Re: [asterisk-users] Skype for Asterisk???
I would have happily bought 20 channels at $10/channel, but at most will be buying only a single channel now :\ Pascal Bruno wrote: Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anonymous Michigan Calls, Skype/Other
My skype number appears to belong to a pool given to Level 3 Communications, and it was out of the same 1000 block as my Google Voice number. You could block all Level 3 numbers for your area, but it would run the risk of blocking legitimate customers from calling you. Casey Boone Jared Armstrong wrote: Does anyone have a list of Skype or other Anonymous VOIP end point phone numbers that I can use to block unsolicited calls with? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Dave Platt wrote: OpenVPN doesn't suffer from this problem. Although it's SSL-based (and one might think it does everything through SSL-over-TCP), it actually sends the VPN traffic via UDP... it uses TCP only for the negotiation and administrative aspects of setting up the VPN connection. UDP is the default, but OpenVPN can be configured for TCP as well ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
just for a test, run service iptables stop as root on the asterisk server and then reboot your phones. after that, try again and see if the phones are making communications with asterisk. you can turn the firewall back on with service iptables start jonas kellens wrote: Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /bindport=5060/ /bindaddr = 0.0.0.0/ /[BT201]/ /type=friend/ /context=intern/ /host=192.168.4.210/ /secret=testpaswoord/ /[GXP1200]/ /type=friend/ /context=intern/ /host=192.168.4.211/ /secret=testpaswoord/ extensions.conf : /[r...@asterisk asterisk]# cat extensions.conf/ /[intern]/ /exten = 210,1,Dial(SIP/BT201)/ /exten = 211,1,Dial(SIP/GXP1200)/ Asterisk CLI shows me : /asterisk*CLI sip reload/ /Reloading SIP/ / == Parsing '/etc/asterisk/sip.conf': Found/ / == Parsing '/etc/asterisk/users.conf': Found/ / == Parsing '/etc/asterisk/sip_notify.conf': Found/ /asterisk*CLI sip show peers/ /Name/username HostDyn Nat ACL Port Status / /GXP1200192.168.4.211 5060 Unmonitored / /BT201 192.168.4.210 5060 Unmonitored / /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]/ /asterisk*CLI dialplan show intern/ /[ Context 'intern' created by 'pbx_config' ]/ / '210' = 1. Dial(SIP/BT201) [pbx_config]/ / '211' = 1. Dial(SIP/GXP1200) [pbx_config]/ I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Special Information Tones
GrandCentral/Google Voice does just this, although I have no idea what they use for a back end to make it happen. When someone calls your GC/GV number, it forwards out to a list of numbers you have given the service. You can choose to answer the call, send it on to voicemail, or a couple of other things by hitting 1-5 after you answer. Cary Fitch wrote: From a cell user level perspective... The cell companies are doing it like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give 4-10 rings of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but beyond that it is time to send it to VM, charge for the call :-), and move on. Ideally, a find me call forwarding system should have a real person identifier and local voice mail. Real person means that all called external numbers should not be assumed to be answered until they send back a DTMF tone. Something like a Background announcement with some silence, waiting for DTMF. It could be a Boing or You have a forwarded call, press any key to accept the call Then the call should be cut through to that extension. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype update
Enjoyed the podcast :) Does anyone have any idea what the pricing structure will be for this? are we talking $10/channel? $100/channel? Does this log into the Skype network as multiple users? One global user for the business as a whole? Do I have to have 1 user login per inbound channel? What I am hoping to be able to do with this is allow for 10-15 simultaneous inbound from Skype calls, no interest at first for receiving nor making PSTN calls via Skype. Casey Boone Tim Panton wrote: On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype update
Enjoyed the podcast :) Does anyone have any idea what the pricing structure will be for this? are we talking $10/channel? $100/channel? Does this log into the Skype network as multiple users? One global user for the business as a whole? Do I have to have 1 user login per inbound channel? What I am hoping to be able to do with this is allow for 10-15 simultaneous inbound from Skype calls, no interest at first for receiving nor making PSTN calls via Skype. Casey Boone Tim Panton wrote: On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype beta news ?
