RE: [Asterisk-Users] SIP Authentication
Title: Message Hi, Does anyone know the solution to this issue? Regards,Stojan Sljivic -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDSSent: Friday, June 10, 2005 13:21To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] SIP Authentication Hi, I use SIP softphone that is not registered at Asterisk. When I dial some extension defined in the dial plan ([EMAIL PROTECTED])with my SIP softphone, Asterisk will not ask me for username/password (will not return response 407) as I expected. The response 407 - Authentication required will be returned if username defined in the softphone's setting matches one of the SIP peers defined in sip.conf. This means that anyone can dial extension at my Asterisk and that is not good, since that person could then dial over my ZAP line. How can I configure Asterisk to allow only peers defined in sip.conf to register and dial? Regards, Stojan Sljivic ___ 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] Macro support in realtime
Is there any way to accomplish the following? (searched and searched and can not find any examples) In extensions.conf (text file) define a macro that accepts a handful of arguments From realtime mysql (extensions) - call the macro with arguments (where the macro is static in the text file) If not, what about putting the macro in mysql? Just trying to find a way to reduce the number of db records per extension to 1 from 6+ by calling a macro with 6+ arguments from a single record. Possible? ___ 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] Phantom incoming calls on x100p
Hi! I have a problem with one box running asterisk, one pots line and an X100P. Almost every night the phones give 2-3 rings and then stop. There are no actual incoming calls, I verified by putting a device that lists the incoming telephone numbers parallell to the X100p and it doesn't list any calls. This is the output on the console for a real incoming call: == Spawn extension (inbound-analog, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jun 12 17:20:26 NOTICE[8653]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... Jun 12 17:20:27 NOTICE[8653]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Dial(Zap/1-1, SIP/201SIP/202|70|tm) in new stack -- Called 201 -- Called 202 -- SIP/201-1947 is ringing -- SIP/202-e2e7 is ringing -- SIP/202-e2e7 answered Zap/1-1 This is one of the phantom calls: == Spawn extension (inbound-analog, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jun 13 02:49:43 WARNING[8653]: chan_zap.c:5445 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Dial(Zap/1-1, SIP/201SIP/202|70|tm) in new stack -- Called 201 -- Called 202 -- SIP/201-04fd is ringing -- SIP/202-5f9d is ringing == Spawn extension (inbound-analog, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there any way to kill this? Thanks! ___ 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] Need Help with pickup *8
Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib ___ 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] GXP2000 and hint LED's
On Fri, 10 Jun 2005, Peter Svensson wrote: On Fri, 10 Jun 2005, James Bean wrote: Peter seems to be on the ball more then me about these phones as grandstream gave me the standard replies, Peter do you know for sure if grandstream have a timetable for the function led's cause I need to rollout about 50 phones and need 6-7 led's for display, which means a snom220+expansion, and gxp2000 seems perfect if it worked. I am certain that at least some documentation mentioned that the buttons will provide subscribe/notify in the future. I will ask our distributor to see what the official Grandstream position is. I received word from Grandstream today. The subscribe functionality is expected to make the next release. It is expected to ship in 1-2 months. No promises, but it is apparently high on their list of requested features. 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] Asterisk code
Hi- I want to learn asterisk code and its archetecture where can i get help. --- Kib Eki [EMAIL PROTECTED] wrote: Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib ___ 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 Best Regards Ibrar Ahmed Project Manager. Comcept (Pvt) Ltd. Islamabad Pakistan www.com-cept.com [EMAIL PROTECTED] [EMAIL PROTECTED] Ph # (Off) +92-51-111784784 Ph # (Res) +92-51-2271283 Ph # (Mob) +92-3009543001 Fax # 92-51-111784785 www.com-cept.com Pick battles that are big enough to matter, small enough to win __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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 installation error after CVS update
Good morning!
Asterisk 1.0.7 runs fine on my machine with Suse 9.3 when using a
downloaded tarball. But as I wanted to have a look at Realtime I
decided to download everything again via CVS with
# cvs checkout zaptel libpri asterisk
and install it.
Unfortunately though the Asterisk installation itself stops at some
point with the following errors:
=== snips ===
chan_sip.c:36: internal compiler error: output_operand: invalid
expression as operand
Please submit a full bug report,
with preprocessed source if appropriate.
See URL:http://www.suse.de/feedback for instructions.
{standard input}: Assembler messages:
{standard input}:148310: Warning: partial line at end of file ignored
Preprocessed source stored into /tmp/ccbflXC6.out file, please attach
this to your bugreport.
make[1]: *** [chan_sip.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
=== snip ===
Checking the file /usr/src/asterisk/channels/chan_sip.c, line 36 says:
ASTERISK_FILE_VERSION(__FILE__, $Revision: 1.759 $)
Is this now actually an Asterisk error, or a Suse error?
Regards,
Gunde
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[Asterisk-Users] Problem with DTMF Relay and Oh323
When the inbound leg of the all is SIP and the outbound leg is Oh323 (Voip-to-Voip only here), the DTMF relay (either RFC2833 or SIP Info), fails to go through, while it works perfectly when both legs of the call are SIP. Is this a shortcoming of the Asterisk core or the Oh323 channel? Is this solvable at all with some configuration change or a simple rewriting of the Oh323 channel driver? Second question: how can I force the Oh323 to propose only one codec to the outbound H323 endpoint, and do not negotiate? The choice of codec is a business decision: if the gateway is located in my own subnet I don't need compression, but if not I need to use only G29, etc. ___ 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] Modprobe wctdm hang at command prompt
Title: Message Hi Chee, We are experiencing the same issue. Did you find a solution for this and can you please share it with us? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with DTMF Relay and Oh323
I have the same on calls originating from a sip phone and going into a ZAP channel.Andre- Oorspronkelijk Bericht -Onderwerp: ___ 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] 407 Proxy Authentication Required
Title: 407 Proxy Authentication Required I am getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network using sjphone snom. How to overcome this..!! Pls advice..! Shahan This e-mail may contain confidential and/or privileged information. If you are not the intended recipient or have received this e-mail in error, please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ 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 code
Ibrar Ahmed wrote: Hi- I want to learn asterisk code and its archetecture where can i get help. :) You could try the psychiatrist. Or maybe just a local support group. :) Jokes aside, some good resources are: www.voip-info.org www.asteriskdocs.org my news (www.sineapps.com/news.php) IRC (irc://irc.freenode.net/asterisk) Or make progdocs from asterisk or simple 'use the source luke' And post a question if you have one. -- 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] about timeouts
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
I've this infrastructure:
|voip services| -- |*| -- |cme| -- |isdn|
the voip services are logged on my *, then forwarded to number 601 on
cme. The isdn calls too are forwarded to 601. On cme I've a timeout X
for call-forward noan (no answer) to a specific number on * (5901) that
is my x-lite software client. If 5901 is unreacheable the call is
forwarded to voicemail (always on *) -- u601
Tests:
- - for isdn calls all works great, even if X=30sec
- - for voip services: if X=20sec, and 5901 is unreacheable, all works
fine; if X=30sec, I've a tear down without voicemail; if X=20sec, but
5901 is logged in, I've a tear down after 7-8 sec on x-lite without
wait 20sec+voicemail as configuration.
I think there's a problem with * timeouts. What could I do?
my extensions.conf:
exten = _59XX,1,Dial(SIP/${EXTEN},20,tTr)
exten = _59XX,2,Hangup
exten = 5901,1,Dial(SIP/5901,20,tTr)
exten = 5901,2,Voicemail(u601)
exten = 5901,3,Hangup
Any suggestion will be appreciated
Regards
Andrea
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[Asterisk-Users] MySQL: max realistic size of extensions table.
Hi, I'm using *CVS Head version and read the dialplan from MySQL. I'm making A-Z termination to over 4000 different country and city codes.I have 3 different dialing rules depending on the price level of the dialed number. Should my extensions table contain 4000 lines? Is this realistic? Or is there any other (more clever) way doing this? Regards, Cenk. ___ 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] SNOM, Asterisk and Attended transfer (bug?)
