[asterisk-users] Help needed creating gateway
Hi, all Can someone give me an example on how to do following: Asterisk receives incoming call from SIP Asterisk asks for a pin number Astersisk provides dialtone Asterisk collects digits from the caller and places a call on another interface Any pointers are greatly appreciated. Thanks, Rudolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] FRAUD: BE AWARE
Palestine? I would certainly refer this to police. Just in case. Rudolf On Thu, Jul 3, 2008 at 2:03 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Case wrote: Hi List, I made the mistake of having auto payments via PayPal. Just had some one put in payments and have them all denied. So far this person send in funds from: julie tosh - [EMAIL PROTECTED] David Somerville - [EMAIL PROTECTED] Gaetane Fortier - [EMAIL PROTECTED] ray stewart - [EMAIL PROTECTED] Cédric Girard - [EMAIL PROTECTED] The IP's I have are 213.6.185.243 and 83.233.182.229. The seem to be calling Palestine Mobile. There's hundreds of them. They change IP addresses and accounts. And if you accept payments, the owner of the account will quite often reverse the payment once they find out. I've tried emailing the address associated with the paypal account but have often found their hotmail/gmail etc to have been compromised by the same person. It's a real problem with not much of a solution in sight. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIbE+jDQNt8rg0Kp4RAsEYAJ9Z9PaTWPT3wOemr14Mgo5FQ+9LLgCfc564 JlqLuZYN0QX7XPM8y9+E1Uk= =Obty -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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
[asterisk-users] Problem with SPA3000 -- dropping calls
Hi, all I have SPA3000 (in Linksys reincarnation) and it has very annoying problem. Sometimes, incoming PSTN call drops the moment one picks up analog phone on FXO port. Most of the times it works, other times phone on FXS rings, I pick it up and all I get is a dial tone. Any ideas what may be wrong? Thanks, Rudolf ___ -- 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
[asterisk-users] SPA3000 + asterisk +call waiting
Hi, all A quick question. I have SPA3000 and trying to get call waiting to work. I do receive call waiting tone, however hook flash does not seem to work. I think, I set up SPA3000 correctly. Basically, doing HF 2 (switching calls in Australia) does not do anything. Is there any examples on how to setup hook flash operation in Asterisk? Thanks, Rudolf ___ -- 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
[asterisk-users] Configuring Polycom SP300 -- weird problem
Hi, all At some stage while back I had Polycom SP300 that was connected to the server 192.168.1.50 and all was fine. Phone was getting configuration via TFTP. I dug it out now and trying to connect to a new server 192.168.1.1 and can not change the address in the phone! I tried to change configuration file on the TFTP server, tried to change the address on the phone and via web interface. Upon reset, phone defaults to the old IP address. At the moment I got it going by aliasing IP address on the server, but I would like to get it to work properly. Can someone hint me on where this configuration may be stored? Thanks, Rudolf ___ -- 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
[asterisk-users] SPA3000 -- PSTN to VoIP
Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and configured PSTN as a VoIP extension in asterisk, but on incoming PSTN call, I do not see anything on asterisk console. Can someone point me into the right direction? Thanks, Rudolf ___ -- 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
[asterisk-users] How to exit from console?
Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --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] How to exit from console?
Thanks to all. All OK now. I thought that -c option is equivalent to starting an asterisk daemon and connecting to it. Obviously I was wrong. Thanks again, Rudolf On 1/23/07, Doug Lytle [EMAIL PROTECTED] wrote: Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? If you start Asterisk without any options: asterisk And then reconnect to it via the -r option asterisk -r Then typing exit on the console will exit without stopping Asterisk. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] SIP/RTP Nat problem, can't solute it.
NAT changes address of the packet, but does not go inside of the SIP packet itself. And SIP packet contains address as well. If you look at debug output, you will see that SIP packets have remote host local address in them, not the public IP as one would expect. At least this is the problem I have. Basically one needs some software to NAT the addresses inside of SIP packets. STUN server is one alternative. I am about to put one in. Rudolf On 1/7/07, C F [EMAIL PROTECTED] wrote: Change To canreinvite=no On 1/6/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote: Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar my server name on the internet localnet=192.168.5.0/255.255.0.0 my LAN nat=yes canreinvite=yes And this are the ports i opened on my firewall script iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT But still can't hear a thing from an outside call, any hel will be appreciate Thanks a lot -- _ Facundo Agustin Barrera -- www.openlabs.com.ar Let the penguins do the work - Buenos Aires - Argentina _ ___ --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 ___ --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] Zaptel under FC6
Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running on FC3 before. Thanks, Rudolf ___ --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] Zaptel under FC6
Thanks for suggestion. Just tried that Was surprised with download size of only 72k. Anyway, command work, but I still have same ptoblem. I tried to run modprobe -- it failed. Tried to run service zaptel start and get: service zaptel start No functioning zap hardware found in /proc/zaptel, loading ztdummy Loading ztdummy: FATAL: Module ztdummy not found. [FAILED] Running ztcfg: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected [FAILED] . I guess, modules are mot there. Running find / -name zaptel* did not find any modules. Seems that make is broken in some way. Rudolf On 12/15/06, Howard Lowndes [EMAIL PROTECTED] wrote: Have you done yum install zaptel. It's part of Fedora 6 Extras along with openpbx, a fork of Asterisk. Yuan LIU wrote: From: Rudolf Ladyzhenskii [EMAIL PROTECTED] Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running on FC3 before. I'm not running any Fedora, but I suspect that the installation layout no longer symblink under /lib/modules/ from [full-version] to [major-version]. Such is the case with Ubuntu I'm using. For example, if your full kernel path is 2.6.15-27-386, you'll find zaptel modules in /lib/modules/2.6.15/misc/; in the meanwhile, Linux is looking under /lib/modules/2.6.15-27-386/ for any loadable kernel modules. Of course module not found. If this is the case, there are two ways to get around. . Remove physical /lib/modules/2.6.15/, symblink /lib/modules/2.6.15 to /lib/modules/2.6.15-27-386/, rerun make install; or, alternatively, . Move /lib/modules/2.6.15/misc/ to under /lib/modules/2.6.15-27-386/, run depmod. Both should lead to a happy ending. I prefer the first one as it makes future zaptel upgrades happier; of course you can also make symblink after the second. Hope this helps. Yuan Liu Thanks, Rudolf ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannetlinux.com When you want a computer system that works, just choose Linux; When you want a computer system that works, just, choose Microsoft. -- Flatter government, not fatter government; abolish the Australian states. ___ --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] HELP! Bad sound quality
Hi, all Suddenly I started to have bad sound quality. Happens with all providers as well as with softphones connected to my * server on the Internet. It was OK on 1.1.17, then I migrated to 1.2.5 and it was OK for some time. Nothing has changed in my setup, but voice quality degraged greatly. When I call someone, they can hear myself quite clear, but I hear lots of interruptions in the voice. It actually gets worse. It may start as a good clear conversation and in a few seconds it slips and I can not make out what they say. Seem to only happen when calling via Internet. I have tried to restart both * server and my main server/gateway. I also made sure no other traffic is going through. Nothing like P2P. When I do use P2P I am getting my usual dowload speed, so looks like my ISP is fine. I run ADSL 512/128. Any ideas on what could happen? Thanks, Rudolf ___ --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] HELP! Bad sound quality
I think you are right. 1.0.7 I connect via VoIP providers -- via Internet only. No direct PSTN connection. (Well I do have TDM400, but did not have time to set ot up yet). I use Polycom SP300 phones I even have problems when talking to people with softphones registered on my * server. Somehoe, I am starting to suspect that my ISP have something to do with that. Is there any way to check quality of Internet connection? Not just speed but quality. Thanks, Rudolf On 4/14/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Apr 14, 2006 at 05:47:32PM +1000, Rudolf Ladyzhenskii wrote: Hi, all Suddenly I started to have bad sound quality. Happens with all providers as well as with softphones connected to my * server on the Internet. It was OK on 1.1.17, then I migrated to 1.2.5 and it was OK for some time. Nothing has changed in my setup, but voice quality degraged greatly. 1.1.17? Do you mean 1.0.7? Could you also give some details about your setup? How do you connect to your provider? If via ISDN: what ISDN channel? When I call someone, they can hear myself quite clear, but I hear lots of interruptions in the voice. It actually gets worse. It may start as a good clear conversation and in a few seconds it slips and I can not make out what they say. Seem to only happen when calling via Internet. I have tried to restart both * server and my main server/gateway. I also made sure no other traffic is going through. Nothing like P2P. When I do use P2P I am getting my usual dowload speed, so looks like my ISP is fine. I run ADSL 512/128. Any ideas on what could happen? Thanks, Rudolf ___ --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 ___ --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] Multiple phones in same call
Hi, all This is what I would like to do: Someone is on the phone and nother person ant to join in. Like in house wheer all phones are connected to same line. I can do it with MeetMe, but my understanding is that all parties have to call meeting room number. What I want instead is to have some magic extension or a * or # service that will allow people to join in. I am running Asterisk at my home and normally have 3 or 4 phones only, so ideally I would number all phones 1-4 and pressing say *1 on phones 2-4 will join to the call on phone 1. Is there an easy way of doing that? Thanks, Rudolf ___ --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] Voipstunt, voipbuster, .... not working properly?