I am curious as to if there are any updates on this? Olivier wrote: Hi, Has anyone any return to share about Skype-Digium beta program ? I would be very curious to know how things are going on this. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
guys can we take the flame fest off list please? kthx Douglas Garstang wrote: -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: That's funny. I remember asking a question, and I remember you immediately attacking my intelligence, and now I'm suddenly a troll. I can comprehend that Nufone is not the operation you are involved with. However, it's the first time you've stated that, so if you think I should have known that already, then you've either a) lost touch with reality or b) think your some big shot and I should know who you are by name. You made an obvious assumption by looking at my email address, without bothering to consider any other operations I may be involved in or have assisted in development and deployment. I could care less who you think I am, really and who knows, I may have no clue what reality is - But who really does? I see you haven't addressed the specifics of my reply to you. I don't see any specific questions, only statements that prove you haven't fully comprehended what you have gotten yourself into. What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster and so on while ALSO letting customers maintain all this via a web interface? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] finding mac addresses
as long as they are in the same network segment as the asterisk server you can use arp man arp mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config Revision Control
you could always do a subversion checkout to a temp path and then do search/replace courtesy a perl or a sed script (ie, replace something like BINADDR with the address to bind to on that box). after that rsync/cp/mv/whatever into /etc/asterisk just a thought Casey Boone ShawneeLink Corporation Douglas Garstang wrote: But what if all the files are not the same? What if the binaddr is different in sip.conf on each server, or what about DUNDi? That's completely different. Do you have to go to each box one by one, check the file out, edit it, and check it back in again? I'm trying to find a way to avoid that. -Original Message- *From:* Bruce Reeves [mailto:[EMAIL PROTECTED] *Sent:* Friday, June 02, 2006 3:55 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Config Revision Control If all 3 servers are the same then no. I import to the svn server the check out the files on each server. I f I change a file on server A I can then commit the change to the repository, on the central server, and then do a svn update on the other 2. On 6/2/06, *Douglas Garstang* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Bruce, But, if you have three servers that function the same, don't you have to check the file out three times and check it back in three times? Doug. -Original Message- *From:* Bruce Reeves [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *Sent:* Friday, June 02, 2006 3:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Config Revision Control I use subversion on a central server and then store each server that is different. The purpose behind it for me was 2 fold, first I have a backup of my configs centeralized and I can roll-back any changes. Second, I can checkout a servers files on a different machine to edit them if I want and check them back when finished. What I meant by file-level is if I edit sip.conf and check it in then the whole svn goes to a new version, not just that file. We use a M$ product that has version control at the file level, so for each file in the library there is a version history. -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] clipcomm versus sipura/linksys
I am wondering if anyone has any experience with both clipcomm and sipura ip phones, i am looking to get a test unit of one or the other and cannot decide which route to go. i am looking at probably a sipura 941 or the clipcomm cp101. the clipcomm is quite a bit cheaper but i havent noticed many people commenting on it, while i have seen several people speaking positive of the sipura 9xx line. the clipcomm includes an fxo port, which is nice and would be appreciated but not required for my testing purposes. i have a couple of grandstreams at my disposal but honestly i dont much care for them, half the grandstream stuff i have quits responding to the web interface after 10 seconds or so of being turned on. im sure it is a firmware issue, but to me it is an issue that should have never made it out the door. consequently i have been looking at other vendors. so anyone have anything to offer one way or the other for clipcomm and sipura? anyone tried them both out? im leaning towards the clipcomm at the moment just because of price and the included fxo. Casey Boone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: testing email routing
please ignore this is a test email, i am testing email routing Casey Boone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Resolving timing issues with dual PRIs in a TE411P
you could try setting the * box to pull timing from each pri connected to it and set the nortel to be a master for that circuit and see if that helps any Casey Anthony Rodgers wrote: Greetings, everyone, and Happy New Year! I have a question relating to running two PRIs into a single TE411P. We have been experiencing echo, noisy MOH, poor audio call quality and so forth that started at around the time we introduced the second PRI into the equation. Here is our zaptel.conf: span=1,2,0,esf,b8zs bchan=1-23 dchan=24 #clear=1-24 loadzone = us defaultzone=us span=2,1,0,esf,b8zs bchan=25-27 dchan=48 #clear=1-24 loadzone = us defaultzone=us And here is our zapata.conf: [trunkgroups] [channels] context=incoming usecallingpres=yes echocancel=128 callerid=asreceived usecallerid=yes echocancelwhenbridged=yes echotraining=500 ; Nortel PBX rxgain=2.0 txgain=-17.0 group=1 callgroup=1 pickupgroup=1 switchtype=national signalling=pri_net facilityenable=yes channel = 1-23 ; Allstream PRI rxgain=-2.0 txgain=-17.0 group=2 callgroup=2 pickupgroup=2 switchtype=dms100 signalling=pri_cpe facilityenable=no channel = 25-27 musiconhold=default We are master to the Nortel PBX, which gets its clocking from Telus (a telco), and slave to the Allstream (another telco), from where the Asterisk gets its clocking. We have been through the IRQ/APIC/hyperthreading thing already, and our gain levels were set using ztmonitor. zttest gives Best: 100.00 -- Worst: 99.987793 -- Average: 99.998923 after 885 passes. We have another, identical server that plays MOH with no problems without a PRI attached (presumably using internal clocking), and that same server and another similar server had no issues for months with only one PRI attached (one had the Nortel and one had the Allstream, with an IAX trunk between them). Is it possible that the two PRIs have timing differences between them that could cause problems? If so, how best could we resolve them? Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
