Re: [Asterisk-Users] fedora core 1 install problem
Justin Sinclair wrote: From: David Luyens [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] fedora core 1 install problem Date: Mon, 22 Dec 2003 14:20:55 +0100 Reply-To: [EMAIL PROTECTED] Hi Ernest, I have installed as you described, and now it worked. Seems that installing a minimum system and afterwards installing the necesary packages with their dependencies seems to not have worked for me Thanks for the help all... David It actually is possible to get Asterisk running on a Fedora Core 1 minimal installation, installing the necessary packages afterwards. I also had the original problem you had, but figured out how to fix it. Here are the steps I took to do so (this is not a recommendation of how to do it, just how I did it): 1. Install Fedora Core 1 (minimal install) 2. To make installation of additional packages super-easy, I install yum: #wget http://ftp.freshrpms.net/pub/freshrpms/fedora/linux/1/yum/yum-2.0.4-2.fd .fr.i386.rpm #rpm -U yum-2.0.4-2.fd.fr.i386.rpm 3. [Optional] Run a yum update to get latest version of installed packages (good idea to reboot after this, especially if you get a new kernel): #yum update 4. Install required packages using yum (other required packages are already installed as part of minimal install): #yum install cvs gcc kernel-source libtermcap-devel newt-devel ncurses-devel openssl-devel readline-devel 5. Now here's the trick. Download and install this older version of Bison (the newer version causes the compile errors): wget ftp://rpmfind.net/linux/redhat/9/en/os/i386/RedHat/RPMS/bison-1.35-6.i38 6.rpm rpm -U bison-1.35-6.i386.rpm 6. Your Fedora Core 1 installation is now ready for Asterisk. Download, compile, and install as normal. I hope this proves useful to others who prefer a bare minimal install, and want to use the excellent Fedora. -Justin Cool Justin, I was going to attempt an install on Fedora in the next little while.. Has anyone created a bug report about the errors when using the new version of bison?? While your work around is fine it does add one more thing to worry about when installing package updates.. If bison is updated by YUM or APT then Asterisk will have issues.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] automatic voice dialout call
I need to make automatic voice calls from a Linux server, so when the system receive special signals it must use a wave (or .au) audio file, dial the number to call a person, and speak using the audio file. What can i use for this subject? I need a specific hardware device or a normal analogic voice-modem is ok? which software can i use (i need to invoke from a Perl script)? Regards -- Davide Giunchi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Config thru web interface or any GUI
hi, i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers? Suggest. Chandra
Re: [Asterisk-Users] automatic voice dialout call
Davide Giunchi wrote: I need to make automatic voice calls from a Linux server, so when the system receive special signals it must use a wave (or .au) audio file, dial the number to call a person, and speak using the audio file. What can i use for this subject? I need a specific hardware device or a normal analogic voice-modem is ok? which software can i use (i need to invoke from a Perl script)? Please check the docs! A good starting point for both questions http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ In short, Yes, you can generatte automatic voice calls. No, you can't use a voice-modem, you propably need - at the lowest level - an ISDN bri card or a subscription with a VoIP provider. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI
Chandra wrote: hi, i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers? There's some info to help you on the FAQ http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI
Chandra, Take a look at: http://sourceforge.net/projects/astguiclient/ it may be what your looking for or you could use the ideas if you want to make changes. I believe it was written by Matt Florell, Thanks Matt. At 14:42 30/12/03 +0545, you wrote: hi, i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers? Suggest. Chandra Peter Brown
[Asterisk-Users] I wanna buy a new X100P
I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing calls from a T1 based on DNSI.
Title: Routing calls from a T1 based on DNSI. Dear Group, I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX. On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality. Does anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving me a sample of what this would look like in the configuration file? Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management
Re: [Asterisk-Users] I wanna buy a new X100P
I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? Isamar Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html You won't get the whopping 95 cent discount from BSD Mall but you'll be buying it directly from Digium AND have their support. http://www.digium.com has likes for ordering their hardware. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI
is there a installation guide? i didn't find any. just the readme file. - Original Message - From: Peter Brown To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 2:58 PM Subject: Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI Chandra,Take a look at: http://sourceforge.net/projects/astguiclient/ it may be what your looking for or you could use the ideas if you want to make changes.I believe it was written by Matt Florell, Thanks Matt.At 14:42 30/12/03 +0545, you wrote: hi,i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers?Suggest.Chandra Peter Brown
[Asterisk-Users] E100P configuration
Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this make sense ? context=default switchtype=euroisdn signalling=pri_net ;pridialplan=national overlapdial=yes group=1 channel = 1-15,17-31,32-46,48-62 what does channel include ? all the channels d and b ? Thanks for your help. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS Closed?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- All good here also.. this has got to be in the to 10 stupidest things posted to the mailing list today. Stupid as in development stopped, agreed. However, there have been times in the past where cvs.digium.com has not resolved for me either. Now, seeing that both nameservers apparently are on the same subnet (216.207.245.1 and .12 respective), I would guess that the intermittent unresolved messages are due more to connectivity problems to the nameservers. Two rules of thought when I comes to placement of NS servers: 1) If the name servers are local to the domains the resolve for, no need for geographic distribution. I.e., if the network is unavailable for a [short] period of time, ain't no one getting there no how. 2) Place a slave DNS server at some other location. In the event of that same outage or lack of connectivity to the destination, names will resolve but the connection will timeout. I guess the morale is, if you can't resolve cvs.digium.com, shake the Magic 8 Ball and Try Again Later. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk support legacy Dialogic products?
Patrick Wong wrote: Hi all, I just checked out that Asterisk which is a platform I am interested of. I would like to install it to the Linux box for a trial. I have some legacy Dialogic hardware on hand, don't know they will work with Asterisk or not. For analog loop start interface I have Dialogic D/41 E which is of ISA bus with four Telco interfaces. Will it work on Asterisk? Best regards, Patrick. That card's haardware is not capable of providing any VoIP functionality. It is not full duplex. The newer JCT cards can be used, but they still don't work that well, due to card limitations. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk support legacy Dialogic products?
Steve Underwood wrote: That card's haardware is not capable of providing any VoIP functionality. It is not full duplex. The newer JCT cards can be used, but they still don't work that well, due to card limitations. This is exactly why people should support Digium. After all they have GIVEN the world Asterisk. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing calls from a T1 based on DNSI.
