Re: [Asterisk-Users] anyone know about this company? www.blue-wireless.net
I've got a customer who is thinking of installing this www.blue-wireless.net Know nothing about the software but if it's anything like their website then I'd be wary. Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] DID from an analog phone
how can i dial DID to asterisk from analog telephone? Zapata does not report a DID for the incoming calls - as I mentioned in a recent post about seperating incoming calls on a TDM02B (see the archive). Effectively you will need to point each port to the appropriate dialplan context in your zapata.conf. Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Seperate Incoming calls on TDM02?
Yeah, in your zapata.conf just give each channel a different context setting. It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to use the zapata_custom.conf file, instead. You also need to use the extensions_custom.conf file, too, though there might be a better way I don't know about. I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Hatton zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf zapata_additional.conf is empty zapata-auto.conf: ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; channel 1, WCTDM, inactive. ; channel 2, WCTDM, inactive. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from_pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from_pstn group=0 channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Seperate Incoming calls on TDM02?
I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Found the source of the problem and it wasn't in the config files, rather in the way that Asterisk reloads when issued that command - it does NOT rebuild the Zapata channel table. I had to perform a complete restart of Asterisk to get the changes (which I made in my zapata-auto.conf file) into the system. After I did that everything started running right as rain! Thanks for the assistance, hopefully this will make it where people will see it in the future. Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Fwd: [Asterisk-Users] Seperate Incoming calls on TDM02?
I have 4 FXO ports, 2 on one number and 2 on another and want to have different incoming rules\IVR depending upon channel called. Is it as simple as changing the contexts in the zapata.conf or is there more to it. Here is what my experience was. Understand when reading it that I am running [EMAIL PROTECTED] version 1.5. I have 6 VoIP DIDs, two point to one auto attendant (aa_1 in my dialplan as created by AAH), two point to a second IVR (aa_2 in my dialplan as created by AAH) and the last two point to a direct extension. The From PSTN rules point all incoming calls to aa_1. AAH includes a context= in the main zapata.conf file. The rules for zapata are odd, basically you define a set of parameters and load it into a channel. If you don't redefine them and create another channel the parameters already set are included in the second channel. Therefore you can set the main setting parameters once and then change things as needed. The other thing that gets created by AAH is a file called zapata-auto.conf; this file is created by the genzaptel script and automatically sets your Zaptel channels up. If you open this script you'll see each of your incoming ports with a parameter setting of context=from_pstn. What I did (against the suggestion of the file) was change the context in the zapata-auto.conf file to point to the correct IVR. The point that I missed originally that I figured out was that the reload command in asterisk does NOT redefine the channels. You have to execute a restart command, either restart now or restart when convenient. That will reread the zapata.conf file and redefine the zaptel channels. Hope this helps, Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Seperate Incoming calls on TDM02?
I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file to do this as it is my understanding that the control can be taken care of there. My system is running [EMAIL PROTECTED] 1.5. Help! Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is Health and Welfare with Logistics being the second to it. You might be interested to take a listen to the latest ARRL News - they give a count of Priority traffic messages passed for Katrina... http://www.arrl.org/arrlletter/audio/ The site is ARRL and it's their ARRL Letter feed to be presented on repeaters. The ARES response to Katrina articles have the info I'm referring to. Sorry for the OT addition to the thread but I find it worth mentioning. Also, for my two cents I'll toss in that the first thing I thought of when someone mentioned using Asterisk with Ham was to get a Laptop with a WiFi connection, Asterisk and a radio interface on scene to provide comm links. 73 de NY5I Hatton Humphrey ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Where did my DID's go??
Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. Hatton ___ 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
Re: [Asterisk-Users] Where did my DID's go??
Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. I've sent them a message and still no action - is there anything I can do in the interim other than deflect complaints from family, friends and system users?? Hatton ___ 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] Where did my DID's go??
Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service. Well, I guess the porting request got approved... but not completed. Now when I call any of my DID's I get a message that says: Your Call Cannot Be Completed as Dialed ISUP0056115 Announcement 24 Packet8 is claiming ignorance and says that the numbers must have ported to the new carrier. The new carrier has checked their incoming port queue and says they don't see the numbers listed. In the meantime I've got 3 phone numbers that go no where and office mates that are boiling mad because they're loosing business because of lost calls! This has been going on since noon yesterday EST and I have yet to get an acceptable answer from either Packet8 or the new carrier. My guess is that the third party that transfers the numbers from old to new carrier has the numbers somewhere in limbo but as of this point I have no frelling clue where that might be... is there a way to translate that message I'm getting to find out what nimrod company I need to call and get my DID's pointed to the correct account??? Thanks! Hatton ___ 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] TDD over Asterisk
I'm working on a setup for a company that needs to be able to handle TDD/TTY calls for the hearing impaired. Is there anything special that I need to consider for this? We're looking at an Asterisk solution for them, the tones need to be stored in the IVR as well as coming across extensions. Any thoughts? Hatton ___ 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
Re: [Asterisk-Users] Full T38 sip Faxing now Available
On 7/29/05, Michael D Schelin [EMAIL PROTECTED] wrote: We are a VoIP service provider and now have full Sip T38 capabilities. This is not a secret. There are no tricks. I'm just alerting the community that if they need to fax reliability through VoIP, we now support T38 through our SS7 network at the same low rates we sell voice. I have been reading for months about fax problems in this forum. This is why I post. I'd like some info as well Hatton ___ 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
Re: [Asterisk-Users] Stupid hold music
Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? You should be flogged pubically for bringing up this subject - that last space people song almost made me wash out my ears with sulphuric acid!!! 73 de NY5I Hatton ___ 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
Re: [Asterisk-Users] Howto get streaming mp3 at an extension?
stream = /var/lib/asterisk/stream,http://sourcepfstream.com:8001/ Then add an externsion number to extensions.conf that uses the stream variable to play the hold music. There's quite a bit about this in the wiki. The stream I'm trying to listen to has a filename in the URL ... http://.:12030/listen.pls Do I need to do anything different? Hatton ___ 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
Re: [Asterisk-Users] rxfax not answering
rxfax doesnt work with voip, you need something like NVFaxDetect from Newman Telecom to detect the incoming fax. Picking up an old thread here but I have to ask this - I have NVFaxDetect or NVBackgroundDetect working to *detect* the incoming fax call; what I can't get working is the actual reception of the fax. Any suggestions? Hatton ___ 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
Re: [Asterisk-Users] Colored asterisk -R?
I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? It works fine for me to run astersik -rc Hatton ___ 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
Re: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing
You would be better using extensions_custom only because of the fact that when you restart ampportal, it will overwrite extensions_additional with what ever it has stored in the Database. I've actually taken to adding the code that I build onto what AMP generates into the database. For example I had to add in a default option for a Digital Receptionist I used the phpMyAdmin that's installed with [EMAIL PROTECTED] and inserted the data into the extensions table. Hatton ___ 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] ASDI Programming through an ATA/SIP device?
As I've said before, I have a number of phones that I've converted from Packet8's Virtual Office to work with my Asterisk stetup. They are basically Leadtek 85XX SIP devices with Astra 390 phones connected. I've been investigating reprogramming the Astras to say something other than Virtual Office powered by Packet8 but when I try ADSIProg I get something about CPE not being available. I'm afraid to say but I think that I'm going to have to get a Zaptel device to get the phones programmed (a single port that I can plug the phones into and then have them call the programming extension, maybe) unless there's something that I'm missing --- can I program an ADSI device using a SIP interface? Thanks! Hatto n ___ 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
Re: [Asterisk-Users] handle wrong extensions in Dialplam
i am trying to make a dial plan that can handle any wrong extensions dialled from the local sip phone for example so that if i dialled the right extension it rings but if i dialled wrong or existing extension it redirect him to the Main menu for example? You might see if you can put the i extension to work for you in your local dialplan. Hatton ___ 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] Always forward an extension?
