Re: [Asterisk-Users] IPP g729 x86_64
I'm curious, how are you licensing your codec? The source is open, but the codec usage licensing is not. I think you'll find that licensing it from Digium will be much simpler, not to mention their code will Just Work(tm) without any messing around. -d At 12:08 PM 4/17/2005, you wrote: Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ 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] Petition for IAX firmware
Hi all, I've put together a quick petition, in hopes that we can possibly persuade Sipura (or any other large-scale IP handset manufacturer) to include firmware support for IAX. The IAXy has proven that an IAX product is in demand, and very useful, and I think we'd all like to see a handset manufacturer follow Digium's lead. I'm not particularly endorsing Sipura, however I do know that they have seriously considered support for IAX, and have decided to hold off until the demand is there. I'm hoping that with some numbers, we can prove to them that the demand is already here, and that IAX is already a viable technology. I'd like to encourage everyone to show your support -- hopefully Sipura, and/or other manufacturers will see these hard names and numbers, and realize it's time to move something into production. Petition: http://www.petitiononline.com/IAXPhone Thanks, -d ___ 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] Webmin
First, I suppose, you'd have to write it .. AFAIK, the webmin project was abandon a couple of years ago. I don't think it was ever even remotely near being completed. -d At 05:25 PM 3/31/2005, you wrote: How do I install the asterisk module for webmin? ___ 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] Hold Pickup
Look at bkw's valet parking -d At 03:58 PM 3/21/2005, you wrote: I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial prefix12345 to pick up a call holding on ext. 12345. It looks like the closest to what I want (without changing Asterisk) would be Park followed by an AGI that pokes my manager-port client (which in turn would redirect the target extension's channel to...well, something) -- if the parking space was returned to the dialplan somehow (or if Park() didn't ignore its arguments). At the moment, I'm using the v1-0 branch and at this point it looks like our phones will be all SIP (i.e., madding chan_sip to make sendonly channels visible to the dialplan somehow isn't automatically out of the question). -- Joshua P. Dady ___ 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] Hardphone deployment recommendation
Isn't the idea of this, to sort of use the desk as it's plane to pick up sound? I thought a few vendors did this kind of thing... -d At 03:33 PM 3/7/2005, you wrote: The microphone is [somewhat inexplicably] mounted in the base over a hole that faces downwards, between two of the rubber feet, its like a 'U' with a dot inside it. ___ 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] simpletelecom.com??? are they a SCAM?
I don't know - they look kinda lame. I mean, why is their SIP server
seemingly better-routed than their IAX server? In my case, their IAX server
is almost 20ms further away than the SIP one -- seems odd to me.
Think I'll stick with Nufone - very well routed, and only ~15ms away. :)
-d
At 06:37 PM 2/19/2005, you wrote:
I have been using simpletelecom.com for over 2 months now to make
outboudn long distance calls, I didn't have any problems what so ever
with them.
To send callerid this is how I do it:
exten = 81NXXNXX,1,SetCallerID(MY NAME 1235551234)
exten = 81NXXNXX,2,Dial(SIP/${SIMPLETELE}/${EXTEN:1},60,tr)
exten = 81NXXNXX,3,Congestion
Caller ID will be passed like this, however in most cases the name is
not passed along but looked up in an SS7 database against the number
passed along and that name is displayed, meaning you can't have your
name changed.
On Sat, 19 Feb 2005 13:31:12 -0500, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On February 19, 2005 10:56 am, Madhawa wrote:
Hi List!
any body use www.simpletelecom.com?
so anyone here has experience with them? are they a SCAM?
This is -biz material. Spew your is this a scam bullshit there.
-A.
