In the future I will be wanting to upgrade to VOIP capabilities for my SOHO Long Distance, is this as simple as getting another card with a T1 interface and an interface port for the phone, then plug it into my existing LAN to get internet connectivity, and still use the TDM31B for regular analogue conversations? 4. Does * support 'ring tone identification' ? Currently I have the outside line coming into the house, then it's split to go off to two phones, then from one of the phones the 'second extension jack' is going to my fax machine, which recognizes the distinctive ring the phone company gave me for the fax #. Will this still work with asterix, or would the fax machine have to be coming directly off the port on the PCI card? 5. Relating back to the splitting of the phone lines, if I have a card with two FXS jacks, and one FXO, and I only wanted two extensions on the line (upstairs, downstairs), could I chain the upstairs lines on one analogue line, and then if i transfer a caller to that extension it will ring on both phones upstairs? Hopefully I'm clear on my questions, Thanks a lot in advance. Matt Gibson Unix Administrator Experthost / NJ Tech Solutions
> > From: Matt G <gibson@experthost.com> > Date: 2004/07/28 Wed PM 08:50:03 GMT > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Newbie Questions > > Hi everyone, > > I'm going to be helping to set * up for the company I work for, and in > doing all my research about it, have found it to be a very viable > solution for my SOHO side business at home. I do however have a few > questions, forgive me if they're stupid but I'm new to all of this. > > 1. I want to be able to handle 3 analogue phone lines, with a regular > bell telephone line coming into the house. So am I to assume that I want > one PCI card for a P300mhz or above with three FXS ports and one FXO > port? (the TDM31B). Am I correct in the card that I want?I believe so. You will need one FXO and three FXS ports, and you should be able to get them all on one card.> > 2. Relating to my first question, say I instead get a card with only one > FXS port and one FXO port, can I 'chain' my phones together from the one > FXS port and still get the same functionality? (what i mean is one phone > line coming out, with a splitter going to my three telephones)?Yes, that should work, but there's a limit on how many telephones one can drive.> > 3. In the future I will be wanting to upgrade to VOIP capabilities for > my SOHO Long Distance, is this as simple as getting another card with a > T1 interface and an interface port for the phone, then plug it into my > existing LAN to get internet connectivity, and still use the TDM31B for > regular analogue conversations?No. A T1 card does not plug into a LAN. You will need to use the ethernet port on your server, or add a NIC. Once you do so, and get all of that configured, you can bridge calls between your analog ports and SIP/IAX/other VoIP phones.> 4. Does * support 'ring tone identification' ? Currently I have the > outside line coming into the house, then it's split to go off to two > phones, then from one of the phones the 'second extension jack' is going > to my fax machine, which recognizes the distinctive ring the phone > company gave me for the fax #. Will this still work with asterix, or > would the fax machine have to be coming directly off the port on the PCI > card?Asterisk has distinctive ring support, but I have not worked with it yet.> 5. Relating back to the splitting of the phone lines, if I have a card > with two FXS jacks, and one FXO, and I only wanted two extensions on the > line (upstairs, downstairs), could I chain the upstairs lines on one > analogue line, and then if i transfer a caller to that extension it will > ring on both phones upstairs? >Yes, it should. No different than a normal analog line that you get from a telco. Again, there's a limit on how many phones one line can drive.> Thanks a lot in advance.You're welcome...