On Tue, 2003-12-30 at 04:04, Shad Mortazavi wrote: Dear Group, I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX. On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality. Does anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving me a sample of what this would look like in the configuration file? What type of T1? You need to know more about the services being provided to you. DNIS(dialed number information service) can be done via EM wink or via PRI. On EM wink channels it is likely to be 3 to 4 digits, PRI can be anything up to the full 10 digit number. After that you just define those numbers provided from the telco as extensions in your incoming context. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ser and Arterisk works together ?
Hi, Anybody knows if Asterisk work fine with ser ? We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN termination for inbound and outbound calls. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E100P configuration
Hi- Not sure that I understand your question about grouping, but here is what I use for 2 E1's connected to a private switch (in addition to the other parameters) Note that I use the pri_cpe (customer premise equipment) setting. The defined channels act as one big group of 60 channels, if that's what you mean. Your telephone company will define the call distribution for your incoming calls: pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 Regards, Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dawid Mielnik Sent: Tuesday, December 30, 2003 11:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E100P configuration Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this make sense ? context=default switchtype=euroisdn signalling=pri_net ;pridialplan=national overlapdial=yes group=1 channel = 1-15,17-31,32-46,48-62 what does channel include ? all the channels d and b ? Thanks for your help. 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
Re: [Asterisk-Users] CVS Closed?
Carl, You would have to know me to understand that was a JOKE! Most people on the list and in the irc channel know already! Lighten up and live a little. Digium is about to setup CVS mirrors because if * is on /. one more time and I have to do that .5kb/sec CVS checkout.. i'm gonna scream bkw On Tue, 30 Dec 2003, Carl A. Cook wrote: On Monday 29 December 2003 11:18 pm, Brian West wrote: All good here also.. this has got to be in the to 10 stupidest things posted to the mailing list today. bkw Very nice Brian, thanks for the adversarial 'welcome'. Is this the kind of treatment I am to expect here? If so, it's not worth it. (My DNS is working fine) On Tuesday 30 December 2003 06:16 am, Adams, Gavin wrote: Stupid as in development stopped, agreed. Be advised, that the newest tarball is 4 months old. Can you explain to me what a normal visitor to the website is supposed to think, when CVS does not resolve and old tarballs? Not sure any more I'm interested in learning this, to put up with reflexive schite. There's enough crap in the world. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Output from gpg gpg: Signature made Tue 30 Dec 2003 10:57:28 AM CST using DSA key ID F15C649B gpg: Can't check signature: public key not found ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virtual PC -- Asterisk ?
Hi, - Original Message - From: Steven Critchfield [EMAIL PROTECTED] Ahh, but the question is worded such that the virtualization is running on windows. Therefore you have a lot of display overhead due to a windows environment. You also are just an application running in an OS, so you have to convince the OS to give you appropriate resources. So while the application isn't necessarily too inefficient, you are already running in an OS that can starve your emulator, and then you have another OS that can starve asterisk from running at the required speed. I have done some tests in the past: - Athlon [EMAIL PROTECTED], 384MB RAM - Windows XP Pro as the host platform - VMWare Workstation version3 - Asterisk (a CVS from Oktober I think) - RH9 - no Digium Hardware (cannot be used with a virtual platform) - two analog phones connected to an ATA186 Again, it is possible, just not recommended no matter what the underlying hardware is. Give asterisk at least a chance of working properly on its own before you handicap it. As the learning curve is enough already, don't augment it by adding artificial barriers. Major drawbacks: - the cost of the virtualization platform (VMWare or something else) is bigger than a good old dedicated hardware platform. - no way to use any Digium hardware, which is in my opinion unacceptable - the load on the system is far bigger than on a dedicated one (increase dramatically with the number of active channels). As a conclusion: - it was a nice proof of concept - works preatty well for small demo's on a notebook (at the customer site), but for sure not as a production (even for home) environment. - can be used to better understand how Asterisk works - all those if you want to pay the price for the virtualization platform. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS Closed?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Carl A. Cook On Tuesday 30 December 2003 06:16 am, Adams, Gavin wrote: Stupid as in development stopped, agreed. Be advised, that the newest tarball is 4 months old. Can you explain to me what a normal visitor to the website is supposed to think, when CVS does not resolve and old tarballs? Well, since you subscribed to the list, I would think that the daily volume of messages would indicate the status of *. Carl, you'll be surprised at some of the messages that come through here. :) However, you are preaching to the choir in regards to the tarballs and release schedules for OSS projects. The good news is that a few users have spent a goodly amount of time prepping CVS to support tags, releases and other goodness. This should make it easier to support easier things such as daily tarballs. Jesse over at RT (http://www.rt.com) has a nice environment doing just that. Not sure any more I'm interested in learning this, to put up with reflexive schite. There's enough crap in the world. It's a rough crowd around here, but you'll get used to it. Just don't incorrectly attribute or create a message from an old thread and you'll miss most of the truly shite posts. :) Best regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Closed?
On Tue, 2003-12-30 at 10:57, Carl A. Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 29 December 2003 11:18 pm, Brian West wrote: All good here also.. this has got to be in the to 10 stupidest things posted to the mailing list today. bkw Very nice Brian, thanks for the adversarial 'welcome'. Is this the kind of treatment I am to expect here? If so, it's not worth it. (My DNS is working fine) The treatment isn't too inappropriate. The exact question you posed could have easily been answered by any number of search patterns in just about every search engine. Even a lame amount of work on your part should have uncovered a mailing list of CVS checkins that usually doesn't goes more than a day between updates. BTW, either your DNS wasn't working, or it was a long time since anyone down your way had pulled that information since a caching name server will hold that data for at least 1 day from the last successful lookup. But this also points to the fact that you could have done at least the minimal effort of having checked to see where the DNS problem was occurring. I know that can be difficult during a outage of sorts, but seeings how your message went to the mail server located near asterisk took some time between hops, you very well could have had problems on your end. Maybe not all encompassing, but problems non the less. Don't expect to be treated well when you ask questions that are easily answered with low effort. For that matter, don't expect to be treated too well when asking questions that are already answered either. If you ask questions that enhance the knowledge of all, you will get respected. No matter what level of expertise a person holds here, we all are here to learn. The archives are available for your perusal. The wiki is a great place where Olle has condensed a lot of the information from the archives into a single place without the extra fluff. On Tuesday 30 December 2003 06:16 am, Adams, Gavin wrote: Stupid as in development stopped, agreed. Be advised, that the newest tarball is 4 months old. Can you explain to me what a normal visitor to the website is supposed to think, when CVS does not resolve and old tarballs? Not sure any more I'm interested in learning this, to put up with reflexive schite. There's enough crap in the world. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iEYEARECAAYFAj/xrngACgkQnQ18+PFcZJtBTACeO2zdE7i8loyEsvBXPbMQ9pcK BbIAn1tOnD91eynKCO+8rHo0TXsWjH0W =4cxA -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS Closed?