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RE: [Asterisk-Users] asterisk@home scary log
Why would you even want SSH exposed to the world? In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? -d At 10:08 AM 2/10/2005, you wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiport Fax over softphone
Hi Tim, No hardware - been done, see rxfax. Dumps it to a tiff, you can do whatever you want with it (email it out, convert to pdf, send to a printer .. OCR and voice to speech it and play it over the PA system ... :) -d At 01:07 PM 1/31/2005, you wrote: How hard would it be to write (or has this been done?) a module that would receive a fax and save or deliver the file? Would you need any hardware at all ? ___ 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] Multiport Fax over softphone
I assume he meant no fax-capable card, or anything with fax DSPs, as that's generally needed. Having seen his previous posts, I assume he's more familiar with * than to assume the phone line will telepathically find it's way into the server. -d At 01:26 PM 1/31/2005, you wrote: No Hardware ?? u only so whats going to determine its a fax ? you need a dedicated box if you have no line card in the box. asterisk + SIP is not capable of determining whats a fax and whats voice. you need a card unless you have a dedicated number On Mon, 2005-01-31 at 13:12 -0600, denon wrote: Hi Tim, No hardware - been done, see rxfax. Dumps it to a tiff, you can do whatever you want with it (email it out, convert to pdf, send to a printer .. OCR and voice to speech it and play it over the PA system ... :) -d At 01:07 PM 1/31/2005, you wrote: How hard would it be to write (or has this been done?) a module that would receive a fax and save or deliver the file? Would you need any hardware at all ? ___ 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 -- skamp [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
Re: [Asterisk-Users] G.729? Worth it?
At 01:49 PM 1/19/2005, you wrote: There are systems that use G.711 when traffic is light, but switch to compression codecs under heavy traffic to conserve bandwidth. I don't know how/if this can be done in Asterisk. --Stewart I don't think there's anything like that built into * as it is now, but it would be pretty trivial to write a script to handle such a thing. Effective monitoring of the traffic conditions may require a bit more work .. monitoring latencies, and maybe monitoring current utilizations via snmp on your core routers/switches/etc. A quick probe to the management interface would also give you some insight on current call volume, of course. -d ___ 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] long delays in list posts?
So? it's a big list .. I'm sure if you'd like to donate some quad xeons and gigE pipes, it could be resolved very quickly.. -d At 04:41 PM 1/13/2005, you wrote: OMG! 1 hour?!?! I just now got this at 4:40PM. It takes an hour for my emails to get posted to the list? Geez.. -Matthew - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 13, 2005 3:24 PM Subject: [Asterisk-Users] long delays in list posts? Hey guys, I sent an email to the list at 2:57PM central. I just now see it on the list, and its 3:23PM. Anyone else experience this? I am sending this email at 3:24PM central. Lets see when it gets posted to the list. -Matthew ___ 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 cancelation on Digium T1 cards
That's data or fax CNG, not dtmf. And yes, it's disabled for the duration of the fax or data session. -d At 11:36 PM 1/10/2005, you wrote: Hello all, I am getting console debug messages about tone detected on channel XX, disabling echo cancelation on channel XX when using echocancel=yes with a Digium T1 card. does this mean that DTMF breaks the echo can? Does Asterisk permanently disable the echo can or is it for that channel instance only? ___ 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 1.0 released
Old news, Asterisk 1.0 released .. :) Here's another mirror -- should be very fast from most anywhere. Take it easy on Digium's bandwidth. :) http://asterisk.paperwork.com -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 1.0 released
You can snag em from http://asterisk.paperwork.com and if you drop me a note with your url, I'll add it to the list. -d At 10:11 AM 9/23/2004, you wrote: If anyone who got the 1.0 tar's would be able to get them to me I'd be more than willing to donate traffic toward the effort by mirroring it on some bandwidth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 1.0 released
hehe .. I think we have more bandwidth than sourceforge now.. I've got like 9 on my list now. -d At 10:58 AM 9/23/2004, you wrote: Maybe someone should make a bittorrent? I will contribute some BW if there is a torrent. Steve Kenneth Shaw wrote: To be Slashdotted within 30 minutes. -Ken Shaw... On Thu, 2004-09-23 at 07:28, Lex Lethol wrote: Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 , ISA Firewall/VPN , STUN and other