What is the SIP server you specified? Rudolf On 4/10/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: I am still trying to figure out how to overcome this problem. I use for International calls a, for USA calls b, ... Most of the time I get: Forbidden - wrong password on authentication for INVITE I would like in that case that the next gateway will be used. How can I do that? bye Ronald Wiplinger ___ --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] G729 codec problems
I think I sort of solved the problem. It is related to Voipstunt provider. I tried it today with another provider and all is fine. Well, I get some warnings from G729 like: Jun 26 09:28:45 NOTICE[8440]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Bu nothing like before. Use Voipstunt and I have the problem. It seems that Voipstunt does use more than 1 lisence on a single call. Can someone check/verify that? Thanks, Rudolf On 4/3/06, Steve Kennedy [EMAIL PROTECTED] wrote: On Sun, Apr 02, 2006 at 12:55:13PM -0500, Kevin P. Fleming wrote: Steve Kennedy wrote: Each channel needs TWO licenses, one for each way (I think). Nope. The encoder/decoder licenses are counted separately, and each license you purchase entitles you to one encoder and one decoder. Hmm, sorry about that. Thanks for the clarification. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] G729 codec problems
Hi, I wonder if VoIP providers consume two licenses when one calls via them? One license for my phone to the provider and one license when call is passed to the recepient. Is that possible? Rudolf On 4/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Steve Kennedy wrote: Each channel needs TWO licenses, one for each way (I think). Nope. The encoder/decoder licenses are counted separately, and each license you purchase entitles you to one encoder and one decoder. ___ --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] G729 codec problems
I am not. I have one license and use i channel. It seems to detect the fact there are no more channels left and keeps warning me about it in case I want to use more. It is fine, but the warning is constant. All you see on Asterisk console is running warning message. Rudolf On 4/2/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: RumaTech wrote: And it keeps running like that. Call usually come through OK. If i try to use show g729 command, it shows that all codecs are in use. Well, this is fine, I am using one, but I do not want to see those warnings. Once is quite enough. Those continuos warnings make it impossible to se any other asterisk output. How does one turns them off? You can't make them stop except by not trying to use more channels than you have licenses for. ___ --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] Voicemail to Email
Voicemail uses sendmail on your system. If your machine can send mails using sendmail, so will asterisk. Rudolf On 3/27/06, voipman [EMAIL PROTECTED] wrote: Could anyone provide me some link in order to voicemail to email working, I believe I have to give SMTP settings but do not know where. Thx Voipman ___ --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] G729 codec problems
Hi, all I have a license for G.729A codec from Digium. When asterisk starts it shows: Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:460 load_module: G.729 transcoding module Copyright (C) 1999-2005 Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:461 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:462 load_module: Please see the full license text supplied by the accompanying Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:463 load_module: register utility, or ask for a copy from Digium. == G.729 Host-ID: cc:20:a3:86:01:93:53:92:2c:37:ae:e7:ad:16:6e:f0:39:f6:88:4e == Found license 'G729-190B962C' providing 1 channels == Found total of 1 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 20 == Registered translator 'lintog729' from format slin to g729, cost 115 All is fine, however when trying to make a call I am getting: WARNING[4063]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! No other calls are active. Any ideas what is going on? Thanks, Rudolf ___ --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] Mailing list problems with gmail!!!!
Hi, all I stopped receiving messages from the list. tried to change the address, gettimg confirmation, but no messages!. All addresses I use are via gmail. I can see my messages reach the list (looked in archive), but nothing in e-mail (nothing in spam folder either). Was ok until sometime ago. If anyone knows how to fix it, please e-mail direct, do not reply to the list as i will not see it. Tnanks, Rudolf ___ --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] exclude context?
Hi, all Is there a way in extensions.conf to exclude context? For example, I have contexts A, B and C. I want something like this: [A] some extensions [B] some extensions include=A [C] some extensions include=B excludeA If I only do include=B in [C], it will automatically include [A] in there as well, right? This is something I want to avoid. Thanks, Rudolf ___ --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] Asterisk and Fedora
Here is a link to get you going: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 Rudolf - Original Message - From: Luke Kearney [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 16, 2005 11:20 PM Subject: [Asterisk-Users] Asterisk and Fedora Hello List, I have been beating my head against the wall for a little while now trying to get my TDM400 card to work with Fedora Core 4. Not a great deal of success even after successful builds of zaptel and the other required componentry. The machine doesn't even recognize the card. Using Debian it was recognised but as a newbie to Asterisk most of the documentation I can find and understand is written with RH or Fedora in mind it seems. Deb does things ever so slightly differently. I read on the list a little while ago comments that indicated that Fedora Core 4 was not suitable and am contemplating going back to Fedora Core 3. The hardware is pretty generic Intel Pentium 4 based hardware. Has anyone had good experiences with Fed Core 3 ? Or is it something to stay away from ? Kind Regards, - Luke Kearney [EMAIL PROTECTED] ___ --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] No voice - one way - both ways
Firewall/NAT problem? Are all phones on same subnet? Rudolf - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, October 16, 2005 9:29 PM Subject: [Asterisk-Users] No voice - one way - both ways I got four phones: 601 is a SIP phone (no brand) 615 is Snom 190 621 is a Grand stream 628 is a remote SIP phone (no brand) 601, 615, 628 can call each other without any problems 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 11 the remote party cannot hear me. 615 never could call remote 628, both party hear nothing. 601 can always call 628 [Oct 16 00:52:13] -- Executing Dial(SIP/621-673f, SIP/628|60|r) in new stack [Oct 16 00:52:13] -- Called 628 [Oct 16 00:52:13] -- SIP/628-9d23 is ringing [Oct 16 00:52:15] -- SIP/628-9d23 answered SIP/621-673f [Oct 16 00:52:15] -- Attempting native bridge of SIP/621-673f and SIP/628-9d23 She cannot here me!!! [Oct 16 00:52:30] == Spawn extension (default, 628, 1) exited non-zero on 'SIP/621-673f' [Oct 16 00:52:30] -- Executing Hangup(SIP/621-673f, ) in new stack [Oct 16 00:52:30] == Spawn extension (default, h, 1) exited non-zero on 'SIP/621-673f' [Oct 16 00:53:06] -- Executing Playback(SIP/621-88e8, demo-echotest) in new stack [Oct 16 00:53:06] -- Playing 'demo-echotest' (language 'en') [Oct 16 00:53:26] -- Executing Echo(SIP/621-88e8, ) in new stack [Oct 16 00:53:33] == Spawn extension (default, 690, 2) exited non-zero on 'SIP/621-88e8' [Oct 16 00:53:33] -- Executing Hangup(SIP/621-88e8, ) in new stack [Oct 16 00:53:33] == Spawn extension (default, h, 1) exited non-zero on 'SIP/621-88e8' Echo test no problem, means phone is ok!! [Oct 16 00:53:41] -- Executing Dial(SIP/621-b113, SIP/628|60|r) in new stack [Oct 16 00:53:41] -- Called 628 [Oct 16 00:53:41] -- SIP/628-b3b6 is ringing [Oct 16 00:53:51] -- SIP/628-b3b6 answered SIP/621-b113 [Oct 16 00:53:51] -- Attempting native bridge of SIP/621-b113 and SIP/628-b3b6 [Oct 16 00:53:58] == Spawn extension (default, 628, 1) exited non-zero on 'SIP/621-b113' [Oct 16 00:53:58] -- Executing Hangup(SIP/621-b113, ) in new stack [Oct 16 00:53:58] == Spawn extension (default, h, 1) exited non-zero on 'SIP/621-b113' She cannot hear me [Oct 16 00:55:19] -- Executing Hangup(SIP/615-a5bd, ) in new stack [Oct 16 00:55:19] == Spawn extension (default, h, 1) exited non-zero on 'SIP/615-a5bd' [Oct 16 00:55:23] == Spawn extension (VoIP_customer_Phone_routes, 621, 2) exited non-zero on 'SIP/628-aba4' [Oct 16 00:55:35] -- Executing Dial(SIP/615-31a8, SIP/628|60|r) in new stack [Oct 16 00:55:35] -- Called 628 [Oct 16 00:55:36] -- SIP/628-7293 is ringing [Oct 16 00:55:42] -- SIP/628-7293 answered SIP/615-31a8 [Oct 16 00:55:42] -- Attempting native bridge of SIP/615-31a8 and SIP/628-7293 [Oct 16 00:55:51] == Spawn extension (default, 628, 1) exited non-zero on 'SIP/615-31a8' [Oct 16 00:55:51] -- Executing Hangup(SIP/615-31a8, ) in new stack [Oct 16 00:55:51] == Spawn extension (default, h, 1) exited non-zero on 'SIP/615-31a8' We both cannot hear [Oct 16 00:56:08] -- Executing Dial(SIP/601-bb26, SIP/628|60|r) in new stack [Oct 16 00:56:08] -- Called 628 [Oct 16 00:56:09] -- SIP/628-0be9 is ringing [Oct 16 00:56:16] -- SIP/628-0be9 answered SIP/601-bb26 [Oct 16 00:56:16] -- Attempting native bridge of SIP/601-bb26 and SIP/628-0be9 [Oct 16 00:58:36] == Spawn extension (default, 628, 1) exited non-zero on 'SIP/601-bb26' [Oct 16 00:58:36] -- Executing Hangup(SIP/601-bb26, ) in new stack [Oct 16 00:58:36] == Spawn extension (default, h, 1) exited non-zero on 'SIP/601-bb26' Call ok!!! SIP.conf: [601] type=friend username=601 secret=youdontneedtoknow canreinvite=no host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,601 qualify=1000 [615] ; snom 190 type=friend ; Friends place calls and receive calls username=615 ; Username to use in INVITE until peer registers secret=youdontneedtoknow host=dynamic ; This peer register with us dtmfmode=rfc2833 qualify=1000 [EMAIL PROTECTED] ; Mailboxes for message waiting indicator restrictcid=yes ; To have the callerid restriced - sent as ANI disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! allow=alaw allow=g729 callerid=Ronald Snom,615 callgroup=1 pickupgroup=1 621 and 628 are in realtime and have similar settings. Important I think is only the codec: 621: ulaw;alaw 628: g729;ulaw;alaw How can I solve it? bye Ronald Wiplinger === First they ignore you, then they laugh at you, then they fight you, then you win. —Mahatma Gandhi ___ --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:
Re: [Asterisk-Users] Asterisk and Fedora
I got TDM400 to work on FC3. You have to manualy update udev rules. I will post you the instructions when get home tonight. Rudolf - Original Message - From: Luke Kearney [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 16, 2005 11:20 PM Subject: [Asterisk-Users] Asterisk and Fedora Hello List, I have been beating my head against the wall for a little while now trying to get my TDM400 card to work with Fedora Core 4. Not a great deal of success even after successful builds of zaptel and the other required componentry. The machine doesn't even recognize the card. Using Debian it was recognised but as a newbie to Asterisk most of the documentation I can find and understand is written with RH or Fedora in mind it seems. Deb does things ever so slightly differently. I read on the list a little while ago comments that indicated that Fedora Core 4 was not suitable and am contemplating going back to Fedora Core 3. The hardware is pretty generic Intel Pentium 4 based hardware. Has anyone had good experiences with Fed Core 3 ? Or is it something to stay away from ? Kind Regards, - Luke Kearney [EMAIL PROTECTED] ___ --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] TDM400 not working -- SOLVED
Sorry, I was pretty busy and did not work on my *. Problem was in zaptel not properly registering driver with udev. Manually updating udev rules fixed the problem. Thanks, Rudolf - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 6:20 AM Subject: Re: [Asterisk-Users] TDM400 not working On Mon, Oct 10, 2005 at 07:25:59PM +1000, Rudolf Ladyzhenskii wrote: Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open '/dev/zap/channel': No such file or directory The device file does not exast Jan 2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable to open channel 2: No such file or directory here = 0, tmp-channel = 2, channel = 2 Jan 2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: Unable to register channel '2' Jan 2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 06:28:08 WARNING[3473]: loader.c:523 load_modules: Loading module chan_zap.so failed! Ok, I look in the /dev and I could not find /dev/zap at all! But, there is a /dev/zapchannel character device. Is that a typo? It should be /dev/zap/channel . Do you use udev? If so, see README.udev . If not: you need to generate those device files. Anyway: could you please post the output of: lsmod | grep zaptel Any ideas what can be wrong? And last question. Does zaptel driver reads configuration file on startup? If so, how do I force the driver to update if config file was changed? ztcfg loads the configuration to the zaptel module from /etc/zaptel.conf . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] TDM400 not working
Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open '/dev/zap/channel': No such file or directory Jan 2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable to open channel 2: No such file or directory here = 0, tmp-channel = 2, channel = 2 Jan 2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: Unable to register channel '2' Jan 2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 06:28:08 WARNING[3473]: loader.c:523 load_modules: Loading module chan_zap.so failed! Ok, I look in the /dev and I could not find /dev/zap at all! But, there is a /dev/zapchannel character device. Any ideas what can be wrong? And last question. Does zaptel driver reads configuration file on startup? If so, how do I force the driver to update if config file was changed? Thanks, Rudolf ___ --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] Configuring TDM400 in Australia
Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf ___ --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] Fax detection question
Hi, all Here is what I plan to do: Have an asterisk server with 1FXS and 1 FXO port. Will have fax machine connected to FXS and will use IP phones. I want asterisk to detect incoming fax and swith it to fax line automatically. Something like this: Incoming on FXO. Asterisk to pick up. Asterisk to detect if there is an incoming fax and switch to fax machine. If call is voice call, then ring IP phone(s). Detecting the fax is a grey area for me. Can asterisk do it? How do set it up? (HW is TDM400 card with 1FXS, 1FXO port). Thanks, Rudolf ___ --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] How to retrieve voicemail from an IP phone?
Hi,
You need a single extension to call voicemail. I am using 100.
extensions.conf
exten =100,1,VoiceMailMain(${CALLERIDNUM})
exten =100,2,Hangup()
Now, if you simply call VoiceMailMain() without parameters, voicemail system
will ask you to enter the number of mailbox you want to access. This is
useful if you want to read any mailbox from any phone.
However, if you specify a parameter like I did, voicemail will automatically
go into mailbox for the extension you have called from. There is a little
trick to get it work, though. Normally caller ID is a name like Joe Smith
You will have to specify caller ID per user like that: (sip.conf for
example)
[user1]
callerid=Joe Smith 101
This will present asterisk with a way to get both name and extension number.
Rudolf
- Original Message -
From: Ryan Pagquil [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 21, 2005 8:58 PM
Subject: Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?
Hi Rich,
Does the user need to dial his extension just to retrieve the
voicemails or he will dial other number to access those voicemails?
In the config does it mean that when a user dial 3998 he will be able to
retrieve those voicemails? So it means that every users must have a
mailbox number for which they will retrive their voicemails? I'm really a
newbie. =)
Thanks fo the help,
--ryan
Rich Adamson wrote:
How can I retrieve those voicemails using my ip phone? and how
will i confiugre it on asterisk?
Please help I'm very new in asterisk.
Add something like this in your extensions.conf file:
; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup
; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup
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Ryan Pagquil
Infodyne Inc. - PhilOnline.com
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Tel: 687-0715
Web: www.philonline.com
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Re: [Asterisk-Users] HW Question (TDM400)
Thanks, I do realise that I will have to upgrade, just did not want to do it so soon. Digium says that it is not recommended to use main mainboard with PCI less than 2.2. They did not say it will not work. I just wanted to find out if anyone had any experience with this particular chipset. So I will know if I have to go and get new hardware. I will try TDM400 in my machine on weekend. Rudolf - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 19, 2005 7:30 PM Subject: Re: [Asterisk-Users] HW Question (TDM400) On Mon, 2005-09-19 at 14:27 +1000, [EMAIL PROTECTED] wrote: Hi, all Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1 I do not want to upgrade yet. Since Digium is so clear about using a mainboard with PCI 2.2 I think you already know that bad/weird/unexpected/unexplainable things can happen when using the card in a PCI 2.1 board. Basically YMMV. If it works for you the better, if not you know why. I use an old Asus P2B-LS (PCI 2.1) board with a PII-400 and an Eicon Diva Server card as an Asterisk server and it works fine. Regards, Patrick ___ --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] Voipbuster in Australia -- delay problem
Hi, all, I got my * to work with voipbuster service. And it works quite well when I am calling USA or Europe. However, for local calls, I am experiencing long delays (About 1s). As far as I know, voipbuster application does not have this problem. I am using IAX and gsm codec. Any ideas on how to combat this? Thanks, Rudolf ___ --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] Voipbuster in Australia -- delay problem
Thanks, Paul I have tried to set tos=0x18 and this improved it a lot. Also, looks like I have to seriosly look into setting up QoS on my firewall. At the monet I shut down P2P application running on one of the PC and this has helped too. Not sure why it was affecting delay. I would expect actual quality to degrade if not enough bandwith present. Anyway, it is much better now. Rudolf - Original Message - From: Paul [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 19, 2005 9:23 PM Subject: Re: [Asterisk-Users] Voipbuster in Australia -- delay problem Rudolf Ladyzhenskii wrote: Hi, all, I got my * to work with voipbuster service. And it works quite well when I am calling USA or Europe. However, for local calls, I am experiencing long delays (About 1s). As far as I know, voipbuster application does not have this problem. I am using IAX and gsm codec. Any ideas on how to combat this? Try all supported codecs with no jitterbuffer. If that won't reduce the delay it is probably due to their choice of provider for AU. Keep in mind that your internet route for IAX to them may be good but it is probably totally different from their route to AU PSTN. ___ --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] Slightly OT: VoIPbuster
Hi, all I have managed to connect my * to voipbuster server using IAX. So far, so good. I have paid them the 5 euro and now my PC aplication has unlimited local calls. However calls from * are still limited to 1 minute. I have sent them an e-mail asking about it, but no answer as yet. Could it be that * is not registering properly? Did anyone experienced this problem? At the moment I think that their IAX server does not have access to accounting information, but I might be wrong there. Thanks, Rudolf ___ --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] VoipBuster again -- WORKS NOW!!!!