look into linux advanced routing and traffic control lartc Casey Boone Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem Connections to PPP Server
I think that timing issues will kill you, but if it were going to work you would want to use ulaw all the way around as your codec. a better option would be to use more traditional terminal server/remote access server type hardware off of an actual copper pstn line. you can pick up terminal servers off of ebay if cost is an issue, such as lucent max 4048s and cisco 5400s. voip just is not a good way to carry a modem call Casey Boone Denis Vella wrote: Hi, I'm trying to use modems with Asterisk+VoIP Gateways in an attempt at providing an Internet service. Home_PC--Modem--PSTN--VoIP_Gateway_FXO--Ethernet--Asterisk--Ethernet--VoIP_Gateway_FXS--Modem--PPP_Server--Internet I've been trying to use G711u and G711a codecs on the VoIP Gateways but, so far, no joy. Has anyone got this to work? Any pointers to setting this up? Thanks, Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited ___ --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 ___ --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] minor(? ) Grandstream phone issue
i am having a problem with grandstream as well, mainly in the web interface to configure the units. it isnt helped by the fact that unless they fixed it today their website is broken on the support page where you can download updated firmware. i have been in contact with their tech support department and have been given some beta firmware to test with so hopefully it will resolve the web interface issues i have (the interface just stops working after a random amount of time, except on one gxp2000 in which it works for about 15 seconds after the unit starts up, then quits) i havent had bad voice quality i have to say at least. that part has seemingly worked rather well Casey Boone Ade Agbero wrote: I have had numerous problems with Grandstream HT-386 new and old firmware, my convidence in Grandstream is at a very low point right now. I wish you luck, Ade. */Bob Weber [EMAIL PROTECTED]/* wrote: I hate to bother the list with this potentially minor issue but I just wonder if it's a symtom of some other problem. Every time I make a call the BT-102, with the latest firmware, she just keeps the LED display lit and the timer counting after hangup. I check the CLI and the hangup is being executed, I certainly was concerned it might be keeping the line open but that doesn't seem to be the case. I thought that she should go back to the date/time display but that doesn't happen. I don't have silence suppresion on and I've gone over the other configs compared with what's on voip-info many time. One other thing, there isn't a dial tone when I pick it up. It seems to 'work'; both inbound and outbound. It's connected to * 1.09 which is registered @broadvoice. Thanks for any insights bestowed on this noob :) Bob ___ --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 Yahoo! Messenger http://us.rd.yahoo.com/mail/uk/taglines/default/messenger/*http://uk.messenger.yahoo.com NEW - crystal clear PC to PC calling worldwide with voicemail http://us.rd.yahoo.com/mail/uk/taglines/default/messenger/*http://uk.messenger.yahoo.com ___ --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 ___ --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] grandstream handytone 488 fxo
nope, i havent :\ Keith Yoder wrote: Casey Boone escreveu: can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? Keith Yoder ___ --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 ___ --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] grandstream handytone 488 fxo
i greatly appreciate the information and will be giving it a whirl later today :) Casey Soner Tari wrote: I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) That's for making outbound calls. Once you've done this, you can direct incoming calls to a context like this: exten = 50,1,Goto(MainMenu,s,1) You should enter 50 to Forward to VoIP box at the bottom of HT488 config page also. (Choose an extension as you like instead of 50) But beware, hangup detection method of HT488 was too simple for my needs. Incoming calls may leave the port open indefinetly. (In combination with the FXS port of a HT486, it works, but that's it.) Hope this helps, Soner nope, i havent :\ can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? ___ --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 ___ --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] grandstream handytone 488 fxo
can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that [EMAIL PROTECTED] has this built in to just a button hit, but i dont want to reinstall the box and would prefer to use asterisk directly Casey Boone ___ --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] IP Cop as a firewall and QOS
i have never used ipcop in conjunction with asterisk (just starting asterisk) but i have used ipcop quite a bit. ipcop is a GREAT alternative to appliance firewalls and does almost anything i have ever needed to do. it is easy to install and easy to maintain and just works. the hardest thing is the way they name their interfaces and it isnt all that hard. red is the internet port green is the local lan port orange (if you have one) is the DMZ blue (if you have one) is for wireless typically you just need to worry about red and green i set up a series of vpn links based on some cheap celeron 1.7s with 256megs of ram and i could saturate a 100MBit connection without the ip cop machine breaking a sweat (this was during testing, would have went with much lower powered machines except for the great deal the company i worked for had for those boxes.) i would try it and see personally, find a box you are willing to use for an ipcop test and go to it. Casey Boone Mojo Jojo wrote: We are looking for a good firewall replacement which will basically do pot blocking and QOS. Our current solution just plain stinks.. We basically need to handle the traffic of a few web servers, mail server and asterisk box. The most traffic this device will need to handle is what can be shoved through a T1. I don't mind buying an appliance to get something solid but IP Cop just looks better than he appliances I see out there. I am only concerned if it is stable for a production environment. It says it's designed for a SOHO environment, we are doing a bit more than that. Will this thing hold up? Can it be trusted? Anyone using this for QOS and Asterisk in a production setup. Any thoughts or suggestions or warnings would be appreciated! Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.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 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