Hi, I am using a number of snom190 phones, and an asterisk gateway server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP (ie. ISDN) I hope that some of this makes sense... When I look at the SIP trace for the sequence of A calls B and is transferred to C, I see: A makes call to B: A calls B B picks up A and B are bridged (re-INVITEd) and talk directly. B then puts A on hold: (A and B are both INVITE to talk via Asterisk) B makes a call to C, I see: B calls C C picks up B and C are bridged (re-INVITEd) and talk directly. B presses transfer: (Same as putting B and C on hold, B and C are re-INVITEd to talk via Asterisk) B selects which line to transfer to C B REFERs A to C by asking Asterisk. Asterisk accepts this. B is notified that A is disconnected B gets BYE for call to A B gets BYE for call to C C gets INVITE to talk to B via Asterisk Why? Why not to 'A' B requests that call to A is closed down. The upshot of all this is that B is correctly out of the loop, and that Both A and C think they have opened communications with a new phone, both via Asterisk. Unfortunately there is no Audio. If one of the parties hangs up, the connection is correctly closed. I am curious why Asterisk would put a From: of B in the final INVITE to bridge the calls. Perhaps this is just how SIP spoofs the communication so that C does not need to know about the 2 callers? Is there some way I can track down where my audio is going? As mentioned, this problem only seems to occur if A,B,C are all SIP phones, but not if A is an ISDN call. Thanks, Steve ___ 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] Guidance , for which card to buy
Hi I am planning to try to use Asterisk for testing purpose , for PBX systems , I have one PC with RHEL 4 I want to buy digium cards for this purpose , but I am not sure about which card to buy , this field is totally new to me , requesting guidance for selecting the card I am in middle-east , UAE , Dubai . where the OpenSource / Linux usage is very less . I would also like to know from the members that whether any one in this area is using Asterisk Thanks Joseph John ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.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] PRI trouble
On Sun, Jun 12, 2005 at 06:10:53AM -0400, Michael Di Martino wrote: However i still get the same error. Please help we cannot connect call form my norstar to asterisk w/ it dropping in 10 seconds. Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of span 2 Jun 12 10:51:51 WARNING[213005]: No D-channels available! Using Primary on channel anyway 48!Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 35 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 36 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 37 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 38 Hmmm. Links works and suddenly stops. No changes to either side both of which you control. To test the cable idea someone had, pull the cable and see if messages are the same or different. If the same, then it's likely the cable that's the problem. If different, the it's likely that the cable is not the problem. My vote is for clocking. I've had non-Asterisk T1/E1 circuits act fine for long periods and then flake out. Correcting the clocking relationship fixed the flakyness. One way to investigate clocking is to use the pri commands to watch the D channel activity. If you see lots of Q.921 messages (SABME, RR, etc.) then it might be the result of fouled up messaging from bad CRC and other broken protocol events stemming from bad clocking. I've collected some articles and discussions on T1/E1 clocking here: http://www.voip-info.org/tiki-index.php?page=Asterisk+PRI -- Mike ___ 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] Cepstral partnership with Digium
I just read about the partnership but was wondering what is actually going to happen? Is asterisk going to be bundled with cepstral voices for free :)? Or whats the deal? ___ 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] Oh323 and Caller ID missing
I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I set the caller id correctly in my perl AGI script $AGI-set_callerid($ani); , the gateway does not see any caller id coming from my Asterisk box. I use the very latest version of Oh323 as published in the Inaccess web site, and Asterisk HEAD from two days ago. The caller ID important for this client, because he will further authenticate the call based on the ANI. I am only doing a codec conversion. Any help is appreciated from Jeremy McNamara. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: POLYCOM IP 500 Setup
Hi Matt -Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP I created SIP extension 205 to be used with my Polycom phone. I setup username = 205, secret = 123, context = from-internal. I setup my phone to have a static IP address, then pointed my web browser at it, to setup my phone. I setup Sip Conf with: Address = "IP of * server", Server1 = "IP of * Server" Under Registration, I setup: Identification: Address = "IP of * Server" , Auth User ID = 205, Auth Password = 123, Server1: Address = "IP of * server"For your phone-specific file, address isn't the asterisk address, it is the sip address of the phone - you can just use "205".- Noah___ 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] Anyone noticed Voipjet voice quality problems?
I never noticed any problems.. so I can't comment :) hehe On 6/11/05, Pedro [EMAIL PROTECTED] wrote: Finally got a response from voipjet support and they say they have switched to a new provider for US termination. I have yet to test this out as I have not had a chance to build them back into our routes but will report my findings once I do. Anyone else notice any improvements? On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ 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] Re: POLYCOM IP 500 Setup
[EMAIL PROTECTED] will not be able to configure polycom500 phones. You need to add this entry in sip.conf manually with one additional line as under: progressinband=no Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah MillerSent: Monday, June 13, 2005 9:44 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: POLYCOM IP 500 Setup Hi Matt - Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP I created SIP extension 205 to be used with my Polycom phone. I setup username = 205, secret = 123, context = from-internal. I setup my phone to have a static IP address, then pointed my web browser at it, to setup my phone. I setup Sip Conf with: Address = "IP of * server", Server1 = "IP of * Server" Under Registration, I setup: Identification: Address = "IP of * Server" , Auth User ID = 205, Auth Password = 123, Server1: Address = "IP of * server" For your phone-specific file, address isn't the asterisk address, it is the sip address of the phone - you can just use "205". - Noah NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] nativ bridging problem with ilbc!!
hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---asterisk -- ( asterisk2 -- sip grandstream (alaw) ) \-native bridge--/ 1. if i use on iaxcomm as default speex, nativ bridging between iaxcomm and my sip phone is working 2. if i use ilbc on iaxcomm and try from an pat network (no native bridging!!) the code translation on my * (asterisk2) server works fine 3. if i use ilbc on another (not mine, mine does not work with ilbc) grandstream adapert everything works (no code translation) 4. if i use like the same like in 1. but ilbc instead of speex, it sometimes works, most of the time not, only one side hears the other. does sombody has an idea what teh problem could be? why this only happens with ilbc codec? i have ilbc enabled and as first codec in all iax.conf's and also in the sip.conf thanks, alex ___ 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] presence and video conference
Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar. ___ 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] SNOM, Asterisk and Attended transfer (bug?)