Don't say that. Does that mean that from now on we will get a voice asking if we really want to do that at every button press? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: 30 December 2003 18:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS Closed? Tried to DL using CVS this eve, and it says: Unknown host cvs.digium.com. Has Asterisk development stopped? Digium was just sold to Microsoft. Philipp ___ 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] Backup Proxy Automatic Failover
Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term TDM is banded around too, but from my knowledge, TDM is trunking (probably some clever acronym relating to trunking), and in Asterisk's case, using the IAX protocol. This leads me to the big question; Is there anyway of shifting the load of one Asterisk server to another without breaking or loosing a call? I know that with Survivable Routing (Cisco's big on this), the ISDN interface is actually a router; so the Proxy is just used to decide the destination and LCR functions, and then hands off to a router. This of course, if a Proxy went down, would just prevent new calls from being made, whilst existing calls can continue merrily - until someone switches the Router off, or corrupts the IOS settings :-) At least with Routers, you can configure them to load manager effectively, but how do you backup and load manage Asterisk?? I using SIP, and will be using a bit of SCCP too, so any suggestions would be most grateful!! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS Closed?
On Tue, 2003-12-30 at 12:34, David J Carter wrote: Don't say that. Does that mean that from now on we will get a voice asking if we really want to do that at every button press? heh, I can imagine it now, a call that says, A call has been received. You will now need to restart your computer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: 30 December 2003 18:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS Closed? Tried to DL using CVS this eve, and it says: Unknown host cvs.digium.com. Has Asterisk development stopped? Digium was just sold to Microsoft. Philipp ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Backup Proxy Automatic Failover
I simply have 2 asterisk servers and have the clients point to a DNS SVR record for their proxy. The DNS record lists the primary and secondary with preference for the primary. This won't stop calls from being dropped if the primary goes down if you are routing them through the server, but it does ensure that calls placed while the primary is down will still go through. You could do some load management by putting multiple servers in the DNS record and use a DNS server that supports round robin responses. Stephen -Original Message- From: Adthrawn [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 12:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Backup Proxy Automatic Failover Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term TDM is banded around too, but from my knowledge, TDM is trunking (probably some clever acronym relating to trunking), and in Asterisk's case, using the IAX protocol. This leads me to the big question; Is there anyway of shifting the load of one Asterisk server to another without breaking or loosing a call? I know that with Survivable Routing (Cisco's big on this), the ISDN interface is actually a router; so the Proxy is just used to decide the destination and LCR functions, and then hands off to a router. This of course, if a Proxy went down, would just prevent new calls from being made, whilst existing calls can continue merrily - until someone switches the Router off, or corrupts the IOS settings :-) At least with Routers, you can configure them to load manager effectively, but how do you backup and load manage Asterisk?? I using SIP, and will be using a bit of SCCP too, so any suggestions would be most grateful!! Regards, Ad. ___ 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] Polycom Sip Registration
Hey, I am currently working on a Polycom 500 phone Asterisk solution, and the key is definitely to use the xml config files that Matt spoke of. That combined with an FTP server (setup like the sip docs say) work very well in getting the phone to do what you want. It then becomes getting the config files for Asterisk that will make it all work. I will update on what I finally have when I am done Sean -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 9:02 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Polycom Sip Registration Hello, The best thing to do is to use the XML config files. the web interface isn't the best way to do anything, it's best to kind of ignore it. MATT--- -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Friday, December 26, 2003 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom Sip Registration Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent ___ 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] Multi-line, multi-registration phones
Title: Multi-line, multi-registration phones I have hard phones that are capable of handling three calls at once. That is setup (apparently) through multiple registrations. My question is has anyone done this and what is the proper way of doing it? Do I have to setup (for 2 phones that have three lines) 6 sections in my sip.conf and setup 6 extensions to handle the registrations? Also, if I found by searching the web sample code for making both sip extensions ring when a call comes in, but what if I had 100 extensions? Seems like the string would get pretty long, is there a way to put all extensions in a single group and ring the group? All kinda is the same question. But thanks for the answer anyway Sean Garland
RE: [Asterisk-Users] Programming an unlocked ADSI phone?