Keep in mind, PPTP will only tunnel through the NAT, as long as GRE (prot 47) is properly tunneled along with tcp 1723. This support is relatively standard in common NATs, but it's not a given. -denon At 09:23 PM 9/21/2004, you wrote: First, I assume that you will be running NAT at both locations, if that is not the case, then the configuration will change. When you said VPN, are you using PPTP or IPSEC? Microsoft supports PPTP. In order to connect a PC over VPN to the office, which has a PPTP VPN Server, you will need to runs VPN software. After it is run, the PC obtains a new IP address from your office, If you are using soft phone on a Windows PC, you can communicate without any problem as the IP is now tunnelled via internet to the office from your PC. Please note that PPTP VPN can tunnel through NAT and should be allowed to tunnel through Firewall as well before soft phone can be used. From your description, I feel that IPSEC may be a better solution. IPSEC can route/merge two remote subnets together. If you are connecting two sites using IPSEC, you can start any service and a hardware SIP phone should be fine. Just make sure your DNS and gateway setups are correct. And your PC can see across the IPSEC tunnel without loading any VPN software. Henry -- From: Shawn Dillon [mailto:[EMAIL PROTECTED]] Sent: Tuesday, September 21, 2004 2:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk , ISA Firewall/VPN , STUN and other issues I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone , on the SIP client nothing. The Debian machine has two NIC's , one with a static external IP and one with an internal IP. Our remote offices are behind a mixture of firewalls. I have some questions with regards to our testing and setup. 1) Is there a way to get the SIP/IAX client to work via the VPN? This would be the easiest way. 2) If not can I install a STUN server on the same machine as the * server? Can it use the same internal and external IP's as the * server? 3) Is there a hardphone that supports VPN that has been tested? 4) What is the best hardphone to use with Asterisk? Thanks for the input Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] The CISCO 7940 Tranfer Button..
I agree entirely! I've been told it's not possible, but I'd love for someone to prove me/them wrong. -d At 11:54 PM 8/10/2004, you wrote: Hi, I was wondering if any CISCO users out there knows if it is possible to Change the locations of the BUTTONS along the bottom of the screen. I ask this as the TRANSFER button is only accessible after pushing the more button. This is a pain as it's the MOST used button. So having to push two buttons to get to it is a real pain. Thanks, James Gardiner ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] I wanna buy a new X100P
Sure, head to : http://store.yahoo.com/asteriskpbx/wildcardx100p.html -d At 12:25 PM 12/30/2003, you wrote: I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD problems
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I had one before, but I lost
it.. :)
I've got it running through Asterisk - all working fine from a SIP
standpoint. I can dial FWD numbers like 612/613/etc and everything
works. However, if I dial *18005551212 or *408xxx (say, a USA number),
I either get a fast busy or a This service is only available to FreeWorld
Dialup members.
Am I missing something? I signed up, got my password .. the sip is
registered, firewall is open, no NAT, etc. I've tried a variety of combos
in dialing/etc .. to no avail. Is my account pending some type of
activation or such?
Possibly / likely related, it seems that the * doesn't work when I'm
trying to set up the voicemail either. I'm using a Cisco 7960 (but
remember, it's actually Asterisk linking us together). The 7960 does have a
* dialplan, so that shouldnt be an issue.
Any ideas you guys have would be great!
Here's what my sip.conf looks like:
register=9:[EMAIL PROTECTED]/453
It shows that it's registered in sip sho reg..
[fwd]
type=friend
secret=password
username=9
fromuser=9 ; I dont need this .. but worth a shot, tried with and
without
nat=yes ;I'm not behind nat, but I thought I'd try it
anyway
fromdomain=fwd.pulver.com ; Don't need this either. .but what the hay
host=fwd.pulver.com
canreinvite=no ; worth a shot, right?
reinvite=no
I then have an extension that does:
exten = _7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
-d
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[Asterisk-Users] Call Recording
Hi all, I know there have been some neat changes to how call recording works lately, and so therefore it's much easier to do .. but I haven't been able to find any details in the archives about the best way to accomplish it these days. Does anyone have a sample of their config they'd share, or a macro they use? I'd really appreciate it .. -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue only ringing one agent at a time
That works better here in Minnesota .. Land of 10,000 Lakes .. go pick one and jump. :) -d At 05:16 PM 12/12/2003, you wrote: Instead of quacking out useless information, it's more useful to not answer. Their are alot high places on this planet.. pick one and jump. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP (peer to peer?)