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, June 13, 2005 6:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?) Hi, I am using a number of snom190 phones, and an asterisk gateway server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP (ie. ISDN) I hope that some of this makes sense... When I look at the SIP trace for the sequence of A calls B and is transferred to C, I see: A makes call to B: A calls B B picks up A and B are bridged (re-INVITEd) and talk directly. B then puts A on hold: (A and B are both INVITE to talk via Asterisk) B makes a call to C, I see: B calls C C picks up B and C are bridged (re-INVITEd) and talk directly. B presses transfer: (Same as putting B and C on hold, B and C are re-INVITEd to talk via Asterisk) B selects which line to transfer to C B REFERs A to C by asking Asterisk. Asterisk accepts this. B is notified that A is disconnected B gets BYE for call to A B gets BYE for call to C C gets INVITE to talk to B via Asterisk Why? Why not to 'A' B requests that call to A is closed down. The upshot of all this is that B is correctly out of the loop, and that Both A and C think they have opened communications with a new phone, both via Asterisk. Unfortunately there is no Audio. If one of the parties hangs up, the connection is correctly closed. I am curious why Asterisk would put a From: of B in the final INVITE to bridge the calls. Perhaps this is just how SIP spoofs the communication so that C does not need to know about the 2 callers? Is there some way I can track down where my audio is going? As mentioned, this problem only seems to occur if A,B,C are all SIP phones, but not if A is an ISDN call. Thanks, Steve ___ Upgrade your snom firmware to the latest and make sure break key = off in the setup. Use the transfer feature in asterisk for attended transfers. ___ 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] SIP Authentication
Title: Message You may need to look to see if you are using peer or friend in the sip config for this phone. We needed to change ours to friend to make it work for us with still using secret. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Sunday, June 12, 2005 11:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Authentication Hi, Does anyone know the solution to this issue? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Friday, June 10, 2005 13:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP Authentication Hi, I use SIP softphone that is not registered at Asterisk. When I dial some extension defined in the dial plan ([EMAIL PROTECTED])with my SIP softphone, Asterisk will not ask me for username/password (will not return response 407) as I expected. The response 407 - Authentication required will be returned if username defined in the softphone's setting matches one of the SIP peers defined in sip.conf. This means that anyone can dial extension at my Asterisk and that is not good, since that person could then dial over my ZAP line. How can I configure Asterisk to allow only peers defined in sip.conf to register and dial? Regards, Stojan Sljivic ___ 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] SIP Authentication
Title: Message Greetings, You have stumbled on to one of the most troublesome flag for newbies; autocreatepeer. http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer in your sip.conf file add a line in the [general] section autocreatepeer=no Now people can only use your Asterisk SIP connection if you create a peer entry for them in your sip.conf file. Your sip.conf file should be located in /etc/asterisk directory. cd /etc/asterisk vi sip.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Monday, June 13, 2005 2:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Authentication Hi, Does anyone know the solution to this issue? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Friday, June 10, 2005 13:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP Authentication Hi, I use SIP softphone that is not registered at Asterisk. When I dial some extension defined in the dial plan ([EMAIL PROTECTED])with my SIP softphone, Asterisk will not ask me for username/password (will not return response 407) as I expected. The response 407 - Authentication required will be returned if username defined in the softphone's setting matches one of the SIP peers defined in sip.conf. This means that anyone can dial extension at my Asterisk and that is not good, since that person could then dial over my ZAP line. How can I configure Asterisk to allow only peers defined in sip.conf to register and dial? Regards, Stojan Sljivic ___ 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] presence and video conference
Hi Juraj, I have been trying for some time to fund video conferencing support and have offered a personal bounty of several thousands of dollars in order to get it developed. So far 5 people have contacted me but apart from one point to point solution I'm still waiting. In the interim I have purchased www.smiletiger.com software for my video conferencing requirements. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Juraj Bednar Sent: Monday, 13 June 2005 10:21 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] presence and video conference Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar. ___ 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] wiki server session limit?
It seems that the wiki pages at www.voip-info.org are not responding, and this has happened before. Responds to ping but not http requests. Is there a session limit on the web site? Is it too low? Maybe another explanantion? Anyone else notice? ___ 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 to Cisco Unity
Also check out the CISCO GKTMP API, that is their gatekeeper api. There might be some cool stuff you might like to know. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Sunday, June 12, 2005 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity My fault. I understand my terminology was not accurate. Thanks for your reply. Simone Steve Hanselman wrote: With call manager V4 and above it's extremely easy, just connect a SIP trunk to *. BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX so your terminology may be confusing some people. From: [EMAIL PROTECTED] on behalf of Simone Sent: Fri 10/06/2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity I understand what you're saying, but I am not the one who makes the decisions. That decision is made already, so since I am actually getting your point and I agree with that, the only thing I can try to do right now, is try to avoid having Cisco Unity in the other 3 offices. I would love to implement Asterisk in these ones, but if it cannot be connected to Cisco this won't be an option at all, they won't consider it. So, back to the question, is it possible to connect Asterisk to Cisco and have all the functionality expected, and is it hard? Thanks, have a nice day Simone William Boehlke wrote: By the time you install the Asterisk server you have more features than Cisco delivers with Unity, for half the cost and without those annoying viruses. So instead of thinking about connecting Asterisk, consider disconnecting Unity. They make excellent landfill. Regards, William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Thursday, June 09, 2005 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ 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] Asterisk code
Also subscribe to the asterisk-dev mail list. Watch it for a couple of days before you ask a question or they will eat your lunch. Pick a single thing you want to change in the PBX, and then learn how to code for that. Something really simple like adding a parameter to a conf file is a good place to start. If you can do that then you can move to harder stuff. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, June 13, 2005 6:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk code Ibrar Ahmed wrote: Hi- I want to learn asterisk code and its archetecture where can i get help. :) You could try the psychiatrist. Or maybe just a local support group. :) Jokes aside, some good resources are: www.voip-info.org www.asteriskdocs.org my news (www.sineapps.com/news.php) IRC (irc://irc.freenode.net/asterisk) Or make progdocs from asterisk or simple 'use the source luke' And post a question if you have one. -- 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
[Asterisk-Users] Components and suggestions for an asterisk server with 9 to 17 POTS.
Hello, What would be the simplest and the cheapest solution to get an Asterisk server working with 9 to 17 POTS? Because for 1-8 POTS we are using 1 or 2 Digium TDM cards and past 17 POTS in our area it is economic to use a PRI. We are looking for a hardware solution on our side instead of using did provider Thanks in advance Ken ___ 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] wiki server session limit?
Are they running on a windows server? :)=) Maybe it has the Monday Morning Blues. (I can't get it to talk either.) Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 11:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wiki server session limit? It seems that the wiki pages at www.voip-info.org are not responding, and this has happened before. Responds to ping but not http requests. Is there a session limit on the web site? Is it too low? Maybe another explanantion? Anyone else notice? ___ 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] Re: Asterisk forking, Was: Digium Website Update: Asterisk Business Edition
Andrew Kohlsmith wrote: On Saturday 11 June 2005 19:51, Lee Howard wrote: I don't think that lack of mindshare completely defines the reasons behind Asterisk fork failures. It places all of the blame on the forkers. I think the truth, though, is that they not only fail due to lack of mindshare but also due to competition from Digium's own Asterisk community. Forks are not succeeding, yes, but Digium has a hand in that... of course they do. I'm not saying you're wrong, but I'm curious: how does Digium have a hand in a fork failing? That's what I tried to explain in my last post, in particular after this first statement. Forks enter a hostile competition rather than a healthy competition. I've heard more talk about Asterisk forks than I've ever heard about forks of any other other open-source project. I think that this says something about how difficult-to-swallow Digium's dual-license decree is for a lot of prospective contributors/developers. I disagree; if it were that hard to swallow the project would either be 90% digium-written (it's not) or it would be a total flop (again it's not). If you (or someone else reading this post) is in a position to give statistics on what percentage of the code is Digium-written (or Digium-rewritten - in the case where a disclaimer is not obtained for some unpatented work and Digium rewrites the work independently) then I would be thrilled to see it. We see this happen all of the time with the Linux kernel. It happens with HylaFAX. It happened with X. I'm sure it happens a lot with many other open-source software projects. It happens easily and usually is a healthy process because the playing field is even. Agreed. But where are the successful Asterisk forks? I don't know of any successful Asterisk forks (unless http://www.asteriskwin32.com is considered successful - although I'll admit that I'm not really in-the-know). But this was my point: that the way things were set up by Digium makes a successful fork difficult. Digium always has an upper-hand, and things were set up intentionally this way. Again, I don't take particular issue with this. I'm just trying to explain why forking Asterisk would not be a particularly easy task. Of course, this healthy forking cannot be done with Asterisk because Digium will not accept any non-disclaimed code into their repository. ... What you'd described about distribution-maintained patches has nothing to do with this. Digium could take a distribution-maintained patch and rewrite it into Asterisk proper under the dual license (as could any other contributor) and you'd still gain the benefit of the patch. I'm not sure I see where you're going here. If you (or someone else reading this) has the necessary information to provide statistics on how what percentage of the code comes from rewrites of non-disclaimed code, then I would be particularly interested in hearing it. I suspect, though, that it is a rather small - perhaps insignificant - amount. But, yes, providing that there is not a patent involved - yes, the work could be rewritten and integrated. But this was my point: that given the right environment forks can benefit from each other. The one thing that an Asterisk fork can never do, though, is relicense itself. Only Diguim can do that. If Digium had wanted an equal footing in this regard then Asterisk would be LGPL or BSD or something a bit more liberal. So if I'm a manufacturer of PBXes and have some proprietary IP that I do not wish to be GPLed, then if I want to use Asterisk somehow, then I can really only work with Digium for licensing. All of the other forks will be license-prohibitive. I have to admit that I know quite a few people with their own modules and such to replace what they feel is bad code and just won't contribute it back to Asterisk due to the friction they've received about the patch. I, on the other hand, tend to bitch loud and continuously enough and wear them down to the point of accepting it. :-) So we're not in disagreement, it would seem. Getting code contributions into Digium's Asterisk codebase is not something that many average people are going to want to undergo. From what I've seen, friction is a bit light of a term for it. It seems much more hostile than that. And, that's often even before the disclaimer hurdle is reached. Lee. ___ 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] T1 multiplexer (or ?) for failover in large installation
Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ 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] snom 190: dial tone without registration?
Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the make/accept calls without registration feature. Or more specifically, produce a dial tone even if I'm not registered. I would like to set our sip.conf entries to host=ipaddr instead of host=dynamic to enforce the IP addresses used by each sip phone. If we do this, Asterisk does not accept SIP registration from the phone on that IP. This is fine with the SPA-841, but the snom 190 displays a NR status (Not Registered), and refuses to play a dial tone when you take the phone off-hook. The phone will place calls correctly, but it insists on thinking it's in an error condition when it can't register. Note that Asterisk's defaultip option in sip.conf isn't adequate, because it doesn't deny access if the phone attempts to register from a different IP. Does anyone know how to do one of the following: - tell the snom 190 not to register, but to use the outbound proxy for outgoing calls, and produce a dial tone anyway. - tell Asterisk to enforce an IP address restriction for a specific SIP channel, but still accept an incoming registration from that IP. Vitals: - Asterisk 1.0.7 - snom 190 firmware 3.60i Thanks, Alan Ferrency pair Networks, Inc. [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
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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] T1 multiplexer (or ?) for failover in largeinstallation
Just use a cisco with 5 T1 ports and have everything over IP use ultra monkey to load balance your asterisk boxes. I have found this works very well. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Monday, 13 June, 2005 11:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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] Wildly inaccurate CDR records
An accurately CDR it depends, i think, the way you want to make the bills. So its a combination of using NoCDR, ForkCDR commands and billsec, disposition and other database fields manipulation. For example, IAX calls may be recorded with very few time of duration if you dont use the parameter notransfer=yes in iax.conf could you give us a detailed example of what do you need? so we can figure out a solution for your problem? On 6/11/05, snacktime [EMAIL PROTECTED] wrote: On 6/11/05, Obelix [EMAIL PROTECTED] wrote: Quoting Obelix [EMAIL PROTECTED]: Is this question too difficult, or is it simply one that only a few users have experienced? I believe the forkcdr command is what you want, although I've never used it. Chris ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] snom 190: dial tone without registration?
On Monday 13 June 2005 16:42, alan wrote: Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. How about tackling this with iptables and matching specific IP addresses on specific MAC addresses? Cheers, Gavin. ___ 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] SIP-H.323 dial tone and busy tone problem.
Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made? best regards On 6/11/05, Carlos Alberto Lara de Hoyos [EMAIL PROTECTED] wrote: Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] Interfacing to an IAD
I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an IAD which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a Lucent Digital Vina Integrator IAD with a digital TC card. I've searched the web to find any sort of info on how I can connect this IAD to my Asterisk box without success. What I find in general is that this kind of IAD can either provide analog voice output (POTS) or digital T-1 output. I presume the latter is what they're providing. Is a digital T-1 from an IAD the same kind of interface as a PRI T-1? Would something like a Digium TE-110P handle this interface? Does anyone out there have experience with a Vina IAD with digital voice circuit output? Thanks for any assistance. -Corwin Nichols ___ 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] T1 multiplexer (or ?) for failover in largeinstallation
It sounds like your looking for a t1 protection switch with will do what you want It can switch t1's for failover or loss of signal on a t1. These are usualy rather expensive and Might work 100% in your example because they will only switch over if th actuall t1 goes down. So If your server dies/locks up and doesnt tell the t1 to go into alarm it will still think it is up and not switch to the other. - Original Message - From: Mike [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 13, 2005 10:35 AM Subject: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ 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] ProSLIC 3210 version 2 is too old.
Twas an issue with the card. I tried a different TDM20B and it worked perfectly. RMA time. Whenever I load wcfxs I get ProSLIC 3210 version 2 is too old. ___ 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 code
Race Vanderdecken wrote: Also subscribe to the asterisk-dev mail list. Watch it for a couple of days before you ask a question or they will eat your lunch. Or even more likely, eat you for lunch! :D -- 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] DNIS and DID seeking confirmation
Hello all, After much googling I have come to the conclusion that in asterisk land DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are used rather interchangeably. If this is an incorrect assumption Please correct me. Based on this assumption if I have everthing set up to land in the [incoming] context and an 800# such as 1-800-123-4567 with 4 digit DNIS I can have an entry in my incoming context exten = _4567, 1, do something this is where the call to my 800 number will land regardless of which trunk the call comes in on. Like wise if I have a DID number 456-7891 with an exten= _7891,1,do something else this will also work. Is this correct or am I way off base? Also what is Asterisk looking for as far as a delimiter or is that in a config file? Something like Seize (Wink) DNIS (Wink) ANI (Wink) Answer or Seize (*) DNIS (*) ANI (*) Answer John M ___ 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] Sip Trunking
I'm trying to setup SIP trunking between 2 asterisk servers. Eventually
there may be up to 5 servers linked together depending on the growth
needed. I have IAX2 trunking working, but I want both.
For simplicity, I have named the two servers, alpha and beta. Extension
7100 is a Polycom IP600 on alpha and extension 7300 is using kphone on
beta. Both ae using SIP. Below are the relvenant parts of
extensions.conf and sip.conf.
;# server alpha
; extensions.conf
[staff]
extension = 7100,Dial(SIP/7100)
extension = 7300,Goto(siptrunk,7300,1)
[siptrunk]
include = siptrunk-beta
[siptrunk-beta]
exten = _73XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=beta's IP address
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=beta's IP address
;# server beta
; extensions.conf
[staff]
extension = 7300,Dial(SIP/7300)
extension = 7100,Goto(siptrunk,7100,1)
[siptrunk]
include = siptrunk-alpha
[siptrunk-alpha]
exten = _71XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=alpha's IP address
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=alpha's IP address
When dialing 7300 from alpha, I get the following:
-- Executing Dial(SIP/7000-e924, SIP/siptrunk-peer/7300) in new
stack
-- Called siptrunk-peer/7300
Jun 13 11:10:47 WARNING[18099]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 45a607bc5f66cfb363f2cc565b85fa29@alpha's IP
address for seqno 102 (Critical Request)
== No one is available to answer at this time
Any ideas?