What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the first try with current CVS. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 10:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update. As I mentioned in my first post (below), I had managed to get the phone to accept a download from asterisk, but it still said PLEASE PROGRAM ME after the download was completed. After further investigation it does appear I downloaded SOMETHING to the phone, because if I select the services button on the front of the phone, I get into a menu that says: Services Asterisk PBX Asterisk slot 2 available available If I select the first one in the list, the phone does change from PLEASE PROGRAM ME to ** Asterisk PBX** and there's a VMail softkey!! However, if I pick up the handset and replace it, I'm back to PLEASE PROGRAM ME again. Is this normal? My second problem is that when I dial voicemail from this handset now, which is ADSI-enabled, I see the following message on the text screen: Comedian Mail (C) 2002 LSS, Inc. Downloading Scripts I don't see any activity on the console in terms of logging, and I don't see any way to elevate ADSI logging. Within less than a second, I see: Comedian Mail download refused. Services is full At this point the voicemail's welcome script plays and the display changes to: Comedian Mail (C) 2002 LSS, Inc. Load Cancelled ADSI Unavailable at this point voicemail's works like it used to before ADSI. Is any of this ringing any bells with anyone? Any tips appreciated ... I've looked around but I'm really not finding any documentation on how to make this work. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 8:12 PM Subject: [Asterisk-Users] Programming an unlocked ADSI phone? Folks, I have an Astra 480 that was delivered without any programming. On the main LCD, it says: PLEASE PROGRAM ME Asterisk I thought I would program it, so I added adsi=yes for this extension to zapata.conf, and then I created an extension: ;; ;; Temp hack for programming phones ;; exten = 998,1,ADSIProg(asterisk.adsi) exten = 998,2,Hangup Dialing 998 from the phone worked - the phone said a download was available and asked me to accept, which I did. I saw: Connected to Asterisk CVS-12/17/03-02:39:14 currently running on testbed (pid = 31583) -- Starting simple switch on 'Zap/5-1' -- Executing ADSIProg(Zap/5-1, asterisk.adsi) in new stack -- ADSI Available on CPE. Attempting Upload. -- Executing Hangup(Zap/5-1, ) in new stack == Spawn extension (default, 998, 2) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1' -- Hungup 'Zap/5-1' When it's done and asterisk hangs up though, my phone seems unchanged. Still asks to be programmed. I can't seem to find a how-to on the wiki, or in the PDF. Looks like an addition might make sense, and I'd be glad to help if I could get this sorted out. Am I missing something obvious? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax ___ 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] X100p always busy - update
Title: X100p always busy - update Well, after bummin around thinking I had bad fxo cards, I finally discovered that I was loosing two phones in my home when I had the cards plugged in Turns out the jack I had plugged into, was wired for two phones (line 1, green/red and line 2, yel/blk) and I also was using a 4 wire phone cord for the connection. It turns out that the x100p cards are wired in such a way internally that if you are running a 4 wire phone cord to them, that you might short out between the two lines at the jack (which worked separately with phones and computers) and get funky results.. Moral of the story is to always use two wire phone cords with the x100p fxo cards. Problem solved, and I was able to continue my development. Thanks Sean Garland
[Asterisk-Users] Mac OS X
Hi, I've just read on the Wiki, that Asterisk can be compiled to run on Mac OS X (BSD). How?! I've just tried running 'make' through the command line, and it dies from a gcc bug. According to the Wiki, it does state that there is a specially modified make file that has been ported to correctly compile for *nix. Where is it?!? Best, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months -- .~. /V\Lance C. Arbuckle // \\ /( )\ ^'~'^ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line, multi-registration phones
On Tue, 2003-12-30 at 14:29, Sean Garland wrote: I have hard phones that are capable of handling three calls at once. That is setup (apparently) through multiple registrations. My question is has anyone done this and what is the proper way of doing it? Do I have to setup (for 2 phones that have three lines) 6 sections in my sip.conf and setup 6 extensions to handle the registrations? Also, if I found by searching the web sample code for making both sip extensions ring when a call comes in, but what if I had 100 extensions? Seems like the string would get pretty long, is there a way to put all extensions in a single group and ring the group? All kinda is the same question. But thanks for the answer anyway This is where variables come in handy, and also macros. For example, I just put several new variables into my extensions.conf file to deal with the changing nature of our dialplan. I defined a variable for each user that we have, and I created a ALL variable that strings all the users together. You could also make variables for all the users in certain departments, and then your ALL variable could then just include your department variables. The benefit of the variables is just that you can change it in one location and it ripples through all the other definitions. As for the multiple call appearances in SIP, I don't know how to deal with that. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 Register with 2 * Servers causes phone to reboot over and over
Hi, I have been trying to get my 7960 7960G to register with two seperate * servers. Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15 Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30 7960 is on the LAN running: P0S3-04-4-00 7960G is on the LAN running: P0S3-06-0-00 In sip.conf I have nat=yes to get the phones to register properly. And for a while both phones do actually work. However about every few mins they just restart! If I remove the second line that is registering with the server on the LAN they stop restarting Ideas? I assume this is a Cisco Bug of some kind, as the phone should reboot no matter what garbage * sends it. - Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] include a file ?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lance Arbuckle Sent: Tuesday, December 30, 2003 4:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] include a file ? Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months Check here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20openhours The wiki is a good place to start for these questions. -Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
On Tue, 2003-12-30 at 15:16, Lance Arbuckle wrote: Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months It does include the hash mark. I don't think it is capable of time dependent includes, but it can create contexts that are normally included with the time dependent options. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
No you guys need to pay attention.. include = context #include filename.conf They do totally diffren things. bkw On Tue, 30 Dec 2003, Lance Arbuckle wrote: Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months -- .~. /V\Lance C. Arbuckle // \\ /( )\ ^'~'^ ___ 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] CVS Closed?