Hi, all
Finally got it to work.
TWo problems.
1. Stupid erro on firewall caused authentication failure. BAsically IAX
ports were forwarded to the * box without noting the interface they were
coming on. Thsi is OK when external clients tried to connect, but when I
tried to connect to outside, firewall was forwarding my requests back to
asterisk box, so it was trying to authenticate against itself. Runnign
Ethereal on firewall helped to find this problem.
2. Still could not connect call, although registration was working.
Had to change dialing string to:
exten = _0.,2,Dial,IAX2NAME:[EMAIL PROTECTED]/00613${EXTEN:1}
By some reason I had to use full server name, it was not picking it up from
iax.conf Will look at it later when I have a chance.
Rudolf
- Original Message -
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Sunday, September 11, 2005 5:26 PM
Subject: RE: [Asterisk-Users] VoipBuster again
Try this ip for register something looks wrong with iax.voipbuster.com
I changed it a while ago because i had some dns problems in with my
provider
and this ip came up when i pinged now you can't ping to the adress and
it's
another ip
register = username:[EMAIL PROTECTED]
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf
Ladyzhenskii
Verzonden: zondag 11 september 2005 0:25
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] VoipBuster again
Here is what I get when reloading IAX2:
Not every time, though
== Parsing '/etc/asterisk/iax.conf': Found Sep 11 08:48:29 WARNING[3240]:
chan_iax2.c:5402 iax2_register: Host 'iax.voipbuster.com' not found at
line
164
Strange, because name resolves to IP address.
Ok, I reload IAX2 again and no more warning.
Then it tries to register and fails:
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00017ms SCall: 1 DCall: 0 [213.61.187.146:4569]
USERNAME: USERNAME
REFRESH : 60
And so it goes.
Call then fails too...
I am suspecting two things:
1. I am starting to wonder if registering a user in Australia using
VoipBuster application does not create an IAX account
Can someone who has an IAX account try creating one for me? Bogus name and
password. my e-mail is [EMAIL PROTECTED]
2. Firewall ports are not open. I am sure all the right ports are
forwarded
to my * box (5060, 4569, 1-2).
I will set up ethereal on my firewallbox to see what comes out to the www
and what comes back.
Thanks,
Rudolf
- Original Message -
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, September 10, 2005 11:32 PM
Subject: RE: [Asterisk-Users] VoipBuster again
Iax.conf
register = username:[EMAIL PROTECTED]
Extensions.conf
exten =
_0.,2,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:\1\},\
60,r)
Good luck :) Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf
Ladyzhenskii
Verzonden: zaterdag 10 september 2005 13:57
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP
or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port
2
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster?
If
so, can you you help me to set it up? I am really lost.
My setup is :
sip.conf
[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com
iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no
extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP) exten
=
_0.,1,SetCallerID(CID Name CIDNUMBER) exten =
_0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}
exten = _8.,1,SetCallerID(CID Name CIDNUMBER) exten =
_8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}
Thanks,
Rudolf
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[Asterisk-Users] VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 2
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster? If
so, can you you help me to set it up? I am really lost.
My setup is :
sip.conf
[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com
iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no
extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP)
exten = _0.,1,SetCallerID(CID Name CIDNUMBER)
exten = _0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}
exten = _8.,1,SetCallerID(CID Name CIDNUMBER)
exten = _8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}
Thanks,
Rudolf
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Re: [Asterisk-Users] VoipBuster again
Here is what I get when reloading IAX2:
Not every time, though
== Parsing '/etc/asterisk/iax.conf': Found
Sep 11 08:48:29 WARNING[3240]: chan_iax2.c:5402 iax2_register: Host
'iax.voipbuster.com' not found at line 164
Strange, because name resolves to IP address.
Ok, I reload IAX2 again and no more warning.
Then it tries to register and fails:
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00017ms SCall: 1 DCall: 0 [213.61.187.146:4569]
USERNAME: USERNAME
REFRESH : 60
And so it goes.
Call then fails too...
I am suspecting two things:
1. I am starting to wonder if registering a user in Australia using
VoipBuster application does not create an IAX account
Can someone who has an IAX account try creating one for me? Bogus name and
password. my e-mail is [EMAIL PROTECTED]
2. Firewall ports are not open. I am sure all the right ports are forwarded
to my * box (5060, 4569, 1-2).
I will set up ethereal on my firewallbox to see what comes out to the www
and what comes back.
Thanks,
Rudolf
- Original Message -
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, September 10, 2005 11:32 PM
Subject: RE: [Asterisk-Users] VoipBuster again
Iax.conf
register = username:[EMAIL PROTECTED]
Extensions.conf
exten =
_0.,2,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:\1\},\
60,r)
Good luck :) Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf
Ladyzhenskii
Verzonden: zaterdag 10 september 2005 13:57
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP
or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port
2
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster?
If
so, can you you help me to set it up? I am really lost.
My setup is :
sip.conf
[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com
iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no
extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP) exten
=
_0.,1,SetCallerID(CID Name CIDNUMBER) exten =
_0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}
exten = _8.,1,SetCallerID(CID Name CIDNUMBER) exten =
_8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}
Thanks,
Rudolf
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Re: [Asterisk-Users] VoipBuster with astersisk?
Hi, all
Still struggling.
Here is my iax.conf entry:
[voipbuster]
type=peer
host=iax.voipbuster.com
secret=MYSECRET
notransfer=yes
context=pignet
Here is my extension.conf entry:
exten = _0.,1,SetCallerID(CID Name CIDNUMBER)
exten = _0.,2,Dial(IAX2/[EMAIL PROTECTED]/00613${EXTEN:1})
(I am using 0 to dial out and country and area code is appended
automatically).
When I attempt a call I can see I am calling the right place, but I do not
get any responce and call fails. Here is log:
-- Executing Dial(SIP/phone1-f02f,
IAX2/[EMAIL PROTECTED]/0061395433089) in new stack
-- Called [EMAIL PROTECTED]/0061395433089
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00018ms SCall: 1 DCall: 0 [213.61.187.150:4569]
VERSION : 2
CALLED NUMBER : 0061395433089
CODEC_PREFS : ()
CALLING NUMBER : 101
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Rudolf Ladyzhenskii
LANGUAGE: en
USERNAME: MYNAME
FORMAT : 2
CAPABILITY : 65283
ADSICPE : 2
DATE TIME : 2005-09-02 23:08:14
phonebox2*CLI
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00018ms SCall: 1 DCall: 0 [213.61.187.150:4569]
VERSION : 2
CALLED NUMBER : 0061395433089
CODEC_PREFS : ()
CALLING NUMBER : 101
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Rudolf Ladyzhenskii
LANGUAGE: en
USERNAME: MYNAME
FORMAT : 2
CAPABILITY : 65283
ADSICPE : 2
DATE TIME : 2005-09-02 23:08:14
phonebox2*CLI
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE
Timestamp: 00017ms SCall: 5 DCall: 0 [213.61.187.150:4569]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE
Timestamp: 00017ms SCall: 5 DCall: 0 [213.61.187.150:4569]
Sep 2 23:08:19 NOTICE[3240]: chan_iax2.c:2756 auto_congest: Auto-congesting
call due to slow response
-- IAX2/voipbuster-1 is circuit-busy
-- Hungup 'IAX2/voipbuster-1'
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/phone1-f02f' status is 'CONGESTION'
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
Timestamp: 04022ms SCall: 1 DCall: 0 [213.61.187.150:4569]
CAUSE CODE : 0
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
Timestamp: 04022ms SCall: 1 DCall: 0 [213.61.187.150:4569]
CAUSE CODE : 0
If I call same number from voipbuster application it connects OK.