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RE: [Asterisk-Users] 407 Proxy Authentication Required
Title: 407 Proxy Authentication Required We also have the same problem over long latency networks ATA also gives Call Rejected: 407. We have tried a lot of different phones and soft phones and the only one working is Xten. In any case this is apparently only problem with newer versions of * - you can use very old version you can avoided the problem. We were not yet able to find final solution for this problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahan Kalutanthri Sent: Monday, June 13, 2005 3:20 AM To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] 407 Proxy Authentication Required I am getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network using sjphone snom. How to overcome this..!! Pls advice..! Shahan This e-mail may contain confidential and/or privileged information. If you are not the intended recipient or have received this e-mail in error, please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ 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] Interfacing to an IAD
I have had experience with both the Vina and with XO. If you ask for it, you should be able to get an Adtran 600 series on the circuit. I never had any success with the Vina and it really is not a piece of equipment I would bet the farm on. They may have improved but I would still just as fo ran AdTran since it definitely has a T1 interface that works. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corwin Nichols Sent: Monday, June 13, 2005 8:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Interfacing to an IAD I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an IAD which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a Lucent Digital Vina Integrator IAD with a digital TC card. I've searched the web to find any sort of info on how I can connect this IAD to my Asterisk box without success. What I find in general is that this kind of IAD can either provide analog voice output (POTS) or digital T-1 output. I presume the latter is what they're providing. Is a digital T-1 from an IAD the same kind of interface as a PRI T-1? Would something like a Digium TE-110P handle this interface? Does anyone out there have experience with a Vina IAD with digital voice circuit output? Thanks for any assistance. -Corwin Nichols ___ 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] Re: Digium Website Update: Asterisk Business Edition
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes: Protecting freedoms by putting limits on (thus restricting freedoms). Interesting concept. I need to repeat here. The gpl's purpose is to protect the freedoms that comes with free software. So, you have only the freedoms that comes with free software as defined by the FSF. You are not allowed to do what you like. You are constrained to the freedoms that follows the software and I think that is a very interesting concept. copyright and license to use are different. I never claimed otherwise. You can technically put software out there with no copyright but under the gpl license Then there would be no one to enforce the license, which would be bad. it only restricts *their* code (ie modifications). Yes, but we also want all modifications to be free the cost of restricting freedoms on others and what they can do with their code Yes, by using the GPL you restrict everyone to the four freedoms defined in the free software definition. This is exactly what we want. The BSD license for example lets your code remain free while giving people the freedom to create code of their own, as a modification of yours, and use their code how they want. This is exactly the reason I choose GPL, because it doesn't allow people to do whatever they want. They only have the freedoms that comes with free software, which is exactly what we want. This ensures that the code stays free and any modification too it is also free. This is what we want and you obviously want something else. When we, the saints of the church of emacs, speaks about free software we are referring to the freedoms that comes with free software (nothing more, nothing less). Free software has a definite definition for us, which is that of the fsf. If people want your version they can always get that from you, and so it is intact as 'free'. Yes, but we also want the modifications to the software to be free. We basically want what's defined in the GPL. It does not give full unrestricted modification clauses. You can modify it as much as you want as long as the modifications also are free, just as the original code. proposed GPL 3.0 I rather not discuss GPL 3.0 before a draft. Your version which you released 'free' would still be there. In its unmodified glory. By using the GPL, we also ensure that any modification to it, be free. This is desired. The GPL does not ensure freedom to all It ensures the freedoms that are defined in the free software definition. it works like a parasite and infects future code Yes, this parasitic effect is exactly what we want. All it does is force others who write code to be assimilated into the same doctrine. Yes, which is exactly what we want. If you choose to use GPL code, you have to follow the rules. I guess what I am trying to say is that GPL does little to protect the original author The copyright protects the original author by law. it removes freedoms from subsequent authors by forcing them to license in the same way. Yes, and that's what I love about free software. The software stays free. it doesnt guarantee the freedom of subsequent authors, it curtails that freedom. Once again, it only guarantee freedoms that follow free software. And you can copyright (and infact do) without the GPL. Yes, but we use the gpl to protect the freedoms that follows free software. The GPL is *not* a copyright it is a license for use. They are very different things. You can copyright something and distro it without GPLing it. Indeed. The free software continues to be as free as the author wants. Yes, the copyright holder can do whatever he feels like with the code. Once he puts a GPL on it and release it, the code is free for ever and any modifications to it is also free. By holding the copyright, he can also choose to change the license, but only on the code that he holds the copyright of. The code that was released as free, however, stays free. it does however curtail the freedoms of any subsequent authors that enhance the code. Which again, it's the desired effect. subsequent authors now have *no* choice in how they license it, they are forced to license it the same way as you, which curtails freedom. Yes, glad you understand cause this is the purpose. The freedoms that follow free software will continue to follow it and neither you nor anyone else can change that. The modifications are the *only* difference between what you release and what they release, so if they use your code as a base and make changes to suit a particular need, their code, which they did write all of, cannot be licensed how they choose This is exactly what we want. the parasitic nature of the GPL means that their modifications, *their* code, must also be GPLed You're just explaining what we want. The GPL doesnt protect freedom, it curtails freedom of future developers. The GPL protects the
RE: [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
This is a very interesting converation, but it seems like the BIZ forum might be more appropriate... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: Monday, June 13, 2005 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition Andrew Kohlsmith wrote: On Saturday 11 June 2005 19:51, Lee Howard wrote: I don't think that lack of mindshare completely defines the reasons behind Asterisk fork failures. It places all of the blame on the forkers. I think the truth, though, is that they not only fail due to lack of mindshare but also due to competition from Digium's own Asterisk community. Forks are not succeeding, yes, but Digium has a hand in that... of course they do. I'm not saying you're wrong, but I'm curious: how does Digium have a hand in a fork failing? That's what I tried to explain in my last post, in particular after this first statement. Forks enter a hostile competition rather than a healthy competition. I've heard more talk about Asterisk forks than I've ever heard about forks of any other other open-source project. I think that this says something about how difficult-to-swallow Digium's dual-license decree is for a lot of prospective contributors/developers. I disagree; if it were that hard to swallow the project would either be 90% digium-written (it's not) or it would be a total flop (again it's not). If you (or someone else reading this post) is in a position to give statistics on what percentage of the code is Digium-written (or Digium-rewritten - in the case where a disclaimer is not obtained for some unpatented work and Digium rewrites the work independently) then I would be thrilled to see it. We see this happen all of the time with the Linux kernel. It happens with HylaFAX. It happened with X. I'm sure it happens a lot with many other open-source software projects. It happens easily and usually is a healthy process because the playing field is even. Agreed. But where are the successful Asterisk forks? I don't know of any successful Asterisk forks (unless http://www.asteriskwin32.com is considered successful - although I'll admit that I'm not really in-the-know). But this was my point: that the way things were set up by Digium makes a successful fork difficult. Digium always has an upper-hand, and things were set up intentionally this way. Again, I don't take particular issue with this. I'm just trying to explain why forking Asterisk would not be a particularly easy task. Of course, this healthy forking cannot be done with Asterisk because Digium will not accept any non-disclaimed code into their repository. ... What you'd described about distribution-maintained patches has nothing to do with this. Digium could take a distribution-maintained patch and rewrite it into Asterisk proper under the dual license (as could any other contributor) and you'd still gain the benefit of the patch. I'm not sure I see where you're going here. If you (or someone else reading this) has the necessary information to provide statistics on how what percentage of the code comes from rewrites of non-disclaimed code, then I would be particularly interested in hearing it. I suspect, though, that it is a rather small - perhaps insignificant - amount. But, yes, providing that there is not a patent involved - yes, the work could be rewritten and integrated. But this was my point: that given the right environment forks can benefit from each other. The one thing that an Asterisk fork can never do, though, is relicense itself. Only Diguim can do that. If Digium had wanted an equal footing in this regard then Asterisk would be LGPL or BSD or something a bit more liberal. So if I'm a manufacturer of PBXes and have some proprietary IP that I do not wish to be GPLed, then if I want to use Asterisk somehow, then I can really only work with Digium for licensing. All of the other forks will be license-prohibitive. I have to admit that I know quite a few people with their own modules and such to replace what they feel is bad code and just won't contribute it back to Asterisk due to the friction they've received about the patch. I, on the other hand, tend to bitch loud and continuously enough and wear them down to the point of accepting it. :-) So we're not in disagreement, it would seem. Getting code contributions into Digium's Asterisk codebase is not something that many average people are going to want to undergo. From what I've seen, friction is a bit light of a term for it. It seems much more
RE: [Asterisk-Users] DNIS and DID seeking confirmation
DID number is the number commonly assigned to a PSTN trunk. DNIS and DID may be the same. DNIS refers to the Dialed Number that is passed as signaling with the call (or on ss7). Most calls have ANI and DNIS. Your extensions look ok, assuming that the carrier sends the digits that match. What Asterisk looks for is determined by how you have signaling setup in your config for the card(s) that you have installed. So, this must match the signaling on the carrier side. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Millican Sent: Monday, June 13, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DNIS and DID seeking confirmation Hello all, After much googling I have come to the conclusion that in asterisk land DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are used rather interchangeably. If this is an incorrect assumption Please correct me. Based on this assumption if I have everthing set up to land in the [incoming] context and an 800# such as 1-800-123-4567 with 4 digit DNIS I can have an entry in my incoming context exten = _4567, 1, do something this is where the call to my 800 number will land regardless of which trunk the call comes in on. Like wise if I have a DID number 456-7891 with an exten= _7891,1,do something else this will also work. Is this correct or am I way off base? Also what is Asterisk looking for as far as a delimiter or is that in a config file? Something like Seize (Wink) DNIS (Wink) ANI (Wink) Answer or Seize (*) DNIS (*) ANI (*) Answer John M ___ 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] Asterisk code
On Monday 13 June 2005 12:06, Matt Riddell wrote: Race Vanderdecken wrote: Also subscribe to the asterisk-dev mail list. Watch it for a couple of days before you ask a question or they will eat your lunch. Or even more likely, eat you for lunch! :D Phew! I thought lunches was going to start disappearing... -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
On Monday 13 June 2005 12:38, The VoIP Connection wrote: This is a very interesting converation, but it seems like the BIZ forum might be more appropriate... How on earth is this a business-related discussion? -dev would have been my guess. :-) -A. ___ 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] T1 multiplexer (or ?) for failover in large installation
The box that you are talking about sounds a lot like a DACS. You might google around on that term to see if any might have automatic failover. A DACS can be reconfigured to cross-connect various DS0s on the fly - although, no matter how fast the switchover, the carrier will always see that something happened. Depending upon what type of signalling you are using, the trunks could end up out of service for a period longer than it takes to switch to the backup server. Also, the system can potentially fail in several ways. At the T1 level, at the trunk level and at the application level. Depending upon the nature of the failure, the one-box does it all solution seems unlikely to work - at least not by itself. -Original Message- From: Mike Sent: Mon, June 13, 2005 11:35 am Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ 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] snom 190: dial tone without registration?