heh, I can imagine it now, a call that says, A call has been received. You will now need to restart your computer. Or better yet... Who do you wanna call today? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
- Original Message - From: Lance Arbuckle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:16 PM Subject: Re: [Asterisk-Users] include a file ? Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months I expect the syntax is exactly as posted. I made up an example but then thought better of it. I was imagining it getting ugly, if you create contexts in the file... It might be safer to #include all the files you want ahead of time(near beginning of file), and then reference the contexts you need afterwards. I am curious which files (if not all of them) support this option... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
Sean Cheesman wrote: The # is needed. It's your standard programming syntax. My two cents on the date/time variable would be no. The includes are processed when * starts up, and are all grouped together. It's more of a way to keep everything clean than for a logic basis. Anyone else? Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months ok, Im a bit confused. I was refering to using includes within contexts. I've been doing things like this in extensions.conf : ;;; [local] -allow local area calling ;;; include = outgoing-pstn-local include = outgoing-iax-peer include = outgoing-operator include = outgoing-911 include = outgoing-411 include = outgoing-611 and this ; check for holiday and play special message ; include = newyears|*|*|1|jan My original question was if I could break my extension.conf file up into seperate files and include the smaller pieces back into the main file like this. extensions.conf include = /etc/asterisk/extensions.conf.outgoing.contexts include = /etc/asterisk/extensions.conf.incoming.contexts include = /path/to/some/file|hours|weekdays|monthdays|months -- .~. /V\Lance C. Arbuckle // \\ /( )\ ^'~'^ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Dialing chan_local. Was: [Asterisk-Users] return of the transfer to a busy number
Anton Yurchenko wrote: Philipp von Klitzing wrote: Hi! Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. How about - you store a temporary variable using SetVar() with the name of the callerid and the value of the initially called extension when the first contact is made - modify your dialplan so that in the event of BUSY you check if the above variable exists, and if the initial extension differs from the current one - if yes, dial a local channel, if no do nothing What could that be I have the following in my dialplan: exten = 153,1,Macro(stdexten,153,MGCP/aaln/[EMAIL PROTECTED]) exten = 161,1,Dial(Local/153,,t) but when I dial 161 I get that it is busy, dialing 153 directly works. here is except form my console: -- Executing Dial(SIP/160-43fa, Local/153||t) in new stack == Everyone is busy at this time -- Executing Macro(SIP/160-e769, stdexten|153|MGCP/aaln/[EMAIL PROTECTED]) in new stack -- Executing Dial(SIP/160-e769, MGCP/aaln/[EMAIL PROTECTED]|20|t) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: -1, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/160-e769' in macro 'stdexten' == Spawn extension (icg, s, 1) exited non-zero on 'SIP/160-e769' What am I doing wrong Can't comment on what you're doing wrong, just add an observation: the message Everyone is busy at this time seems to appear in situations when something else is wrong. Is it possible to change this message or add more informative information? Anyone knows when this message is generated? /O -- *** Olle E. Johansson, [EMAIL PROTECTED] Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net Runbovägen 10, 192 48 Sollentuna, Sweden Phone: +46 8 594 78 810, Fax: +46 8 594 78 820 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
Steven Critchfield wrote:
On Mon, 2003-12-29 at 14:54, Olle E. Johansson wrote:
Steven Critchfield wrote:
[incoming]
exten = _.*,1,answer
;exten = _.*,2,agi(timeout-lookup.agi) ; alternative
exten = _.*/some match,2,Absolutetimeout(360)
exten = _.*,2,noop
exten = _.*,3,goto(realcontext,${EXTEN},1)
Hmmm. Is that right, Steven? Does
exten = xxx/match,2
follow
exten = xxx,1
and is followed by
exten = xxx,3
If I have a /match on callid, will both priority 2 steps be executed,
or only one, the actual match?
That's something we need to document, if that's correct.
Like most things in asterisk, the best match is used. After the step is
used, it will advance the priority counter. This is how you can branch
and rejoin. The noop could have just as easily been a goto that skipped
several priority entries so as to rejoin the callerid matched portion of
the branch. And conversely, you could make non callerid matches go
through some form of rigmarole to be authenticated before allowing them
through to the rest of your IVR that you allow callerid users direct
access to.
OK, thanks.
Wasn't aware of that. Interesting feature.
I'll see if I'm able to document it so users understand. It opens for very
unreadable configurations, so, readers, please use with care... :-)
/O
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RE: [Asterisk-Users] include a file ?
include = context is for context inclusion #include filename.conf is for including other files that follow the standard config files (sip, extensions, etc) Don't let the two includes confuse you. They serve two completely different functions on two completely different levels. If you had 1000 extensions on your * box, having them all in one file would be all but impossible to administer. But if you broke those thousand extensions out by department and created separate files for each one, you could easily keep everything straight. #include accounting.conf #include support.conf #include shipping.conf Of course, this is only one of many ways you could use the #include function! Sean -Original Message- From: Lance Arbuckle [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] include a file ? Sean Cheesman wrote: The # is needed. It's your standard programming syntax. My two cents on the date/time variable would be no. The includes are processed when * starts up, and are all grouped together. It's more of a way to keep everything clean than for a logic basis. Anyone else? Brian West wrote: its #include filename.conf Does the synatx include the # at the beginning of the line ? And can this type of include be time/date dependant like the standard include ? include = filename.conf|hours|weekdays|monthdays|months ok, Im a bit confused. I was refering to using includes within contexts. I've been doing things like this in extensions.conf : ;;; [local] -allow local area calling ;;; include = outgoing-pstn-local include = outgoing-iax-peer include = outgoing-operator include = outgoing-911 include = outgoing-411 include = outgoing-611 and this ; check for holiday and play special message ; include = newyears|*|*|1|jan My original question was if I could break my extension.conf file up into seperate files and include the smaller pieces back into the main file like this. extensions.conf include = /etc/asterisk/extensions.conf.outgoing.contexts include = /etc/asterisk/extensions.conf.incoming.contexts include = /path/to/some/file|hours|weekdays|monthdays|months -- .~. /V\Lance C. Arbuckle // \\ /( )\ ^'~'^ ___ 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] CVS Closed?
Philipp von Klitzing wrote: Tried to DL using CVS this eve, and it says: Unknown host cvs.digium.com. Has Asterisk development stopped? Digium was just sold to Microsoft. I must reset my date, is it already april 1:st? exten= 20040401,1,SayKlitzingTime() /O :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
On Tue, 2003-12-30 at 16:25, Olle E. Johansson wrote: OK, thanks. Wasn't aware of that. Interesting feature. I'll see if I'm able to document it so users understand. It opens for very unreadable configurations, so, readers, please use with care... :-) Just like in programming, anytime something isn't glaringly obvious, you should comment. Some people claim you should write comments first and then clarify your comments with code(dialplan). -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Closed?