Any help is greatly appreciated.
Thanks,
Rudolf
- Original Message -
From: Mat Stace, Colewood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 01, 2005 9:54 AM
Subject: Re: [Asterisk-Users] VoipBuster with astersisk?
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK landline numbers :)
In my iax.conf
[voipbuster]
type=peer
host= 213.61.187.150
secret=YOURPASSWORD
notransfer=yes
context=default
In My extensions.conf:
exten = _770[12].,1,SetCallerID(CID Name CIDNUMBER)
exten = _770[12].,2,Dial,IAX2/[EMAIL PROTECTED]/0044${EXTEN:3}
I don't actually know if the first line works (never actually tested it
that far :-| ) and you'll probably want the 2nd line to be something like
this if you want to use it for all calls worldwide
exten = _9.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}
This should give you the 9 for a line mdoe of operation, and require you
to dial full international numbers.
Cheers
Mat
(standard disclaimer - while the above works for me, it's for a particular
purpose. YMMV, don't sue me if it breaks, etc etc etc) ;-D
[EMAIL PROTECTED] wrote:
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use my
normal phones instead of PC application.
Did anyone try to connect astersisk and VoipBuster?
Thanks,
Rudolf
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Re: [Asterisk-Users] NAT and SIP.conf update.
Message I have a standard BT home DSL, which means I cannot have a static IP address, therefore i'm forced to use NAT, I subscribe to a DDNS service and have written a VB app which polls the router every 10 seconds and updates the DDNS if appropriate. There are ready applications to do that This is fine but I need to be able to modify my sip.conf (externip = w.x.y.z) and reload sip, does anyone know of a script/app which does an nslookup and modifies the conf file, then reloads sip? What are you running behind NAT? Do you have asterisk server or SIP client? In case of server, you just set in sip.conf externip my address as FQDN In case of SIP clients, set : host=dynamic in the definition of relevant client. I have both server and client running behind NAT without any problems. Rudolf ___ 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] Voicemail Retrival
Hi,
This procedure will work under one condition --
your user names are same as your extension numbers. I have same problem. I was
giving phones alphanumeric user names, like "phone1".
When VoicemailMain is called with ${CALLERIDNUM},
it is actually called as VoiceMailMain("phone1"). As a result, voice mail is
asking for a mailbox number which is same as your extension number. (BTW, is
there a way to extract extension number rather than phone name?).
As I am experimenting with *, I will rename phones
to match their extensions.
Rudolf
- Original Message -
From:
Sharadindu Mohanty
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 17, 2005 8:32
PM
Subject: Re: [Asterisk-Users] Voicemail
Retrival
I did the same way but it is asking for some password and mailbox. I
think mail box is extension no but what abt password?
Can i overide this procedure?
ThanksChristoph Eicke [EMAIL PROTECTED] wrote:
On
Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote:
Hi,Hi! Any ideas??Yes, I do it in the following way. In
extension.conf add this line:exten =
,1,VoiceMailMain(s${CALLERIDNUM})exten =
,2,Hangup()Here any extension can call and then
automatically gets directed to their voicemail where they have some
options.I hope this
helps,Christoph___Asterisk-Users
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visit:http://lists.digium.com/mailman/listinfo/asterisk-usersSharadindu
Mohanty
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Re: [Asterisk-Users] Voicemail Retrieval
In addition to my previos e-mail.
'callerid' filed in sip.conf or iax.conf
(depends where user is defined) must be set to"
callerid "User Name" EXT
Where EXT is a number that will be picked up by
VoiceMailMain and will be used as a mailbox number.
Rudolf
- Original Message -
From:
Wei Kun
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, August 17, 2005 6:37
PM
Subject: RE: [Asterisk-Users] Voicemail
Retrieval
Take
this as an example
[from-sip]
exten
= 2000,1,Dial(SIP/2000,20)
exten
= 2000,2,Voicemail(u2000)
exten
= 2000,102,Voicemail(b2000)
exten
= 2000,103,Hangup
exten
= 2001,1,Dial(SIP/2001,20)
exten
= 2001,2,Voicemail(u2001)
exten
= 2001,102,Voicemail(b2001)
exten
= 2001,103,Hangup
exten
= 2999,1,VoicemailMain(${CALLERIDNUM})
you then dial 2999 to retrieve
it.
Kun
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
Sharadindu MohantySent: Wednesday, August 17, 2005 4:30
PMTo: asterisk-Users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Retrival
Hi,
I am very new to Asterisk. I wanted to know how to retrive
the Voicemails. I could see some voicemails assosiated with some
extensions.
Any ideas??
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Re: [Asterisk-Users] Echo problem -- network related?
You were right and I was wrong. New sound card fixed all problems. Still can not beleive that problem was caused by audio hardware, but there we are. Thanks to all who replied. Rudolf - Original Message - From: Rob Lith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 6:14 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Echo problem -- network related?
Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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
Re: [Asterisk-Users] Echo problem -- network related?
Thanks for reply. You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. Yes, this would be the logical conclusion, although it is hard to beleive given what I hear. It sound like I am talking to myself at a pretty good quality. Actually echo quality is much better than other party. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Sound card used is a built into the main board -- Gigabyte 8PIE1000 board with Realtek AC97. Not a cheap crapy board. I have tried new drivers too. I am going to try few things -- try his computer on my LAN to rule out any network related issues Try USB handset and/or difefrent sound card I wil let you all knwo when I find something out. Thanks again, RUdolf Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Why NAT problem
At firewall/NAT you have to do port forwarding. If your phone is at port 5060, NAT device will receive a connection and has to know that it is destined for your SIP phone. So, forward port 5060 to the phone. Rudolf - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 6:52 AM Subject: [Asterisk-Users] Why NAT problem hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing not registered. i think asterisk is properly sending request to UA. any commentsthis sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved useragent SJLabs-SJphone/1.40.258 for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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
Re: [Asterisk-Users] Echo problem -- network related?
Hi, The problem is not sound setup related. It present even if microphone is disconnected. Rudolf - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 12:12 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? Rudolf Ladyzhenskii wrote: Hi, all I am running asterisk and my friends are running FireFly IAX phone. All is fine except one of them. When anyone tries to talk to him, tehre is a real bad echo. It is nothing to do with sound setup. Is he using a headset or speakers and microphone? Does he have Stereo Mix selected as a recording source? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
Re: [Asterisk-Users] Why NAT problem
In case of IAX phones, this is not appliacble. IAX uses same port for both control and voice. I have not tried CISCO phones, but I beleive you do need port forwarding if they are SIP phones. Otherwise, they will not accept calls. At least this is the case with Polycom phones. Rudolf - Original Message - From: Tom Rymes [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 12:30 PM Subject: RE: [Asterisk-Users] Why NAT problem As a followup to my own post, AFAIK, my comments apply to SIP clients, but you always have to forward the ports to the asterisk server... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Saturday, August 13, 2005 10:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why NAT problem This is not technically true. For instance, you can take a Cisco 79X0 and put it behind NAT and it will work without port forwarding. You do, however, have to program the phone to enable the NAT features. (There are two, I can't remember their names, though.) I have generally left the WAN IP address blank, with no noticable ill effects, but that might not be a good idea. Also, I believe that you can do this with multiple phones, so long as you use different port numbers for each phone (5061, 5062, etc) Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Saturday, August 13, 2005 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why NAT problem At firewall/NAT you have to do port forwarding. If your phone is at port 5060, NAT device will receive a connection and has to know that it is destined for your SIP phone. So, forward port 5060 to the phone. Rudolf - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 6:52 AM Subject: [Asterisk-Users] Why NAT problem hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing not registered. i think asterisk is properly sending request to UA. any commentsthis sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved useragent SJLabs-SJphone/1.40.258 for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Sendmail question
Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ 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] OT: Sendmail question
Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
I think, I just sorted i out. I have to run sendmail with optiosn -bm to be a mail sender. Without it, it seems that sendmail is trying to use outside server for delivery. Without valid username, this will not work... Rudolf - Original Message - From: Rudolf Ladyzhenskii [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:32 PM Subject: Re: [Asterisk-Users] OT: Sendmail question Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] OT: Sendmail question
Old messages are in the queue. I can see sendmail is trying to talk to the remote mail server, but never gets a responce and times out. So message stays in the queue. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:50 PM Subject: RE: [Asterisk-Users] OT: Sendmail question how come you said mail is send out but still in the queue? Does it send out or not? Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Sendmail question Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail -- newbie question
Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3. How do I tell users that they have message? I use Polycom SP300 phones and FireFly IAX phones. I can do it via e-mails, but prefer visual indication on the phone. I have looked at wiki, but did not find answers to all questions. Is there a voicemail setup for dummies type of resource? Any help is appreciated. Thanks, Rudolf ___ 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] Voicemail -- newbie question