Gavin Hamill [EMAIL PROTECTED] wrote: On Monday 13 June 2005 16:42, alan wrote: I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. snipped enforce SIP channel IP restrictions in Asterisk without host=ipaddr, or get the snom 190 to stop complaining when it's not registered /snipped How about tackling this with iptables and matching specific IP addresses on specific MAC addresses? This solves part, but not all, of the problem. This ensures that only authorized devices can connect to asterisk, and that their IP addresses are also correct. But it doesn't force each device to use only its assigned sip channel. (That is: with dynamic IP registration, a valid IP/MAC could be configured with another device's SIP registration information, and steal calls which should be going to the other device.) I suppose iptables in combination with sip secrets should be enough. But realistically, I can already do what I want the way I want to do it, with the SPA-841. I mostly need to decide: if this feature is lacking, is it enough for me to prefer the Sipura over the snom? Thanks again, Alan Ferrency pair Networks, Inc. [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] More on the IAD connection
As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least workable? Thanks, -Corwin Nichols ___ 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] Cepstral partnership with Digium
You will be able to purchase Cepstral voices from Digium just like you dor for G729 already. I would guess it's 1 way to show the power of asterisk by putting all the TTS orders thru a company such as Digium. ___ 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 on the IAD connection
AdTran can come in either flavor depending on the modules they install. It can dump analog lines or it can be fully digital and split off voice T. I would recommend the digital domain for sure. Get yourself a Digium T1 card and keep everything digital. Get a block of DIDs (20 is the norm for XO) and you will have a great solution. Don't bother with analog lines. They work but consider it this way. T1 card = around $600 and can support up to 23 voice channels with 1 control data channel. 4 Port Analog - $330 - will never support more than 4 ports and IRQ becomes issue as you add more cards Obviously with a split T, you are not coing to use all 23 channels of voice. However, you will have room to grow, get better features, and avoid IRQ problems (probably). That is my $0.02. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corwin Nichols Sent: Monday, June 13, 2005 9:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] More on the IAD connection As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least workable? Thanks, -Corwin Nichols ___ 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] Re: Re: Digium Website Update: Asterisk Business Edition
Andrew Kohlsmith [EMAIL PROTECTED] writes: ABE is a VERY SPECIFIC version of HEAD (or is it STABLE?) with features CUT OUT and nothing added that isn't in HEAD already. This is what I mean with a custom set of features. I never claimed anything was added. I totally fail to see the problem here. The problem arises in two specific areas. Digium needs to hold the copyright of the entire core code base to be able to use a different license. This means that we cannot depend on f.ex sndfile or any other gpl project to do a specific job. Code reuse becomes a thing you cannot take advantage of. Digium cannot ship a proprietary product that includes gpl code and this means that we have to do a lot more work instead of using proven stable free code in the core of asterisk. The other problem is the issue that free software developers are mostly (in my experience) not happy with the fact that their code would be used in proprietary software. It conflicts with the whole religion of free software. This means that fewer contributions would be expected and the development process goes slower. I can only speak for myself, but please understand the clear conflict with the whole philosophy of free software. This is exactly what they are doing. They are supporting a very specific branch with an eye for stability and repeatability. Yes, and as I tried to say; offer support on a said set of features. It can also be a shape asterisk must be in, but it doesn't have to be non free. [..] Oh, it's CVS HEAD from 20050612 and anyway... what? oh, [..] libc? [..] Six what? [..]. Digium's avoiding all this bullshit. It's a specific version of Asterisk compiled by them. This is a good thing, not a bad thing. They can still do this with free software. You can choose to offer support on what you want, pre compiled versions or not, but this whole idea of dual licensing is hurting us, in my opinion. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel modules
I can't load module wcfxs or wcfxo with modprobe command, I don't have any error message, and when i try to start zaptel I've the error below when : ZT_CHANCONFIG failed on channel 4: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Anyone can help me? Thanks. ___ 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] Zaptel modules
as the error says did you forget how to configure FXO and FXS interfaces signalling? post your configs if in doubt... best regards On 6/13/05, Bouchra Benyelloul [EMAIL PROTECTED] wrote: I can't load module wcfxs or wcfxo with modprobe command, I don't have any error message, and when i try to start zaptel I've the error below when : ZT_CHANCONFIG failed on channel 4: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Anyone can help me? Thanks. ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] Re: Digium Website Update: Asterisk Business Edition
On Mon, 2005-06-13 at 18:20 +0200, Esben Stien wrote: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes: Protecting freedoms by putting limits on (thus restricting freedoms). Interesting concept. I need to repeat here. The gpl's purpose is to protect the freedoms that comes with free software. So, you have only the freedoms that comes with free software as defined by the FSF. You are not allowed to do what you like. You are constrained to the freedoms that follows the software and I think that is a very interesting concept. Its not a freedom if its a limit. That is my point. The GPL doesnt give freedoms it takes them away by putting limits on other peoples code, not the original authors. Now if the author is ok with infringing on the rights of others then the GPL is a good choice, however if the original author is truely fore freedom in the code process, not the double speak freedom that FSF talks about (where freedom means taking away abilities) then they should not follow like sheep and repeat what the FSF says (which on its face is an outright lie since its not freedom that it grants). copyright and license to use are different. I never claimed otherwise. If you were the person that was quoting the FSF as fact then you did. Too bad it got cut out, but you can always go back to the original post that was claiming freedom means putting limits on people other than yourself. You can technically put software out there with no copyright but under the gpl license Then there would be no one to enforce the license, which would be bad. Why do you cut out what I said when I addressed that point? I am begining to think that you are doing it intentionally now. it only restricts *their* code (ie modifications). Yes, but we also want all modifications to be free 'we' or you specifically? We is quite a loaded word. The FSF makes a false claim that it *protects* freedoms, when all it does is limit the freedoms of others to write code. Specifically if I take a program and modify it, the original is still under whatever license I got it in, but *my* code, the modifications are MINE not the original authors. The original author has NO right to claim that it is their work, nor do they have copyright on *my* code. But by releasing it under a GPL they can force me to use a license that I may not agree with. This is the reason that I dont contribute to GPL products, I dont like the idea of someone else dictating to me how I will distribute *my* code. The default GPL makes it a lciense violation to run GPL code on a commercial (or even BSD) system. Extra stuff has to be put into the GPL license to say 'its ok if you link this against non GPL libraries and such'. That is not the default, so technically unless someone did that putting a stock GPL license has other limitations on its mere use. At least historically libc on aix, hpux, sunos (4/5), irix were all not GPL libc (I dont know with solaris now they added a bunch of gpl stuff at one point). If any of the GPL licensed software did not take an overt action to say its ok to run it on those operating systems then its a license violation. That level of selective enforcement also calls into question the legal standing of the license (if certain sections are not enforced the whole agreement can be voided on first court challenge). http://www.gnu.org/licenses/gpl-faq.html#TOCLinkingOverControlledInterface for linking proprietary code to libraries - overt actions required to make it work right http://www.gnu.org/licenses/gpl-faq.html#TOCGPLCommercially for reading up on how the license affects others who write code later http://www.gnu.org/licenses/gpl-faq.html#TOCGPLIncompatibleAlone for reading on how you cant really link against libc on a commercial operating system (or anything with a license that is not compatible with the GPL, which BSD isnt becuase it allows someone to take it, write *their own* code in addition to it and not give *their own* code out. Thus by default you cant run GPL software on a BSD licensed system, nor any commercial system *unless* the developer took an overt action to say this is ok (default GPL it is not ok). http://www.gnu.org/licenses/gpl-faq.html#TOCDistributeWithSourceOnInternet for the lack of personal privacy that the GPL forces on those that choose to release under it, specifically you *must* (section 3) provide a mailing address. If you value your privacy and dont want everyone to have your address you must pay extra to get a po box so that you, as the author of the software, can comply with section 3 of the GPL - providing copies by mailorder on physical media. This is *required* not optional. The list goes on... the cost of restricting freedoms on others and what they can do with their code Yes, by using the GPL you restrict everyone to the four freedoms defined in the free software definition. This is exactly what we want. Again with the