Depends if you're phone supports it, and you have reinvites etc enables in *. -d At 03:17 PM 12/8/2003, you wrote: Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer)?xml:namespace prefix = o ns = urn:schemas-microsoft-com:office:office / Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP expert. Thank you for your help. Wim
Re: AW: [Asterisk-Users] ADSI Programs
Some ADSI phones come locked to a certain service provider. You cannot load your own adsi scripts into these phones - you need one that isn't tied to a specific company or pbx. -d At 06:35 PM 8/27/2003 +0200, you wrote: Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog lines
Adtran 750 channel bank and a T100P. -d At 01:06 PM 8/19/2003 -0300, you wrote: Hello, I am looking for hardware for Asterisk. I want to connect analog lines (from 6 to 12 or more) to Asterisk, what will be the best hardware for that? Thanks, Bartosz
RE: [Asterisk-Users] Grandstream Budgetone
Are these locked to the service, though? Look what vonage managed .. :) -d At 12:36 PM 8/17/2003 -0400, you wrote: $75 for the single ethernet port version and $85 for the dual ethernet port version. You can get two for $129 at www.sipphone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer, Budgettone 100
Last I checked, SIP transfer to park doesn't work .. only way to do it is using T and a # transfer .. which is ugly. Has this been fixed? -d At 10:51 AM 7/30/2003 +0200, you wrote: park the call On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote: hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and then the new extension .. cheers Dave --- Email sent using AnyEmail (http://netbula.com/anyemail/) Netbula LLC is not responsible for the content of this email ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone System Questions
Yes, you can do all this .. an X100P and two TDM400s, one with all 4 FXS, one with only 1 FXS. -d At 07:53 PM 7/12/2003 -0400, you wrote: Hi all, I need a phone system that has 2 incomming lines and 5 extentions. I will need music on hold I need all incomming callers to get a message: press 1 for x 2 for y, etc I need extentions to be able to make outgoing calls on the 2 lines Can Asterisk do this, if so, what hardware do I need? Michael Hess Swirltech ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP immediate hangups with latest CVS
I had this a while back, and set canreinvite=no, and it fixed it. -d At 08:42 PM 7/11/2003 -0700, you wrote: I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960 (non-NATed) - RH 8.0 - Asterisk CVS update as of ~8:00 PM EDT - full make clean; make install on [asterisk,zaptel,libpri] - 2ghz box with E1 card (that's pretty much not part of the equation) I have boiled the configuration down to an extremely (_extremely_) simple setup, and it does not work. SIP calls from the 7960 are hanging up almost immediately, with no audio getting through. It seems that the hangup happens just after the moment that the 7960 sends the ACK message (judging from the debug below, at least.) I have verified that demo-congrats is there, as my original problem stemmed from strange behavior with Zap dialing, and I kept simplifying, so this is the culmination of winnowing down the options to the most basic config. The same phone works flawlessly with other lines that are configured on it to other * servers. Here is my entire relevant configuration. It's as simple as you can get, really. I dial 14109850123 (as a test number - it matches the _1X. list) and I get an almost instant hangup. --- ;sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls dtmfmode=rfc2833 allow=all [3015321510] type=friend username=3015321510 secret=fluffernutter host=dynamic context=from-sip allow=all --- ;extensions.conf [general] static=yes writeprotect=yes [from-sip] exten = _1X.,1,SetCallerID(3015321510) exten = _1X.,2,Answer exten = _1X.,3,Playback(demo-congrats) exten = h,1,Hangup exten = t,1,Hangup exten = i,1,Hangup --- Other strange notes: - quite often, when launching with -gcd I get a segfault. I have the cores, if anyone is interested. - I have almost identical systems (same hardware, same MB, etc.) churning away with no problems with slightly older revs of code *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 128.151.224.33 : 5060 (non-NAT) Found audio format 0 Found audio format 8 Found audio format 18 Found audio format 101 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=2c9c06be Content-Length: 0 to 128.151.224.33:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 102 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=3015321510,realm=asterisk,uri=sip:64.33.1.8,response=4a9e7d0429571ec4047634179fc43f2d,nonce=2c9c06be,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 247 Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33
Re: [Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
At 04:11 PM 7/3/2003 -0400, you wrote: We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be there for a few months. We'd like to go ahead and replace the voicemail system with Asterisk now, and as we're ready, drop the Merlin system. My questions: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? Unless I'm mistaken, you just name those lines appropriately, then have matching extensions in your extensions.conf, like any other trunk. The difference being, these lines go straight to an auto attend. You could pass the extension to it via a dialed string, if need be. When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? The dialed extension, or maybe a callerid string.. -denon Thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minimum budget question ...