Thanks a lot, Will try tomorrow. Rudolf - Original Message - From: Cullin J. Wible [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 10:08 PM Subject: RE: [Asterisk-Users] Voicemail -- newbie question 1) Create the following in your dialplan: exten = 100,1,VoiceMailMain() 2) Set their password to 1234. They can change it in the voicemail menu. 3) See: Getting MWI on Polycom Phones to work with Asterisk http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Ast erisk I don't know about Firefly. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Saturday, August 06, 2005 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail -- newbie question Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3. How do I tell users that they have message? I use Polycom SP300 phones and FireFly IAX phones. I can do it via e-mails, but prefer visual indication on the phone. I have looked at wiki, but did not find answers to all questions. Is there a voicemail setup for dummies type of resource? Any help is appreciated. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] network problem -- echo
Hi, all I have problem with echo. I am running Asterisk server and someone else is running FireFly IAX2 phone. When I talk to this person I get very strong echo on my end. His end is OK. At same time, I was trying to set up someone else with exactly same setup and there is no problem at all. So, looks like this is a network problem at this particular site. He is on ADSL 256/64. I have feeling that it is his router, but this is just a gut feel. I do not have other ADSLmodem /router. Are there any way to troubleshoot echo problems? Thanks, Rudolf ___ 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] network problem -- echo
Thanks for reply. :) 1. Ask if they are using a speaker and mic built into the PC (This will create echo) - solution: Tell them to use a headset 2. Check if they have any output volume (in volume control, advanced, recording) set to record. 3. Check if they have a crappy sound card - solution: get another sound card/usb headset/usb phone The problem is not their PC. I am getting effect as if I am talking to myself over the phone. Very loud and clear. Not the echo effect you get because they use speakers. Rudolf ___ 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] Polycom phone echo question
Hi, all I have Polycom SP300 phones. Calls between those are ok and quality is great. Then I have IAX2 soft phones (FireFly). Calls between those are OK too. But when I have call b/w Polycom (SIP) and IAX, I have really bad echo at Polycom phone side. IAX phone side is OK. Any ideas? Thanks, Rudolf ___ 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] Polycom phone digitmap question
Hi, all I have Polycom SP300 phones. My extension range is 1xx, so I added corresponding entry to the digitmap. By some reason this does not affect on-hook dialing. If I have phone off-hook all is ok. dial extension 102 for example and it connects. if phone is off-hooh, however, I have to press DIAL or take it off hook before number is sent. Any ideas? Thanks, Rudolf P.S. Happens on both SIP 1.3 and 1.5 firmware of SP300 ___ 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] Beginners question -- IAX
Hi, all Can someone point me to a good resource on IAX? I am trying to do two things at the moment: 1. I want to learn 2. I want to conenct MozPhone to my * (wiki does not have much on it) 3. I want to connect two * servers at different locations. I have looked at asterisk wiki and dis not find IAX stuff (may be I did not dig deep enough). Thanks a lot, Rudolf ___ 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] beginners question about extension context
Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a given context Here are my contexts definitions: [from-sip] exten =101,1,Dial(SIP/phone1) exten =102,1,Dial(SIP/phone2) exten =103,1,Dial(SIP/phone3) [iax-user] exten=201,1,Dial(IAX2/phone4) exten=202,1,Dial(IAX2/phone5) If I try to call from IAX2 phone to say ext 102, I get request '[EMAIL PROTECTED]' does not exist I have tried to include iax-user in from-sip and I can make calls from SIP phones to IAX2 ones, but not the other way around. Now for an interesting bit. If I include from-sip in tthe iax-user, all is working fine -- I can make calls in any directions. If I try to do cross-include where one context is included into another and vise versa, IAX2 phone does not even register. Is there a better than include way to route calls between contexts? Thanks, Rudolf ___ 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: RE: [Asterisk-Users] Any suggestions for an IP phone?
No, everything is on local network. Rudolf - Original Message - From: dbruce [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 13, 2005 4:05 PM Subject: Re: RE: [Asterisk-Users] Any suggestions for an IP phone? If the phone is behind a firewall, make sure that port 69 is open so that it can reach the TFTP server. Regards, Derek - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 12, 2005 10:08 PM Subject: Re: RE: [Asterisk-Users] Any suggestions for an IP phone? Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say can not contact boot server. Phones are set to use tftp and correct boot server IP is set via dhcp. I will investigate further, but any suggestions are appreciated. List Receiver [EMAIL PROTECTED] wrote: According to voipsupply.com http://www.voipsupply.com/product_info.php?cPath=95_112products_id=817 --Please Note: Polycom phones are not supported under Asterisk Open Source PBX. Polycom certified platform partners include Path Navigator, Broadsoft, Interactive Intelligence, Sphere, Sylantro, Vertical Networks, VocalData, Alcatel and 3COM. For more information on Polycom supported IP Communications platforms-- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, July 12, 2005 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Any suggestions for an IP phone? We just purchased 4 of the Polycom SoundPoint 301's. We are very happy with them so far. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any suggestions for an IP phone? Polycom SP300 is a pretty good phone. Rudolf Alexandre Leclerc [EMAIL PROTECTED] wrote: Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. -- Alexandre Leclerc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any suggestions for an IP phone? TFTP fixed
Hi, all Stupid me! Under RH (FC3) tftp server is part of xinet. So, I have enabled the tftp server and set all up and I forgot to restart xinet! Dough! Now I am having fun setting up phone. Rudolf - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 13, 2005 11:58 PM Subject: Re: [Asterisk-Users] Any suggestions for an IP phone? [EMAIL PROTECTED] wrote: Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say can not contact boot server. Phones are set to use tftp and correct boot server IP is set via dhcp. I will investigate further, but any suggestions are appreciated. I always use FTP instead, it works famously. Make sure you configure the ftp server in DHCP or in the ftp servers settings, as an IP of course, and that you change the ftp password to the password for the user PlcmSpIp on the server. After that it's flawless. Polycom does not support Asterisk. Polycom, the company, does not support the use of the phones with Asterisk. Who cares? SIP is a standard, we don't need any help from them and we don't need their blessing. The phones are excellent quality and work very well with Asterisk, there's no support issue. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SP300 config files
Hi, all Sorry for not exactly on-topic question I got Polycom SP300 phones. Somehow they did not come with software. I will call them on Monday, but in the meantime, I would like to get them going. I need Polycom configuration template files (phone.cfg, sip.cfg and whatever else they supply). Did not find them on the Polycom site. Can someone e-mail those to me ([EMAIL PROTECTED])? Then I will be able to work on the weekend. Thanks a lot, Rudolf ___ 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] Polycom SP300 config files
Thanks, Rudolf - Original Message - From: Scott Kamp [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 10, 2005 1:00 AM Subject: Re: [Asterisk-Users] Polycom SP300 config files http://www.freedomphones.net/polycom/files/ On Sat, 2005-07-09 at 17:03 +1000, Rudolf Ladyzhenskii wrote: Hi, all Sorry for not exactly on-topic question I got Polycom SP300 phones. Somehow they did not come with software. I will call them on Monday, but in the meantime, I would like to get them going. I need Polycom configuration template files (phone.cfg, sip.cfg and whatever else they supply). Did not find them on the Polycom site. Can someone e-mail those to me ([EMAIL PROTECTED])? Then I will be able to work on the weekend. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind NAT -- SIP config file
Hi, all This is the souktion that worked for me. Here is my config again PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall on Asterisk end is Linux RH9 with iptables. I have set it up to forward ports 5060, 1-2 to Asterisk. Firewall at PHONE 2 end is an off-the-shelf router. Firewall was disabled and I port forwarded port 5060 to the phone. Here is my sip.conf file: (PHONE1 is ext101, PHONE2 is ext102). ; SIP configuration file [general] port=5060 bindaddr=0.0.0.0 context=default externip=my poublic ip localnet=192.168.1.0/24 [ext101] type=user host=dynamic secret=ext101 context=default [ext101] type=peer secret=ext101 host=dynamic context=default callerid=Ext 101 [ext102] type=user nat=yes host=dynamic secret=ext102 context=default canreinvite=no [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 canreinvite=no Hope it helps. Rudolf ___ 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 behind NAT -- SIP config file
Why are the sip.conf extensions mentioned twice each? I am using Polycom SP300 phones. You have to separate 'user' and 'peer' part of it to get it working. Search the wiki for description of the problem. Also, if you * box is behind another firewall, by forward ports 5060 and 1-2 and maybe 5004 from the firewall to the * box will that help on the NAT issue? You have to forward port 5060 so that phone from outside can register and call. And ports 1-2 do that voice can go through. Actual port ranfge is isn filr rtp.conf. 1-2 is the default range If phone 2 is behind another firewall, do you need to forward port 5060 only to that phone? Or some other ports...? Yes, only port 5060. If you do not forward 5060, you can not call this phone from outside. Seem to work OK without other ports being forwarded. Rudolf ___ 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 behind NAT -- SIP config file