[Asterisk-Users] Asterisk connecting remote villages in western Uganda
Hi, I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa. The system is up and running since this past Wednesday (June 8th). We have installed 5 units, 4 of which are in villages with with no access to power. The system provides Internet access and phone capabilities to the users. Phone calls among the connected villages are free of charge, with the ability to place and receive calls to / from the Ugandan phone network and voice mail boxes for each station. The systems are linked using 802.11 WiFi links. For more information please have a look at the following links: For more detailed information and pictures of the Uganda deployment: http://www.inveneo.org/?q=uganda For more information about the solution we have built and implemented, here is a link to our PDF datasheet: http://www.inveneo.org/download/inveneoDatasheet.pdf And of course our website: http://www.inveneo.org/ Thank you! Mark Mark Summer co-founder, Inveneo web: http://www.inveneo.org phone: +1-415-901-1969 x 1200 FWD: 603303 cell: +1-415-867-9751 email: [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
RE: [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation
Use Adtran Atlas 800. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of qrss Sent: June 13, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation The box that you are talking about sounds a lot like a DACS. You might google around on that term to see if any might have automatic failover. A DACS can be reconfigured to cross-connect various DS0s on the fly - although, no matter how fast the switchover, the carrier will always see that something happened. Depending upon what type of signalling you are using, the trunks could end up out of service for a period longer than it takes to switch to the backup server. Also, the system can potentially fail in several ways. At the T1 level, at the trunk level and at the application level. Depending upon the nature of the failure, the one-box does it all solution seems unlikely to work - at least not by itself. -Original Message- From: Mike Sent: Mon, June 13, 2005 11:35 am Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ 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] Hiss patch
In bug 0002863 a patch is mentioned that sends hiss every 20 seconds, does anyone know who wrote this or where it is available at? Scott England ___ 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] DID in AMP with 2+ incoming lines
Hello, I know that I can have DID on a single line, but will AMP support 2+ lines with DID? Has anyone tried this? Straight forward? Thank you, Tomas ___ 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] Re: Digium Website Update: Asterisk Business Edition
quote who=trixter http://www.0xdecafbad.com; Protecting freedoms by putting limits on (thus restricting freedoms). Interesting concept. It maybe an interesting concept, but it is absolutely true. True anarchy (no rules what so ever) cannot exist. Your freedom to kill me would impose on my freedom to live. Lift all laws and the law of the universe seems to come into play. The strong rules the weak. You end up with a dictatorship. To keep something free, there must be a law stopping it from not becoming not free. (bad english, but there it is. :) ) -- And, did Guloka think the Ulus were too ugly to save? -Centauri ___ 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 connecting remote villages in westernUganda
Mark, This is a wonderful thing to do for underserved societies like Uganda. The datasheet you have provided and the layout could be the model for many other developing societies both In Africa as well as central and South America. Kudos to Inveneo.org under your able leadership. Keep up the good work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Summer Sent: Monday, June 13, 2005 1:56 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk connecting remote villages in westernUganda Hi, I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa. The system is up and running since this past Wednesday (June 8th). We have installed 5 units, 4 of which are in villages with with no access to power. The system provides Internet access and phone capabilities to the users. Phone calls among the connected villages are free of charge, with the ability to place and receive calls to / from the Ugandan phone network and voice mail boxes for each station. The systems are linked using 802.11 WiFi links. For more information please have a look at the following links: For more detailed information and pictures of the Uganda deployment: http://www.inveneo.org/?q=uganda For more information about the solution we have built and implemented, here is a link to our PDF datasheet: http://www.inveneo.org/download/inveneoDatasheet.pdf And of course our website: http://www.inveneo.org/ Thank you! Mark Mark Summer co-founder, Inveneo web: http://www.inveneo.org phone: +1-415-901-1969 x 1200 FWD: 603303 cell: +1-415-867-9751 email: [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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Unable to support trunking .... without zaptel timing
When I start Asterisk, I receive these errors: Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'zoom_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'zoom_trunk' without zaptel timing I have a TE110P card and IAX setup between two servers that appear to work fine. The only reason this has come to my attention is when I was trouble shooting an error with our Mitle SX200 that was reporting: T1/BRI Card at 02 06 00 00 Has exceeded the maint loss frame threshold. My main question is in regards to the timing error but if anyone has any Mitel experience (scott w!) that has seen this other error.= I'd be happy to hear about it Here is zapata.conf: [trunkgroups] [channels] musiconhold=default busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes callgroup=1 pickupgroup=1 immediate=no signalling=em_w context=zap-incoming group = 1 channel = 1-17 group = 2 channel = 21-24 Here is zaptel.conf loadzone= us defaultzone = us span=1,1,0,d4,ami em=1-24 Here is lsmod: Module Size Used by snd_pcm_oss47648 0 snd_pcm83336 1 snd_pcm_oss snd_timer 23812 1 snd_pcm snd_page_alloc 9604 1 snd_pcm snd_mixer_oss 16896 1 snd_pcm_oss snd51044 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 10080 1 snd ipv6 232320 10 wcte11xp 25760 21 zaptel224132 43 wcte11xp i2c_i8018204 0 i2c_core 21392 1 i2c_i801 hisax 483920 0 crc_ccitt 2176 2 zaptel,hisax isdn 128716 1 hisax slhc6912 1 isdn ext3 124424 4 jbd55064 1 ext3 genrtc 9608 0 evdev 9088 0 pcspkr 3940 0 parport_pc 33220 0 parport33864 1 parport_pc piix9988 0 [permanent] ehci_hcd 30728 0 pci_hotplug31152 0 uhci_hcd 29584 0 usbcore 107896 3 ehci_hcd,uhci_hcd tg379364 0 ide_generic 1408 0 [permanent] ide_cd 38020 0 ide_core 115668 3 piix,ide_generic,ide_cd cdrom 36384 1 ide_cd font8448 0 ata_piix9092 10 libata 42756 1 ata_piix unix 26804 12 ___ 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] Re: Voicemail and MS Exchange Synchronizatio n
-Original Message- From: Iassen Hristov [mailto:[EMAIL PROTECTED] Does this matter? All we are saying is that Exchange supports IMAP and we would use IMAP as the protocol to delete the message from the user's mailbox. How does the user access his mailbox is his choice. I think two threads of discussion got crossed. Somewhere along the line someone brought up the idea of having Asterisk act like an IMAP *server* where people could retrieve their voicemails. ___ 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 and MS Exchange Synchronization
-Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Saturday, June 11, 2005 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail and MS Exchange Synchronization On 6/10/05, Dean Collins [EMAIL PROTECTED] wrote: Actually I think that has changed to 75gb now (or about to change). Really? any links to support that? Since when is Micro$oft so easy on giving up on licensing fees? I'm curious, too. If this is true it might save us a lot of pain, upgrade wise. We've been looking at moving away from Exchange entirely because of that damn 16-gig limit, and Exchange Enterprise Edition is just too expensive. ___ 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] Re: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On Monday 13 June 2005 12:38, The VoIP Connection wrote: This is a very interesting converation, but it seems like the BIZ forum might be more appropriate... How on earth is this a business-related discussion? -dev would have been my guess. :-) Maybe we need an anti-biz list for this kind of thing. ;) ___ 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] Re: Re: Digium Website Update: Asterisk Busi ness Edition
-Original Message- From: Esben Stien [mailto:[EMAIL PROTECTED] The other problem is the issue that free software developers are mostly (in my experience) not happy with the fact that their code would be used in proprietary software. It conflicts with the whole religion of free software. Well, yeah, that's the whole problem, isn't it? You can't follow the religion of free software and still run a company that pays the bills. You have to compromise somewhere. Either you go out of business, or you tick off some of the open source purists. Interesting perspective on this from Forbes: http://www.forbes.com/technology/2005/05/26/cz_dl_0526linux.html ___ 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] Components and suggestions for an asterisk server with 9 to 17 POTS.