Steven Critchfield wrote: On Tue, 2003-12-30 at 12:34, David J Carter wrote: Don't say that. Does that mean that from now on we will get a voice asking if we really want to do that at every button press? heh, I can imagine it now, a call that says, A call has been received. You will now need to restart your computer. Too many calls. Please hold on while we defragment your PBX. PBX failed. Press any key to reboot. Your call is important to us. Ignore, abort, retry? Sorry, I did not start this. /O :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup Proxy Automatic Failover
Steve Dolloff wrote: I simply have 2 asterisk servers and have the clients point to a DNS SVR record for their proxy. The DNS record lists the primary and secondary with preference for the primary. This won't stop calls from being dropped if the primary goes down if you are routing them through the server, but it does ensure that calls placed while the primary is down will still go through. You could do some load management by putting multiple servers in the DNS record and use a DNS server that supports round robin responses. Please note that Asterisk itself only reads the FIRST DNS SRV record. So Asterisk can't load balance this way, when using DNS SRV for outbound calls. I would love to fix this, if I knew how. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * crash when forward voicemail message [problem solved]
Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it seemed to solve the problem I was having. There is still a little weirdness going on but the voicemail forward command is working. During a -dgc session, I get: Urgent handler -- Playing '/var/lib/asterisk/sounds/vm-received' (language 'en') Urgent handler -- Playing '/var/lib/asterisk/sounds/digits/at' (language 'en') Urgent handler -- Playing 'vm-extension' (language 'en') Urgent handler -- Playing 'vm-forwardoptions' (language 'en') Urgent handler Huh? Child handler, but nobody there? Huh? Child handler, but nobody there? Huh? Child handler, but nobody there? Huh? Child handler, but nobody there? Huh? Child handler, but nobody there? Huh? Child handler, but nobody there? -- Playing 'vm-message' (language 'en') Urgent handler -- Playing 'vm-saved' (language 'en') I'm not sure what the Child handler is or what it does or the effect it has on this kernel and the newer kernel but all seems to work of without failing as before with the newer kernel. In the trouble shooting process I upgraded my kernel to the newest Redhat release linux-20.4.20-27.9 but that had the same effect when forwarding voicemail, * shut down. If anyone can shed some light on this, maybe try forwarding voicemail on a newer kernel and let me know your success or not. Or maybe help me to understand what differences in kernels would have on Child Handlers? As it stands, I plan on using the older kernel for my implementations for now. Hope this helps. JR -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 1:57 PM To: JR Richardson Subject: RE: Re: [Asterisk-Users] * crash when forward voicemail message I don't think it's a sound card problem since the sound card is only used when you want to make it a console phone. Also it might be your system's problem since under gdb the asterisk is stack on pclose function that is supposed to wait for sendmail and close the forked process. I don't know about that too much ... but I'd try installing a diffrent kernel / distro. regards Martin On Mon, 29 Dec 2003, JR Richardson wrote: Martin, Over the weekend, I re-built the server and lost that dump file. I am on to something I think. I found that my sound card wasn't loaded properly so I finally got that loaded into the kernel. It is on-board sound and seems to work fine out side of asterisk, but when asterisk -gc is loaded I get: [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[1167272000]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_zap.so] = (Zapata Telephony) That is the only warning I get when launching. I'm thinking that while VoiceMail is playing a message and voice prompts and the forward voicemail command is executed, the sound card isn't being released properly and asterisk can't handle this in certain cases and shuts down. I ordered a new sound card (Creative Labs Ensonic ES1373 128 Voice PCI Sound Card) that seems to be working well with asterisk from what I read on the web. I should have it in a couple of days and get it loaded to see if that is it. The card I currently have is a crystal audio on a dell gxa motherboard. I received an e-mail from one of the other guru's on the list. He said that if sendmail is sending e-mail through the console then it's very unlikely that the problem is with that application. Do you think a sound card issue could be causing this voicemail problem? Thanks. BTW, I ordered another FXO/FXS combo from you guys. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] include a file ?
Lance, Parsing of configuration files is done at CLI reload or startup. That includes the #include *FILE* construct. The include statement - without the # character - includes *contexts* and this can be done at different times, since all contexts are parsed when Asterisk parses configuration files. Continue the CONTEXT path, when you fully understand contexts, it's really powerful. I understand how this can be confusing, with two different commands with the same name. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone as intercom
(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] playback in [macro-stdexten] problem
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the please hold while I try that extension
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten = 8005,1,Macro(stdexten,8005,Zap/2)
exten = 8006,1,Macro(stdexten,8006,Sip/8006)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten = s,3,Voicemail(u${ARG1}); If unavailable, send to
voicemail w/ unavail announce
exten = s,103,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
--
.~.
/V\Lance C. Arbuckle
// \\
/( )\
^'~'^
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[Asterisk-Users] TDM400P related question
Hello All, I managed to Configure TDM400P, now I can call Analog Phones connected to TDM400P from SIP Phones (CISCO and SNOM). But when I try to dial any number from Analog Phones I get following message -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' My extension.conf has following lines to deal with the Analog Phones exten =410,1,Dial(Zap/1) exten =510,1,Dial,Zap/2 exten =610,1,Dial,Zap/3 exten =710,1,Dial(Zap/4) Am I missing something. Best regards Tony
Re: [Asterisk-Users] * crash when forward voicemail message [problem solved]
Did you try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 Best regards, On Tue, 2003-12-30 at 20:07, JR Richardson wrote: Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it seemed to solve the problem I was having. There is still a little weirdness going on but the voicemail forward command is working. During a -dgc session, I ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test