Yes, only port 5060. If you do not forward 5060, you can not call this phone from outside. Seem to work OK without other ports being forwarded. You mean on the remote sip phone firewall? What if there arem ore than 1 sip phone on that network behidn that firewall? Then you are in trouble. Asterisk only sees single public IP address. As far as it concerns there is only single phone out there. If you get multiple phones working, let me know. Another option, I think, may be using VPN, but I have not tried that. Then you can potentially have remote SIP phones to be on the virtual network. Don't you need to forward ports 1-2 for voice? Or does the sip phones just open up the ports from inside (by doing the in to out calls and keep alives)? I have mot tried to sniff on the traffic in details. I think, other ports are opened in responce to connection on port 5060. The only port listens at is port 5060. Rudolf ___ 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] More NAT questions -- SOLVED
Hi, all Got it to work finally. Thanks to all. Had to add [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is Actually, I had 'externip' before, but I have added 'localnet' one. I also had to do port forwarding on the NAT near to PHONE 2 to pass port 5060 to the phone. This is needed if you ever want to call this phone. I can e-mail my sip.conf to anyone who is interested. Rudolf - Original Message - From: Julian J. M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 03, 2005 4:11 AM Subject: Re: [Asterisk-Users] More NAT questions In you asterisk sip.conf: [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is If you don't externip, externip will never be used, because asterisk won't know WHEN to use it. Also, define canreinvite=no in your sip phones sections, as was suggested above. Julian J. M. On Wed, 2 Mar 2005 23:26:56 +1100, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. Any help is greatly appreciated. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More NAT questions
Hi, all Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. Any help is greatly appreciated. Thanks, Rudolf ___ 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] NAT/Routing problem
Hi, all I have Asterisk here and SIP phone sitting at another location. Initially, I had problems registering the phone. Now I have added 'nat=yes' for this phone in sip.conf and phone registers. However, I can not make calls. SIP debug shows that phone registers with public IP address of the site, while calls somehow go to local address. Here is an example of SIP debug message: -- Registered SIP 'ext102' at 147.10.78.157 port 8103 expires 3600 -- Attempting native bridge of SIP/ext102-26a4 and SIP/ext101-1b49 Feb 27 17:02:31 WARNING[3160]: chan_sip.c:755 retrans_pkt: Maximum retries excee ded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Res ponse) As one can see, public IP 147.10.78.157 is used at registration time, while private IP 192.168.1.2 is used for communicating with phone. Remote site does not have firewall. My site does, but I could not see anything wrong there. I have turned on logging on firewall and no suspicios activity goes on. Any help is appreciated. Thanks, Rudolf P.S. Here is extract from my sip.conf file: [ext102] type=user nat=yes host=dynamic secret=ext102 context=default [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 ___ 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] NAT/Routing problem
Thanks for suggestion. Unfortunately did not work. What does this option do anyway? Rudolf - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 27, 2005 8:18 PM Subject: Re: [Asterisk-Users] NAT/Routing problem On 19:45, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote: As one can see, public IP 147.10.78.157 is used at registration time, while private IP 192.168.1.2 is used for communicating with phone. [ext102] type=user nat=yes host=dynamic secret=ext102 context=default [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 try adding: canreinvite=no -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SP300 problem solved
Hi, all Registration problem is solved now. I did not realise phones also have web interface. I used that to set up SIP server and authentication. Settings on the phone itself do not have all the options. Rudolf ___ 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] Seting up for afirst time -- can not call
Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with polycom phones. Here is my sip.conf file: ; SIP configuration file [general] port=5060 bindaddr=0.0.0.0 context=default [polycom_sp300_ext101] type=user host=192.168.1.101 secret=101 context=default [polycom_sp300_ext101] type=peer secret=101 host=192.168.1.101 context=default callerid=Ext 101 [polycom_sp300_ext102] type=user host=192.168.1.102 secret=101 context=default [polycom_sp300_ext102] type=peer secret=102 host=192.168.1.102 context=default callerid=Ext 102 First question is about the secret. Should I set up something on teh phone? Is it phone password (default 456)? Now, I am trying to have some extensions. So I have edited the extensions.conf file and changed the [default] section: [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include = demo exten = 101,1,Dial,(SIP/polycom_sp300_ext101) exten = 102,1,Dial,(SIP/polycom_sp300_ext102) The rest of the file is as is as it came with Asterisk. Now I run 'reload' command as CLI. Is ist all I have to do to be able to call between those two phones? If I try to call from one phone to another, after I enter first two digits '10', I get connecting on phone screen and instant busy tone. Any help is greatly appreciated. Thanks, Rudolf ___ 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] Polycpm SP300 problems
Hi, all I am trying to connect Polycom 300 to Astersik. I do not want to use FTP server for now, so I am tryng to set up phone manually. Network configuration parts is OK, except that it does not ask for SIP server address. Any ideas where to set this? Also i have some problems with setting up authentication. There are settings for user name and password. How does one delete characters? I could not find any way to do backspace or delete! And last question. Are user name and password on the phone should be same as user and secret in sip.conf file? Or those two are different things? Thanks, Rudolf ___ 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] OT - C structure question
Hi,
This is something that I tought of BEFORE I had my morning cofee.
You can use array where your 'variable points to an array element.
You can have array of 'void *'. In this case, an array element can point to
anything.
for example:
int field1;
char *filed2;
struct MyData field3;
void * array[NUMBER_OF_ELEMENTS];
array[0] = (void *)field1;
array[1] = (void *)field2;
array[2] = (void *)field3;
Now you can have functions to select data:
int GetFiled1()
{
return *((int *)array[0]);
}
char * GetFiled2()
{
return (char *)array[1];
}
You got the idea.
Rudolf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk
Sent: Friday, February 25, 2005 3:42 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] OT - C structure question
I hae tried searching the web for the answer, but, man is there a lot of
pages ... :(
in the language I develop in, if I have a structure I can dynamically
refer to the contents of a field of the structure like so:
MESSAGE SomeStructure:Field(SomeFieldName):Value
where SomeFieldName is either a quoted constant or a variable expressions
In C, I beleive that you can refer to the contents of a field in a
structure like so:
chan-context or
chan-exten
Is is possible to refer to these fields like
chan-(variable) where variable is either context or exten or an
expression that resolves to a valid fieldname of the structure ?
My reason for asking is that I want to create an application that would
take a channel and a field name and return the value of the field.
for example
GetChannelData(context)
GetChannelData(exten)
and I didn't want to have to declare a massive case statement, and have
to modify the app everytime some new fields were added to the structure.
I know that some of these variables are already exposed, but was wanting
to get some other values.
Julian
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RE: [Asterisk-Users] TFTP Server
Hi, setup is in /etc/xinet.d/tftp file Default directory is /tftpboot. make sure that this directory is readable by anyone. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary G. Hendershot Sent: Wednesday, February 23, 2005 9:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] TFTP Server On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader files -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 22, 2005 1:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] TFTP Server G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks Ferg ___ 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
RE: [Asterisk-Users] TFTP Server
Any directory name is fine as long as you configured TFTP server to use it. Also, from device (phone) point of view, your /TFTPBOOT directory is '/' (root) directory on server! Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ferguson, Michael Sent: Wednesday, February 23, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TFTP Server I created a different dir, /SIPFONE Now I have to check if it readable by all. Thanks. I set my Windows 2003 DHCP to assign the TFTP server's IP address, default gateway, dns, etc, etc and the phone got all that quite well but not picking up the files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Tuesday, February 22, 2005 5:25 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TFTP Server Hi, setup is in /etc/xinet.d/tftp file Default directory is /tftpboot. make sure that this directory is readable by anyone. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary G. Hendershot Sent: Wednesday, February 23, 2005 9:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] TFTP Server On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader files -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 22, 2005 1:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] TFTP Server G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks Ferg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Minimal hardware requirements
Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Thanks, Rudolf ___ 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] Minimal hardware requirements
Thanks. I guess, I will have to try it and see. Mine is one of those small form factor COMPAQ boxes. I will try to get full specs from COMPAQ/HP. What about load Asterisk puts on processor if you do, for example, IP-IP call and IP-PSTN call? Since I will use Polycom phones, I will use SIP. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Novack ( Mozilla - portable ) Sent: Tuesday, February 22, 2005 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Minimal hardware requirements Rudolf Ladyzhenskii wrote: Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Thanks, Rudolf Depends. If you plan on using the TDM400 with one each FXS and FXO, the MB needs to have PCI Ver 2.2 slots, or the card won't be seen Any MB made after 2000 probably is OK John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modem as PSTN interface?