Here's something I haven't been able to discover as of yet - I need to set up a direct link from my Asterisk box to an external line... basically I need to be able to pick up an internal extension and have it call a local phone number. This is call forwarding, I know - the question that I have is how do I set it up so that the extension always forwards. There will never be a client logging in to it. Thoughts, experiences, ideas? Thanks! Hatton ___ 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
Re: [Asterisk-Users] problem with pf and asterisk
I took the info from here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules and ended up with the following in my pf.conf: rdr on $ext_if proto tcp from any to ($ext_if) port 5060 - $dmz_ip port 5060 rdr on $ext_if proto udp from any to ($ext_if) port 5060 - $dmz_ip port 5060 rdr on $ext_if proto udp from any to ($ext_if) port 4569 - $dmz_ip port 4569 rdr on $ext_if proto udp from any to ($ext_if) port 5036 - $dmz_ip port 5036 rdr on $ext_if proto udp from any to ($ext_if) port :20001 - $dmz_ip port :20001 rdr on $ext_if proto udp from any to ($ext_if) port 2727 - $dmz_ip port 2727 I also have the following lines in there: pass out on $ext_if all keep state # pass incoming dmz traffic pass in on $ext_if proto tcp from any to $dmz_ip keep state pass in on $ext_if proto udp from any to $dmz_ip keep state HTH Hatton ___ 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] Re: Asterisk@Home connecting through firewall/router
The phones register fine on the PBX. I just can't get Asterisk to authenticate out to the IAX server. I've opened up all of the ports listed on voip-info.org and even have gone so far as to put the box in as the DMZ on the DLink. Well okay, I'm not 100% sure what happened but I went into the Asterisk machine, ran 'setup' (thank you for the RedHat friendly base) and made sure the network was configured and the firewall was disabled and then restarted both the FreeBSD router as well as the Asterisk box and *poof* things started working! Now to get the IVR's set up and working... should be a fun trick! Hatton ___ 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] Asterisk@Home connecting through firewall/router
I ditched the idea of using Asterisk straight through my router, there was too much to set up in too little time for me. I've found a spare machine and installed [EMAIL PROTECTED] on it. Things run smoothly except for connecting to my IAX provider; It doesn't even look like packets are going out at all. Here's my config: ANY Cable -- Firewall/Router -- Switch ---[EMAIL PROTECTED] Box |-Other PC's \-SIP DTA's for internal phones (The ANY refers to both a D-Link router appliance and my FreeBSD router) The phones register fine on the PBX. I just can't get Asterisk to authenticate out to the IAX server. I've opened up all of the ports listed on voip-info.org and even have gone so far as to put the box in as the DMZ on the DLink. At least on the FreeBSD install of * I could get connection. Is there something on [EMAIL PROTECTED] (latest release downloaded this morning) that I'm missing here? Thanks! Hatton ___ 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
Re: [Asterisk-Users] A Few Questions
On 6/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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] A Few Questions
I've been watching the mailing list for a few days, have done some archive searching and still have a handfull of questions (I've looked for most of these on voip-info.org and a slew of other asterisk related sites). I'm going to throw the ones that are foremost in this email and will add more as the need arises. Before I do so, let me explain my setup. I have Asterisk running on a FreeBSD machine that is also my router/firewall and MySQL server. It is running fine and I've gotten it working with FWD and will be testing a direct IAX server in the next few days. I'm migrating from a Packet8 Virtual Office setup and have managed to get their DTA-310 working on my installation. Here are my questions.- 1. Does anyone have suggestions for license-friendly MOH sources? Same for reworks of the voicemail and autoattendant prompting? 2. Help! I got the MWI light on the phone (an Astra powered by the DTA-310) but now it won't go off. 3. Is there any way to have asterisk take a phone back to a plain dialtone instead of a fast busy when a call ends? 4. Even though I've got the basics working I keep wondering what else is available. For example I see on the * website that things like transferring and web access to voicemail are available but I don't even know where to begin looking for that stuff. Where are the guides that I'm missing for all of the different configuration issues? 5. All my other boxes are Windows machines - can someone recommend a config tool that I can run on Windows to help me get everything straightened out? 6. What hardware is really needed to bring in a copper pair? I have a single CO line that we're using for faxes and I'd like to be able to include it in our outgoing call system for 911 capabilities. At the same time I don't want to throw down a bill for a card. Thanks! Hatton ___ 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
Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server
I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. I'm new here but have to go ahead and throw in a reply - I'm doing something very similar to this but in a different way... I have a FreeBSD box that is already set up to be my router/firewall/IDS/MySQL server. I'm working on getting it set up with Asterisk... just got it working with FWD and a Packet8 DTA based phone. I have to admit, there's a lot of stuff to wrap your head around with this stuff! Hatton ___ 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