What'd this device set ya back? Have a url? -d At 11:45 PM 6/30/2003 -0700, you wrote: AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with SIP up and running with * Michael Kane wrote: The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16 FXS and 8 FXO. I've delpoyed hundreds of these IAD's signaling with MGCP. Not sure if it supports SIP yet. Hope this helps... Mike Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 508-295-2826 - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 2:51 PM Subject: Re: [Asterisk-Users] Minimum budget question ... Hi Tan, Thanks for the reply. I'll end up asking a load more questions now... What sort of prices are we talking about for the 24 port VoIP gateway? I assume that each port is individually addresable by *? As I recall the 24 port gateways tend to be terminated at the FXS side as some 'wierd' connector (wierd in that it's not rj45/11) do you just wire this to a patch panel? What codec is in use to get all 24 ports 'running' at the same time..G729? Does this cause problems since iirc * needs to run in console mode for the G729 codec to work properly Thanks for the info... interesting site too :D Andy *** REPLY SEPARATOR *** On 30/06/2003 at 19:21 Tan Aks wrote: Hi, We provide asterisk-based solutions to customers based in the uk. One of our customers (9 users) is trialling our low-end solution which comprises of a box with 2 x X100P (analogue line) cards installed, and a voip carrier for outgoing calls. This customer intends to have 13 extensions in his live scenario. The way to use multiple analogue phones is: 1) get a T100P card and use a T1 channel bank sourced from the US 2) use a couple of TDM400P cards to give 8 extensions, and use IP phones for the other extensions 3) use a voip gateway to provide up to 24 x analogue extensions per IP address. VoIP gateways are commonly available and convert analogue lines into a SIP/H323 VoIP stream. You can get an E1 terminated with an RJ45. If you have a coax termination then you can use a balun to get rj45 connectivity. Hope that helps. Tan (telappliant.com) - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 5:26 PM Subject: RE: [Asterisk-Users] Minimum budget question ... Tim, a good comprehensive answer to the question...certainly gave me a few things to think about. I do have a few questions though, since I'm in Europe. Has anyone in Europe set up something equivalent to what Tim suggested? What sort of prices did it work out at? How did you solve the channel bank 'issue' in Europe? I keep reading that E1 lines are coax terminated, is this correct or do you usually get a choice from your teleco? Were there any other issues to contend with? I'd certainly be interested in the experiences of anyone in Europe... Thanks Andy On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote: If this is for commercial use, especially if you are going to be selling this solution, I would suggest that you don't even offer the choice of analog lines except in the smallest of offices. Unless you like to spend a lot of unbillable time supporting them :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAC Access Bank
I just picked up a couple CAC Access Bank 1s loaded with FXS that should be arriving shortly. Does anyone have one that they use with Asterisk? If so, would you be willing to shoot me a note with your current configs? I'm not very familiar with CAC/etc, and it would save me countless hours of muddling through .. :) Any others tips you can give me on these banks would be very appreciated. Also, any cheap source for FXOs on them? I'm told they're hard to come by .. Thanks again, -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid time set
Maybe the hardware date? -d At 05:07 PM 6/17/2003 -0400, you wrote: Where is this time that is sent to the phones with the callerid info coming from ? If I do date at the command line I get the correct time as set by ntp yet the time the phones get set to is 50 minutes slow. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Survey
Hi all, I've put together a quick Asterisk survey, in an effort to attempt to get some insight into the overall direction of the project. I'd appreciate if you could spend a couple minutes and run through this - I'll be happy to share the detailed results with Digium or anyone else who's interested. It's a quick hack, but hopefully it'll answer some lingering questions. The survey is at: http://denon.cx/asterisk Thanks! -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Hardware - Channelbank vs SIP etc