Hi, all Can a normal PCI modem be used to provide PSTN interface? I have seen modems that have answering machine capabilities, so there should not be a problem sending voice through them. Certainly, modem will be cheaper option then dedicated cards. Am I missing something? /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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] Modem as PSTN interface?
Thanks. There goes a good idea ;=( Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PHP Mechanic Sent: Monday, February 21, 2005 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Modem as PSTN interface? Hi, all Can a normal PCI modem be used to provide PSTN interface? I have seen modems that have answering machine capabilities, so there should not be a problem sending voice through them. Certainly, modem will be cheaper option then dedicated cards. Am I missing something? Most modems don't operate at full-duplex. A normal modem can be used to send voice, then switch to recording voice, but it can't send and receive simultaneously. ___ 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
RE: [Asterisk-Users] Which IP phone to use in Australia
Thanks for the info. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Elvish Sent: Tuesday, February 15, 2005 8:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which IP phone to use in Australia Hi guys, I haven't had the opportunity to play with any Polycom products, although they will probably be the best IP phone available. I have used the Grandstream BT-101/102, the HOP-1003 (upgraded 1002) and Zyxel telephone adapters. My recommendation out of the tried ones would be the Grandstream BT-102 where the phone is on a closed network. My problem with the HOP is that there is no transfer button, this can be worked around with key sequences (*2 for attended transfer, # for unattended transfer) and (if I got it working) the flash key but they aren't optimal in an office setup where un-techno people want to just transfer a call with a button and the overall receiver volume seems to be lower (definitely than the analogue adapter). The HOP does have one advantage: it only beeps once when you get an incoming call along with a message on the display unlike the BT-101/102. The Zyxel equipment works great for us and it supports callerID (only the number) but it does mean the cost of an analogue phone. If you have an issue with network cabling, the Zyxel is good because you can run two analogue phones from one network cable, and there is a hub built into the back which allows inline connection to a PC. This having been said, because they are analogue, you need to use the *2 and # keys for transfer etc. I would be interested in trying (and have been offered but haven't had the time) a new Sipura telephone, it includes two line indicators. Whilst this isn't ideal for a receptionist (I think Snom would be the only option for a receptionist with the additional indicator panel) it would be nice for a general office worker who needs one direct line and one general ring group / reception button. They can be expanded to four lines with some sort of software upgrade. One thing to be aware of - where you will be setting up standalone offices (i.e. one person at home behind DSL) you should consider the firewalling etc but seeing as the original question came from the experts - they should be able to sort it out! Our experience is that some hardphones will even have troubles with specific firewalls, yet they will work quite happily with a Billion style product (sorry to bring that up). Hope this helps. Kind Regards Stuart On Tuesday, Feb 15, 2005, at 15:33 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-15 at 18:05, Adam Goryachev wrote: On Tue, 2005-02-15 at 17:54 +1100, Howard Lowndes wrote: On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote: Personally, I quite like the polycom phones such as the IP300 and IP600 I've never really bothered with the IP500... There are a few issues I have with them though, the main one being that I can't disable call waiting on the phone. There are workarounds for this though (in asterisk dialplan). ...which is something to be said for the HOP 1002 - you can disable call waiting. Have you actually used the polycom phones? If so, how do they compare to the HOP 1002, or, would you call the polycom IP600 and HOP 1002 exactly equivalent in all respects except for the call waiting factor? Unfortunately I have never used, or even seen the polycom phones, so I cannot comment on the comparison. I do know that the HOP 1002 serve my purpose and are quite robust. There was a date issue with the software pre v1.41.007 and I have found out how to get a brand name to display on the screen. I have also discovered that, under SIP at least, the phone will only display the caller ID number and not the caller ID name, though that latter is not often sent anyway except for calls from mobiles as MOBILE. Basically they are very robust, almost brick shithouse robust. :) The online manual is about 47 pages of Chinglish which is an Alexander (downer). (Oz joke there for all you yanks) The only down side that I can see is that the 2 port hubbing is only 10 mbps which shouldn't really be a problem for most users who connect their PC in line, but could be a real bummer for the power user PHBs who want to do gaming. I've not seen/used the HOP 1002, I just find it hard to accept that it would be as good as the polycom IP600 phones Note: I would be *pleasantly* surprised if you say it is as good! Regards, Adam -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states.
RE: [Asterisk-Users] Which IP phone to use in Australia
Thanks. I'll see you tomorrow. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Hales Sent: Wednesday, February 16, 2005 9:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Which IP phone to use in Australia I also forgot to add: Zyxel wireless sip handset Sipura Will add more as we think of them! Later, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, 16 February 2005 9:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Which IP phone to use in Australia To improve on my a bit too off-the-cuff answer from earlier (too busy a day to be fully helpful!) We are having an Asterisk get-together this Thursday. It was announced on the list here, and I will re-post the data below. Myself and the tech who set up our asterisk system have tried and tested a lot of phones: Zultys Grandstream Polycom Snom 3com Samsung Leadtek And software phones Installed and using ISDN (ETSI), analogue cards and Mobile Phone gateways (telular brand) Regards, regards, PaulH Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for a casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. Ultimately, I think it would be interesting and useful to turn this into a monthly get-together, so I'd like to talk about that too. Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families. When: Next Thursday evening, the 17th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place. I'm going to try and get us a table, and put an old analogue phone on it, so you'll know how to find us. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED] Hope to see you there! ...jurgen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Elvish Sent: Tuesday, 15 February 2005 8:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which IP phone to use in Australia Hi guys, I haven't had the opportunity to play with any Polycom products, although they will probably be the best IP phone available. I have used the Grandstream BT-101/102, the HOP-1003 (upgraded 1002) and Zyxel telephone adapters. My recommendation out of the tried ones would be the Grandstream BT-102 where the phone is on a closed network. My problem with the HOP is that there is no transfer button, this can be worked around with key sequences (*2 for attended transfer, # for unattended transfer) and (if I got it working) the flash key but they aren't optimal in an office setup where un-techno people want to just transfer a call with a button and the overall receiver volume seems to be lower (definitely than the analogue adapter). The HOP does have one advantage: it only beeps once when you get an incoming call along with a message on the display unlike the BT-101/102. The Zyxel equipment works great for us and it supports callerID (only the number) but it does mean the cost of an analogue phone. If you have an issue with network cabling, the Zyxel is good because you can run two analogue phones from one network cable, and there is a hub built into the back which allows inline connection to a PC. This having been said, because they are analogue, you need to use the *2 and # keys for transfer etc. I would be interested in trying (and have been offered but haven't had the time) a new Sipura telephone, it includes two line indicators. Whilst this isn't ideal for a receptionist (I think Snom would be the only option for a receptionist with the additional indicator panel) it would be nice for a general office worker who needs one direct line and one general ring group / reception button. They can be expanded to four lines with some sort of software upgrade. One thing to be aware of - where you will be setting up standalone offices (i.e. one person at home behind DSL) you should consider the firewalling etc but seeing as the original question came from the experts - they should be able to sort it out! Our experience is that some hardphones will even have troubles with specific firewalls, yet they will work quite happily with a Billion style product (sorry to bring that up). Hope this helps. Kind Regards Stuart On Tuesday, Feb 15, 2005, at 15:33 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-15 at 18:05, Adam Goryachev wrote: On Tue, 2005-02-15 at 17:54 +1100, Howard Lowndes wrote: On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote: Personally, I quite like the polycom
RE: [Asterisk-Users] Which IP phone to use in Australia
You ahve to run Linux anyway. TFTP is very easy to setup. Rudolf You'll need a TFTP server to get the SIP firmware on the phone. For small deployments you can configure the options on the phone itself, but for anything more than 2 phones, I'd recommend a TFTP server. Stuart -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which IP phone to use in Australia
Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Thanks, Rudolf /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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] Which IP phone to use in Australia
I bet it will be! Can you provide more info? I can not find it on the Asterisk website. Thanks, Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Hales Sent: Tuesday, February 15, 2005 5:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Which IP phone to use in Australia The Asterisk meeting in Melbourne Thursday night would be a good place to discuss this! Regards, regards, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Tuesday, 15 February 2005 5:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which IP phone to use in Australia Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Thanks, Rudolf /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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
RE: [Asterisk-Users] Which IP phone to use in Australia
Well, I did not know about it until today (I only joined mailing list today!). But I did work with voip before. I am Melbourne based, so I am VERY interested. Rudolf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Howard Lowndes Sent: Tuesday, February 15, 2005 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Which IP phone to use in Australia On Tue, 2005-02-15 at 17:26, Paul Hales wrote: The Asterisk meeting in Melbourne Thursday night would be a good place to discuss this! Not if: 1. You don't know about it 2. You're not Melb based. Regards, regards, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Tuesday, 15 February 2005 5:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which IP phone to use in Australia Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Thanks, Rudolf /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