___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: +AFs-Asterisk-Users+AF0- RE: +AFs-Asterisk-Users+AF0- Multi-line, multi-registration phones
Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen. +AFs-macro-normal-dial+AF0- exten +AD0APg- s,1,ChanIsAvail(+ACQAew-ARG1+AH0-) +ADs- Determine what line is available in order it was sent. exten +AD0APg- s,2,Dial(+ACQAew-AVAILCHAN+AH0-/+ACQAew-MACRO+AF8-EXTEN+AH0-) +ADs- AVAILCHAN gets set from ChanIsAvail and you use MACRO+AF8-EXTEN instead of EXTEN. +AFs-line-1-outbound+AF0- +ADs- Local call here in California +AF8-NXX exten +AD0APg- +AF8-NXX,1,macro(normal-dial,Zap/1+ACY-Zap/2) +ADs- Notice for line 1 I put Zap/1 before Zap/2 +AFs-line-2-outbound+AF0- +ADs- Local call here in California +AF8-NXX exten +AD0APg- +AF8-NXX,1,macro(normal-dial,Zap/2+ACY-Zap/1) +ADs- Notice for line 2 I put Zap/2 before Zap/1 Now without it looking way to complicated I shortened my macro to make it easy to read here but I have it do a lot more under my setup. So if you add more to your macro then the single line still runs all of the things in the macro. Here is another example of how I use macro's for my extensions. That way I can setup voicemail and anything else I need with a 1 line entry after I have built my macro which will make your extensions.conf file smaller and also allow you to make 1 change instead of many. +AFs-macro-extensions-out+AF0- exten +AD0APg- s,1,Answer +ADs- Answers the call exten +AD0APg- s,2,AGI(MisterHouse.agi,+ACI-DTMF: +ACQAew-MACRO+AF8-EXTEN+AH0AIg-) +ADs- Outbound call logging to MisterHouse home automation software exten +AD0APg- s,3,Dial(+ACQAew-ARG1+AH0-,+ACQAew-ARG2+AH0-) +ADs- Dial for ?? seconds. exten +AD0APg- s,4,Voicemail2(u+ACQAew-ARG3+AH0-) +ADs- If on phone or channel OUTOFORDER go to busy voicemail exten +AD0APg- s,5,Hangup +ADs- Hangup the line after the voicemail. exten +AD0APg- s,104,Voicemail2(b+ACQAew-ARG3+AH0-) +ADs- If not available then go to unavailable voicemail. exten +AD0APg- s,105,Hangup +ADs- Hangup the line after the voicemail. exten +AD0APg- 2000,1,macro(extensions-out,Sip/2000,20,2000) +ADs- Extension with voicemail rings for 20 seconds before going to vm 2000. exten +AD0APg- 2001,1,macro(extensions-out,Sip/2001,180,) +ADs- Extension with a answer machine rings for 180 seconds with invalid vm box. etc... So you see you can save a great amount of time with macros and after you get the hang of them they will cut down on your extensions.conf file size and make your life a lot easier. Robert Mann - Original Message - From: +ACI-Sean Garland+ACI- +ADw-sean+AEA-siskiyoutech.com+AD4- To: +ADw-asterisk-users+AEA-lists.digium.com+AD4- Sent: Tuesday, December 30, 2003 3:00 PM Subject: +AFs-Asterisk-Users+AF0- RE: +AFs-Asterisk-Users+AF0- Multi-line, multi-registration phones Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my production phone system soon. Thanks Sean Garland sean+AEA-siskiyoutech.com -Original Message- From: Steven Critchfield +AFs-mailto:critch+AEA-basesys.com+AF0- Sent: Tuesday, December 30, 2003 1:19 PM To: asterisk-users+AEA-lists.digium.com Subject: Re: +AFs-Asterisk-Users+AF0- Multi-line, multi-registration phones On Tue, 2003-12-30 at 14:29, Sean Garland wrote: +AD4- I have hard phones that are capable of handling three calls at once. +AD4- That is setup (apparently) through multiple registrations. My +AD4- question is has anyone done this and what is the proper way of doing +AD4- it? Do I have to setup (for 2 phones that have three lines) 6 +AD4- sections in my sip.conf and setup 6 extensions to handle the +AD4- registrations? +AD4- +AD4- Also, if I found by searching the web sample code for making both sip +AD4- extensions ring when a call comes in, but what if I had 100 +AD4- extensions? Seems like the string would get pretty long, is there a +AD4- way to put all extensions in a single group and ring the group? +AD4- +AD4- All kinda is the same question. But thanks for the answer anyway... This is where variables come in handy, and also macros. For example, I just put several new variables into my extensions.conf file to deal with the changing nature of our dialplan. I defined a variable for each user that we have, and I created a ALL variable that strings all the users together. You could also make variables for all the users in certain departments, and then your ALL variable could then just include your
Re: [Asterisk-Users] Re: * crash when forward voicemail message [problem solved]
RedHat 9 and Fedora kernels have a new feature (not present in kernel.org): Native Posix Threads This brings all sort of problems to diferente applications. To override this new feature, you have to start your affected programs with that enviroment variable set. On Tue, 2003-12-30 at 21:43, JR Richardson wrote: -Original Message- No I didn't, I don't have a clue what that is or does. Please explain, I'll try it and let you know. Did you try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I wanna buy a new X100P
Sure, head to : http://store.yahoo.com/asteriskpbx/wildcardx100p.html -d At 12:25 PM 12/30/2003, you wrote: I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? 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] Does Asterisk support legacy Dialogic products?
- Original Message - From: Patrick Wong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:20 PM Subject: Re: [Asterisk-Users] Does Asterisk support legacy Dialogic products? Steve Underwood wrote: That card's haardware is not capable of providing any VoIP functionality. It is not full duplex. The newer JCT cards can be used, but they still don't work that well, due to card limitations. snip I searched the archived mail messages and found that drivers for Dialogic are commercially available for $15 per channel? By one channel means a single loop start Telco interface? Dialogic apparently licenses the software as well as sells hardware. You have to buy one of their cards and then buy the channel license. Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P BAD SOUND with NEW ASTERISK
Hi, I move the * on a new DELL server and I get the latest version of Asterisk with CVS. I have 3 FXO cards, X100P and the sound before was fine. With the new version of Asterisk and on new Dell server the sound is SO BAD! Some suggestions are welcome. Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Thanks for the reply Tim - I was beginning to think nobody used this stuff ;-) As you can tell, I'm a relative newcomer to ADSI - I'm really not sure what to expect once the phone gets programmed, but I would not expect to see (PLEASE PROGRAM ME) still, and I would have hoped it would not have broken voicemail so readily. I'm not using a channel bank at all ... I have a very simple setup using two x100p FXO cards and one TDM400P FXS card. As I mentioned below it does appear that SOMETHING was loaded into the phone, and it does appear to at least TRY to use ADSI when accessing voicemail. It's odd ... it's like everything worked but I'm left saying ... okay, now what? The phone isn't incredibly functional at this point - even if I do go into the services menu and select 'Asterisk PBX' this selection only persists until I use the phone once. Also, there aren't soft keys for anything useful like transferring a call ... how WOULD one do that with this phone anyway? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 3:48 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the first try with current CVS. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 10:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update. As I mentioned in my first post (below), I had managed to get the phone to accept a download from asterisk, but it still said PLEASE PROGRAM ME after the download was completed. After further investigation it does appear I downloaded SOMETHING to the phone, because if I select the services button on the front of the phone, I get into a menu that says: Services Asterisk PBX Asterisk slot 2 available available If I select the first one in the list, the phone does change from PLEASE PROGRAM ME to ** Asterisk PBX** and there's a VMail softkey!! However, if I pick up the handset and replace it, I'm back to PLEASE PROGRAM ME again. Is this normal? My second problem is that when I dial voicemail from this handset now, which is ADSI-enabled, I see the following message on the text screen: Comedian Mail (C) 2002 LSS, Inc. Downloading Scripts I don't see any activity on the console in terms of logging, and I don't see any way to elevate ADSI logging. Within less than a second, I see: Comedian Mail download refused. Services is full At this point the voicemail's welcome script plays and the display changes to: Comedian Mail (C) 2002 LSS, Inc. Load Cancelled ADSI Unavailable at this point voicemail's works like it used to before ADSI. Is any of this ringing any bells with anyone? Any tips appreciated ... I've looked around but I'm really not finding any