We're doing a new * installation at a remote office soon, and I was just curious what people's opinions were on hardware these days .. I've had decent luck with T100Ps and Adtran, but I know times change .. I'm looking to do roughly 15 handsets and 15 pstn, with some room to grow. I had planned on two T100Ps and two adtran 750s, one for handsets, one for pstn. I'm thinking of going SIP on the other side, though. I've been looking at the Grandstream budgetone phones, as well as their handytone. Anyone have anything good or bad to say on these? Cisco is out of that office's budget, I'm afraid. We're replacing a cheapo key system there, so it's all about the benjamins.. :\ I was also looking at: http://clipcomm.co.kr/eng/e_product/e_product_voip_analoggateway_4.html (rumored to be D-Link's OEM?) and http://www.yoda.com.tw/SOLUTIONS/vg422r.htm Any thoughts on these? Has anyone had good luck with other low-cost channels banks? (noo, not Zhone.. :) Any tips are appreciated, you can catch me here or on irc as always .. -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Parking on 7960
Hi all, I've got a fairly minor question, but it's getting on my nerves .. hopefully it's an easy answer. I'm having trouble parking calls on our 7960s. It works fine on ZAP devices, though, and they're both using the same context. What I do is: When I'm on the call, I hit More, Transfer, 700, Dial. I then get a fast busy. If I dial 701, for example, I do get the no call parked here, so I know it's somewhat working. If I try a blind transfer, it just says it failed. Do I need to add an explicit extension with a ParkAndAnnounce for these? If so, anyone have a sample of what they use? Thanks for any direction, -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Whitenoise on TDM400p
At 03:31 PM 5/31/2003 +1000, you wrote: I've seen exactly two people with this problem and it seems to be related to the system that it's running in, although I don't know if it's power supply, motherboard, or peripheral related, but somehow noise on the system power supply creeps into the system. On the systems that had trouble, we found the noise was greatly improved by doing: dd if=/dev/zero of=/dev/null which basically spins the processor and keeps it from idling. I'd like to hear from you off-list about this experience. Could there be problems with noise on the line if you use a single CAT5 cable to carry all 4 phone lines? That is what I have done (just attach 4 rj-11 on each end) and while 2 of the lines seem fine, one seems to have a fairly loud 'buzzing' sound... Is that 'white noise' or is buzzing something different? Would/Should I use 4 separate cables, or do I just need to redo some of the wiring to solve the problem? Naah, you can do a lot more than that in a single bundle, unless there's something wrong with your wire .. a crimp somewhere, some scraped up insulation, etc. Just make sure you have your pairs right .. now *that* could cause some interesting problems. :) -d Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Australia users?
I've got an end-user in Australia that needs some gear .. I can ship it over there, but it'd be a bit of a pain to find adapters localized to AU, even though I know stuff is all 110/220 these days. Anyone know a good cheap place to buy a Cisco ATA186 and maybe a Netgear/etc dsl router/switch in Queensland (or close to)? Any advice would be helpful -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI Programming of the Aastra Powertouch 480
How do I do a manual flash hook? And sorry, I didn't elaborate enough - I'm using the Nortel Vista 390. I don't see a flash button anywhere, only the link. And the link doesn't work when I load it with *'s adsiprog. Once I dump it, it's fine again . . At 10:33 PM 3/27/2003 -0600, you wrote: That is an annoying, arguably misfeature, of the Aastra. The idea is that the use of the programmed buttons should eliminate the need for the Link button since manual flash hooks can get your phone out of sync. Don't worry you can use manual flash hooks in the mean time. Mark On Thu, 27 Mar 2003, denon wrote: Does your link button work, after you program it with adsiprog? It broke on my 350 .. had to clear the asterisk load out again to use it. Link/flash is sorta important to an asterisk phone... At 08:21 PM 3/27/2003 -0500, you wrote: Well, I almost cried... Here's why: I followed the unlock procedure on my unlocked 390, and it worked as outlined, so I did the big test; I tried to unlock my 350 that I've been trying to unlock for months. There isn't a Mute button, so I hit the Flash button instead. Voila, it said Memory erase. I tried programming it with ADSIProg, and WOOHOO! It worked! My previously impermeable Vista 350 is now Asterisk-ADSI capable! THANK YOU THANK YOU THANK YOU! There are several others on this list that will be just as happy to hear this! Now, that doesn't solve your problem, but I don't have a 480 to walk you through it. Tim Clark has already replied to your message, and he has a 480. I hope he can help you out! -wade The unlock procedure is hilarious. Hit options. Choose Time/Date Set the time to Jan 1 12:00am Hit done Hit done Hit options Hit Mute A display giving the CPE ID and other stuff will appear QUICKLY press the # key It should reset. All programs will be gone and the directory/caller list/called list/softkeys will all be cleared. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and NAT