>I'm going to be helping to set * up for the company I work for, and in >doing all my research about it, have found it to be a very viable >solution for my SOHO side business at home. I do however have a few >questions, forgive me if they're stupid but I'm new to all of this.OK. I'll bite.>1. I want to be able to handle 3 analogue phone lines, with a regular >bell telephone line coming into the house.In "phone lingo" the phone sets are the "station" side of the PBX. The lines coming into the house are the "trunk" side. Sometimes, but not often, the station side is called "line-side" but usually to referrence interconnecting tie lines.>three FXS ports and one FXO port?Correct.>can I 'chain' my phones together from the one FXS port ...Yes, but they will all be the same extension just like all the phones in your house and not be individually addressable (dialable).> upgrade to VOIP capabilities for my SOHO Long Distance, is this as >simple as getting another card with a T1 interface ...No. A T1 interface that goes directly into the PBX (Asterisk) is usually for voice (23B+D (23 bearer channels for voice + one data channel for signalling)). You will most likely already have a 100BT connection on your server and that is where you will get the most cost effective in/out IP connectivity to your box. You will then connect your network - or that segment of your network - to the outside IP world either directly or via firewalls/routers.>Does * support 'ring tone identification' ?Yes.> Relating back to the splitting of the phone lines,...See above. Matt, you really need to spend _a_lot_ of time reading the documentation and playing with the system. There is no substitute for hands-on experience. I have had a long history in data and telephony and I still played with the product for 4 months before I asked a question. Until you spend that amount of time learning you will not have the background to understand the answers that people give you. Most people on the list won't answer a question like this one because it has been well shown that they are wasting their time teaching someone who is not ready for it yet. The other side of the coin is that the people on this list that have spend copious months of their time gaining expertise are perfectly willing to support peers who have the invested in the same manner. They are not however, willing to spoon feed people who have not yet, or appear unwilling to make that investment themselves. Those people need to hire consultants. If you do want to hire a consultant - which there is nothing wrong with do so - just ask for such on the list and there will be manny people willing to provide rates for their services. I know that email is a cold medium and this may come across badly at first, but that does in fact represent the culture of a user community. You need to read up first to gain a minimum level of expertise _as_a_user_ in order to productively take part in the user community.>Hopefully I'm clear on my questions, >Thanks a lot in advance. >Matt Gibson >Unix Administrator >Experthost / NJ Tech Solutions-- David Cook
Hi all, I'm new to this list. very excited about Asterisk. I'd like to find good docs about it. Can you advise me on docs ?.Or even books ?. If I only want to use asterisk in order for my friends and family to call each other by passing through my pbx (without going in the usual telephone system) I can right ?. All I need in that case is a PC right ?. No Xtra hardware needed ?. /Hitete
----- Original Message -----> Hi all, > > I'm new to this list. very excited about Asterisk. I'd like to find good > docs about it. > Can you advise me on docs ?.Or even books ?.www.voip-info.org/wiki-asterisk www.asteriskdocs.org I think that a fellow who routinely posts to this list named Paul Mahler(? sorry if this is incorrect) recently published a book about *. be prepared though, as with any open source technology, documentation is '..in process..'> If I only want to use asterisk in order for my friends and family to call > each other by passing through my pbx (without going in the usual telephone > system) I can right ?.yes> > All I need in that case is a PC right ?. No Xtra hardware needed ?.that is correct. you may need to simulate some ZAP channels to get timing for certain applications but this is outlined in the references I have noted above. welcome, and have fun. feel free to document your learning processes to help the next newbie that posts the same question, for if you continue to monitor this list, you will see this question again. Jason Kawakami www.optellabs.com
Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ....) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows & gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings & samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson
I've got one freaky budgetone that wont work using dhcp assign ip address via mac code. Basically I need to assign it an ip address using the phones internal web server. Maybe this was your problem as well. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Panco Sent: Tuesday, February 08, 2005 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] newbie questions i installed it the other day but from some reason can only get one of my budgetone 100's to register...any thoughts? I have tried upgrading firmare but that didn't seem to work. thanks in advance, ken Steve Rawlings wrote:> Why not try asterisk@home, it only takes about an hour to install and > be up and running with softphones like x-lite. This takes care of the> os and asterisk in one cd. > > Steve > > > ----- Original Message ----- From: "Shaoul Jacobson - TELLINK" > <shaoul@tellink.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Tuesday, February 08, 2005 4:44 PM > Subject: [Asterisk-Users] newbie questions > > >> Hi, >> >> I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDNcards)>> >> >> 1. the distro >> I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem >> missing >> (some C or C++ or python ...) >> (buy the full version ....) >> >> maybe the latest fedora is more complete ? >> or easier to complete with rpmfind >> (I am green to linux too, but I open my windows & gates to the tux) >> >> (bsd, debian are a bit too tech for me yet, no flaming please.) >> I prefer ready made rpm's or alike than compile AT THIS TIME. >> (I promise to improve over time) >> >> >> 2. download >> any rpm ? or I must download sources and 'make install' ? >> (I found one iso, but it seemed to require a pstn card) >> (RTFM a second / third time could is always a good option) >> >> 3. pure VoIP >> is it ok to use it in pure VoIP mode without any 'phone cards' ? >> all (most) settings & samples I see include such cards. Needed or not?>> >> >> 4. g729 not free. >> It seems that requires some licensing to digium. >> Can that be without using any 'card' (just VoIP) ? >> How to control the licenses then ? >> (I e-mailed them the question, but got no answer) >> >> >> accounting, cdr's, ... that's for later >> (first I have to be able to phone) >> >> >> regards, >> >> Shaoul Jacobson >> _______________________________________________ >> 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 > > >_______________________________________________ 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
Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. -- Jean-Francois Theroux Systems administrator PrivalODC 514.726.3732 http://www.privalodc.com
I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is "Awaiting Proxy Information" in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring
Ignore the error if it isn't messing anything up. Check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk A search of X-lite here also yields proper setup info for the softphone to Asterisk connection. The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> Als try the documentation link at digium.com Regards, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie questions I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is "Awaiting Proxy Information" in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring _______________________________________________ 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've read through a good amount of documentation on voip-info.org, but >hadn't found a solution, so I thought this list might help. I'm not >great with linux, and I suspect there might be a port problem... maybe >Asterisk isn't listening for SIP clients. How would I go about >checking this? X-Lite configuration is pretty straightforward, you >just give it username/password and point it at a SIP proxy. However, >as far as I can tell it isn't able to register, or it's not listening >to Asterisk... hard to tell really.If you RIGHT-click on the sliver skin of the X-Lite console and LEFT-click on "Diagnostic log" you will see the debug information as X-Lite is trying to register with Asterisk. The operative part is: "SIP/2.0" followed by a number and a status code. You will note that the status code is the same as HTTP status codes I.E. "2XX" = OK, did it and "4XX" means your client is the problem and "5XX" is something is wrong with the server. Looking thru the diag log will get you started in the right direction to troubleshoot. For the codes, look here: http://www.zvon.org/tmRFC/RFC2543/Output/chapter5.html hth
I am sure that asterisk is listening for SIP clients. Did you configure your sip.conf correctly? More info to look at... site:lists.digium.com sip x-lite If you are building this form scratch and cannot get the basics compelted, I would just dump it and go to a build of Asterisk@Home. The built in GUI lets you get basic install completed very quickly. Cheers, W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:51 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] newbie questions I've read through a good amount of documentation on voip-info.org, but hadn't found a solution, so I thought this list might help. I'm not great with linux, and I suspect there might be a port problem... maybe Asterisk isn't listening for SIP clients. How would I go about checking this? X-Lite configuration is pretty straightforward, you just give it username/password and point it at a SIP proxy. However, as far as I can tell it isn't able to register, or it's not listening to Asterisk... hard to tell really. -Brian On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler <wsiler@education2020.com> wrote:> Ignore the error if it isn't messing anything up. > > Check out the Wiki here.... > http://www.voip-info.org/tiki-index.php?page=Asterisk > > A search of X-lite here also yields proper setup info for the > softphone to Asterisk connection. > > The archive of this list can be search via google by entering... > site:lists.digium.com <some parameter> > > Als try the documentation link at digium.com > > Regards, > Wiley > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian > Nehring > Sent: Monday, March 07, 2005 2:26 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] newbie questions > > I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and > could use some help. I'm investigating migrating our small business > phone system over to Asterisk and VOIP. Eventually we'll have around 4> incoming SIP (or IAX if I can find one) accounts for PSTN > incoming/outgoing, then SIP hardphones in the office. I installed > Asterisk on OS X, which might be why I'm having problems. I have > Asterisk up and running fine, although it's giving one warning on > startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 > (set_config): Ignoring port for now. > > I'm not too concerned with this, because for now I'm just trying to > get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I > don't need IAX listening on whatever port isn't working. > > I setup a very basic config to let X-Lite connect, but all I see is > "Awaiting Proxy Information" in X-Lite. I see with netstat on the > server that it has a UDP for *.sip open, so I think it should be > listening for incoming, but it seems like it's not. I don't see a > firewall running, so I'm not really sure what's going on. I should be > getting an SIP hardphone in later this week, but I'd like to try to > get this debugged now. > > If anyone could help I'd be much appreciative. If you guys have any > more questions or want to see my config files, please ask. > > Thanks, > -Brian Nehring > _______________________________________________ > 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
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've some questions about asterisk, and in general about voip, please help me :) 1. I've SIP accounts on external servers, and I would that my local server will connect with those and redirect all calls from those to an internal SIP account (just one). It's possible to do that? In this case, I think asterisk will work as UA for external accounts, and as sip server for internal. I've to use SER with asterisk? 2. the internal account it's important that will be SIP, or I could forward calls from my external sip account to an h323 account? 3. I could configure a voicemail account (with an internal number) for all calls that I would redirect from all internal phones? 4. I could use a welcome message on an internal account, and/or auto attendant? I hope this is clear. Any advice to put me in the right direction will be appreciated. Regards Andrea -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCQ+AHMakHrsrHP9wRAmbDAJ428+4F+R/RSv0CGMVZVwo73z1OAwCgoiDe F6eXRsp/JX4QD78tDE9Jiro=Zx81 -----END PGP SIGNATURE-----
i am pretty new to asterisk. hope to learn more. i have this notice from the console. when i was doing the echo testing by putting the context=default. then, i called out 600 to get the echo test, i can hear the operator talking, but i cant really hear the playback. i am trying to dig around from info from the log files. what does it mean? RFC3389 support incomplete. Turn off on client if possible hope to help......thanks
Hello Hiu, Monday, November 7, 2005, 4:51:35 AM, you wrote: HYO> i am pretty new to asterisk. hope to learn more. HYO> i have this notice from the console. when i was doing the echo testing HYO> by putting the context=default. then, i called out 600 to get the echo HYO> test, i can hear the operator talking, but i cant really hear the playback. HYO> i am trying to dig around from info from the log files. HYO> what does it mean? HYO> RFC3389 support incomplete. Turn off on client if possible HYO> hope to help......thanks That means that you have to turn off silence suppression in your softphone (in xlite is named "transmit silence"). Hope it helps! -- Best regards, Alessio mailto:afoc@interconnessioni.it
Hi all I am new to this whole field, being it PSTN or voIP. I am currently reading the "Switching to VoIP" and "Asterisk: The Future of Telephony", so hopefully, I will be less clueless soon :) My first question: if I buy a Wildcard TDM400P, with one X100M and three S100M modules, I would be able to have 1 telephone number given out by my company to come in to my asterisk server, and I could plug in 3 analog phones onto that card, am I correct? Hence, do we have a 1-to-1 relationship here for either modules? My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? Thank you all. Cheers fred -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051117/09dc335e/attachment.pgp
hi,>My second question: for a branch office of about 20 people, which E1 >card do you advise? Would the TE210P be a good choice? (number of >concurrent calls would be max 10 for now) Why? > >An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Jan