Re: [Asterisk-Users] Grandstream Early Dial
On Thu, 18 Dec 2003, Aaron Martin wrote: I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. As an example, my extension is 100 and let's say my password is 1234. Here is what * captures: -- Executing VoiceMailMain(SIP/damin-3099, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[5126]: File chan_sip.c, Line 4667 (handle_response): Peer 'damin' is now REACHABLE! -- Playing 'vm-password' (language 'en') -- Incorrect password '111223' for user '11000' (context = any) -- Playing 'vm-incorrect' (language 'en') Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will yield better results. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote: On Thu, 18 Dec 2003, Aaron Martin wrote: I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. As an example, my extension is 100 and let's say my password is 1234. Here is what * captures: -- Executing VoiceMailMain(SIP/damin-3099, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[5126]: File chan_sip.c, Line 4667 (handle_response): Peer 'damin' is now REACHABLE! -- Playing 'vm-password' (language 'en') -- Incorrect password '111223' for user '11000' (context = any) -- Playing 'vm-incorrect' (language 'en') Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will yield better results. What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
On Thu, 18 Dec 2003, Aaron Martin wrote: I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. Are you using SIP Info for DTMF? It's the only thing that reliably works with GS phones. Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will yield better results. It almost certainly will. (Some would say that two tin cans and a string would work better than Grandstream phones, but I digress...) Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accountcodes
I'm trying to use accountcodes, but experiencing inconsistant
results. I have two * servers, one which appears to be working as
expected and one not. I would like to prepend the device's accountcode
to the dialed number. The sip1 server does not seem to have the
${ACCOUNTCODE} variable set when reading the extensions.conf, but sip2
server does.
What troubleshooting or trace information can I review to determine the
cause?
sip server #1
sip.conf
[2105]
type=friend
username=2105
secret=105
canreinvite=no
host=dynamic
;context=from-sip
mailbox=2105
nat=yes
accountcode=2
extensions.conf
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
sip server #2
sip.conf
[2108]
type=friend
username=2108
secret=108
canreinvite=no
host=dynamic
context=from-sip
mailbox=2108
nat=yes
callerid=9549778081
accountcode=12345
extensions.conf
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
Thanks,
--
David A. Lauer
Tristar Communications
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[Asterisk-Users] Consultancy on Asterisk !!
hi all i need constulancy service for depolying Asterik below are my requirement in point forms All Single Line telephone are attached to an autodialer which is program to dial the access number and store digits dial by the user. Ie. When the user picks up the phone and dials 001214xxx, the dialer knows that it is an overseas call. 00 is our overseas prefix number. One point to note that, the moment dialer detects is an overseas or long distance call. It will start dialing the local access while user is still keying in the remaining digit of the overseas number. The dialer will dial the local gateway access number ie 123-45678 with the remaining digit stored in the RAM. The destination gateway will acknowledge the call by sending a DTMF A tone for about ½ second. Autodialer acknowledge the DTMF A tone and generate the overseas number stored in the dialer RAM. Once the overseas digits are send, the dialer cuts of itself thus stay on the line Now, what I need asterik to do is to stand in the middle between the dialer and destination gateway. I would want asterik to be item 3a. instead of dialing direct to other gateways. The dialer will now dial asterik access number. Asterik will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ½ second to enable the dialer to send the whole overseas digit. Assume the caller is not in database, asterik could give user a busy tone, IVR or just leave it and sends out a DTMF A tone anyway. Once the overseas digit are sent from dialer to asterik, asterik will then decide which telco/carrier/Voip to send the traffic to using LCR. Please note that we need to assign at least 5-10 telco/carrier/Voip access number for backup purposes. Once the least cost destination is selected by asterik, asterik will pick up the PRI line and dial a local access number and waits for a DTMF A tone. Once the A tone is heard from telco/carrier/Voip, it will send the overseas digit which was sent by the dialer earlier on. It would be a bonus if asterik could sends out a musical tone or IVR while connecting to other telco to advice user that the call is connecting, else it would be dead air from there on. The whole process takes less than 5 seconds while the user stays on the line for this whole thing to happen. Of course, asterik MUST be able to give me CDR in text format with callerID, destination number, date/ time call , and duration Can that be done ? __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A Head Check
Hello, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small groups of individuals. As fate would have it, the service will be provided in a building where we have a major Pop, with a DS-3 worth of ISDN PRI circuits, 345 megs of upstream bandwidth and diesel generator backup. We are providing everything for the solution, from the initial wiring to the ongoing maintenance of the PBX and Internet service. I have arranged for a single PRI to be broken out of our DS-3 w/ 100 inbound DID numbers assigned to it and have PICd it to the LD provider of our choice. I intend to plug this PRI into an Asterisk server w/ a Digium TE410P card, and deploy SNOM 200 IP phones to the desktops. We will be using a RedHawk power-injector system to provide power to the phones. Now.. This is our first deployment of Asterisk, and I need a head check here. Am I making the right decision? :) Sepcifically... 1. Are the SNOM 200 IP phones a good choice for standard users? Or should I consider Cisco? Price of the phone is not the important thing.. What is important is ease of use with minimal training and reliability! 2. Does anyone have reccomendations for a solid motherboard to use as the basis for the Asterisk server? Again, reliability and stability are the important issues here. I'm looking for a Dual CPU board (Athlon MP or P4) that will work flawlessly with the TE410P. I've used the Tyan Tiger MPX (2466) http://www.tyan.com/products/html/tigermpx.html with Dual MP processors with incredible success in the future. I'm considering building the box on that platform. 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! 4. I have built some cost for support from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? I'm just trying to make sure that I cover all the bases. This is got to be a bulletproof solution, and I'm departing from my comfort level with Altigen to give Asterisk a run for the money. We've got TONS of Linux experience here, and comfort with customizing code, so I am happy with what Asterisk gives me.. What else should I be worried about? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