I'm having some problems getting an ATA186 behind NAT working. When I had it on the same subnet as the Asterisk server, it worked fine. Now Ive taken the ATA on the road with me, and it's behind a Dlink router+firewall, doing NAT. I pick it up, hear a dialtone .. the firewall on the asterisk side presently has everything open to this subnet, so I know that's not really an issue. (what needs to be open, though?) Here's what the * server is giving me in SIP debug, though: 9 headers, 0 lines Interface is eth0 IP Address is 1.0.0.1 Using latest request as basis request Sending to 192.168.0.150 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=6b5fab60 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Suggestions? (IPs have been changed to protect the innocent) Thanks for any help you can give -- I was sort of relying on using this while I'm on the road. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and NAT - more
Oh, and yes, the * is current as of a few days ago .. so it should have that new SIP code mark was working on a while back. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and NAT - more
Thanks -- I didn't realize that needed to be set. It works now, but there's a horrible echo on the sip client side. (I dont know about the other side, as I havent called any humans yet :) I don't, however, hear an echo when I call voicemail or such .. so I'm assuming it's something with the bridging? I didn't know of any echo cans that need to be enabled for sip - are there? The PSTN line its connecting out on has echocan and whenbridged enabled. Here's an example of one of the pstns, they're all built the same, using an Adtran 750 channel bank with current firmware (actually, the last release, which was considered the most stable by most): context = pstn1 signalling = fxs_ks amaflags = documentation echocancel=yes echocancelwhenbridged=yes adsi=yes channel = 17 Ideas? Thanks At 09:53 PM 3/21/2003 -0600, you wrote: have you tried nat=1 in your friend declaration? I notice in your dump it says non-NAT Mark On Fri, 21 Mar 2003, denon wrote: Oh, and yes, the * is current as of a few days ago .. so it should have that new SIP code mark was working on a while back. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ITSP requested features?
At 01:58 PM 3/16/2003 -0500, you wrote: What else appeals to you? Voicemail delivery via SMTP? Yes, but most pbx users will most likely be doing this on their own, so I don't know how important it is. Automatic call forwarding to a PSTN number when your SIP client isn't registered? Sure, but again, how many PBXs go down for long periods of time? Generally that type of routing would be done with an extensions.conf, even though it's inefficient to take the stream in, and route it back again. . Call record billing statements with custom accounting codes? Nah, in the pbx. . IAX protocol support? Yep. IAX2 preferably. 911 emergency call routing? Unlimited international calls? Unlimited International would be nice, though I don't know how economical it would be. How about unlimited to certain countries .. Keep in mind, you're not the only kid on the block .. lots of people already working to implement IAX support, so you'll need something unique - whether it be price or unlimited international, etc. denon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proposed IAX2 Name
I still like my original idea, SVP - Streamlined Voice Protocol. I think the trick to naming IAX2, is to encourage the thought of streamlined and efficiency. If people want a feature-rich and bloated protocol, they'll run H.323, SIP, etc .. IAX is all about performance and resource utilization. denon At 09:19 AM 3/13/2003 -0600, you wrote: What do you all think of renaming IAX2 as: Telephony Authentication, Signalling, and Transport Exchange (TASTE) TASTE is easy to remember and has a sort of ironic relation to SIP. Is it took hoaky? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax support?
Nope, that would require simulating the a fax dsp, or such. You can, however, route a call to a certain zap port when a CNG tone is detected. At 01:36 PM 3/3/2003 -0500, you wrote: Thanks Martin, I'll set it up tonight. Can asterisk receive the incomming fax and store it to local disk with email notification? Gene -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Mon 3/3/2003 12:41 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Fax support? let's say you have one T1 span configured like this in zapata.conf context=incoming group = 1 channel = 1-23 then in extensions.conf [incoming] exten = fax,1,Dial,Zap/25 #FXS port that fax is plugged to exten = _X,1,... (the rest) when asterisk detects fax tones on incoming call it's going to look for fax extension in the channel context. If it finds it then the fax call is going to be routed according to the fax extension rule. regards Martin On Mon, 3 Mar 2003, Gene Kochanowsky wrote: Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