A small simple non Dell system and T1/E1 card. If you can even think about 17 lines then start with a PRI. Most PRIs can be ordered as small as 4 lines voice and 768 data that leaves ~8 voice channels open. I can give you system specs off list as the change often. On 6/13/05, Ken Dresdell [EMAIL PROTECTED] wrote: Hello, What would be the simplest and the cheapest solution to get an Asterisk server working with 9 to 17 POTS? Because for 1-8 POTS we are using 1 or 2 Digium TDM cards and past 17 POTS in our area it is economic to use a PRI. We are looking for a hardware solution on our side instead of using did provider Thanks in advance Ken ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] VOIP-INFO.ORG website bug
When I try to fing group/call pickup command in www.voip-info.org and made a search like *8 I got an error message. Regards, -- Ing CIP Alejandro Celi Mariátegui [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
RE: [Asterisk-Users] Re: Digium Website Update: Asterisk BusinessEdition
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Tuesday, 14 June 2005 3:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Digium Website Update: Asterisk BusinessEdition On Mon, 2005-06-13 at 18:20 +0200, Esben Stien wrote: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes: Protecting freedoms by putting limits on (thus restricting freedoms). Interesting concept. I need to repeat here. The gpl's purpose is to protect the freedoms that comes with free software. So, you have only the freedoms that comes with free software as defined by the FSF. You are not allowed to do what you like. You are constrained to the freedoms that follows the software and I think that is a very interesting concept. Its not a freedom if its a limit. That is my point. The GPL doesnt give freedoms it takes them away by putting limits on other peoples code, not the original authors. Now if the author is ok with infringing on the rights of others then the GPL is a good choice, however if the original author is truely fore freedom in the code process, not the double speak freedom that FSF talks about (where freedom means taking away abilities) then they should not follow like sheep and repeat what the FSF says (which on its face is an outright lie since its not freedom that it grants). Ahem, The GPL is about the freedown of the code, not the freedom of the individual copyright and license to use are different. I never claimed otherwise. If you were the person that was quoting the FSF as fact then you did. Too bad it got cut out, but you can always go back to the original post that was claiming freedom means putting limits on people other than yourself. You can technically put software out there with no copyright but under the gpl license Then there would be no one to enforce the license, which would be bad. Why do you cut out what I said when I addressed that point? I am begining to think that you are doing it intentionally now. it only restricts *their* code (ie modifications). Yes, but we also want all modifications to be free 'we' or you specifically? We is quite a loaded word. The FSF makes a false claim that it *protects* freedoms, when all it does is limit the freedoms of others to write code. Specifically if I take a program and modify it, the original is still under whatever license I got it in, but *my* code, the modifications are MINE not the original authors. The original author has NO right to claim that it is their work, nor do they have copyright on *my* code. But by releasing it under a GPL they can force me to use a license that I may not agree with. This is the reason that I dont contribute to GPL products, I dont like the idea of someone else dictating to me how I will distribute *my* code. The default GPL makes it a lciense violation to run GPL code on a commercial (or even BSD) system. Extra stuff has to be put into the GPL license to say 'its ok if you link this against non GPL libraries and such'. That is not the default, so technically unless someone did that putting a stock GPL license has other limitations on its mere use. At least historically libc on aix, hpux, sunos (4/5), irix were all not GPL libc (I dont know with solaris now they added a bunch of gpl stuff at one point). If any of the GPL licensed software did not take an overt action to say its ok to run it on those operating systems then its a license violation. That level of selective enforcement also calls into question the legal standing of the license (if certain sections are not enforced the whole agreement can be voided on first court challenge). http://www.gnu.org/licenses/gpl-faq.html#TOCLinkingOverControl ledInterface for linking proprietary code to libraries - overt actions required to make it work right http://www.gnu.org/licenses/gpl-faq.html#TOCGPLCommercially for reading up on how the license affects others who write code later http://www.gnu.org/licenses/gpl-faq.html#TOCGPLIncompatibleAlone for reading on how you cant really link against libc on a commercial operating system (or anything with a license that is not compatible with the GPL, which BSD isnt becuase it allows someone to take it, write *their own* code in addition to it and not give *their own* code out. Thus by default you cant run GPL software on a BSD licensed system, nor any commercial system *unless* the developer took an overt action to say this is ok (default GPL it is not ok). http://www.gnu.org/licenses/gpl-faq.html#TOCDistributeWithSour ceOnInternet for the lack of personal privacy that the GPL forces on those that choose to release under it, specifically you *must* (section 3)
Re: [Asterisk-Users] Group/Broadcast Voicemail
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096] in an earlier replay so I did.
static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
Yeah, if you get the Microsoft Partners Newsletter emails they reported the 75 GB expansion today. Increased Storage Limit in Exchange Server Standard Edition Get more out of mission-critical email. In the fall of 2005 the storage limit for Exchange Server 2003 Standard Edition will increase to 75 gigabytes. It took me a while to find it through the links the give you. But here it is http://www.microsoft.com/exchange/downloads/2003/sp2/overview.mspx Then scroll to the bottom of the page... Mailbox Advancements Drive down operational costs and the complexity of your messaging environments with advances such as: . Increase in mailbox storage size limits to 75 gigabyte (GB) for Exchange Server 2003 Standard Edition in response to customer feedback and evolving mailbox storage needs. . New offline address book format offers significantly improved performance. . Cache mode enforcement with added flexibility. You now can force clients into cached mode to help improve performance and increase the number of active users per server. This is especially beneficial to organizations seeking to further site and server consolidation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Brodbeck Sent: Monday, June 13, 2005 4:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Saturday, June 11, 2005 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail and MS Exchange Synchronization On 6/10/05, Dean Collins [EMAIL PROTECTED] wrote: Actually I think that has changed to 75gb now (or about to change). Really? any links to support that? Since when is Micro$oft so easy on giving up on licensing fees? I'm curious, too. If this is true it might save us a lot of pain, upgrade wise. We've been looking at moving away from Exchange entirely because of that damn 16-gig limit, and Exchange Enterprise Edition is just too expensive. ___ 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] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ 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 Evening in Melbourne (again!)nextThursday
Make it there if you can! PaulH On Fri, 2005-06-10 at 13:16 +1000, jurgen wrote: Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for the second in an irregularly scheduled casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. About a dozen of us got together a couple of months ago, and had a good time chatting about all things Asterisk. Beverages were also consumed. 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 16th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place, people seemed to like it last time. Also, like last time, I'll get an old phone and put it on the table, so those of us who haven't met will be able to recognise each other. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED] Hope to see you there! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ 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. ___ 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] MCI vs. XO/Allegiance
Wiley Siler wrote: Anyone out there using ISDN PRI from either MCI or XO/Allegiance? We have a DS-3 full of PRI from X/O. They work great, mostly, but their tech support sucks. They screw up number ports all the time and about every week there is some local number I can't dial to via XO which once I open a ticket mysteriously gets fixed without a good explanation. Eventually everything works, but you have to beat on them continously to get things done. Better than dealing with SBC though. David ___ 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] MCI vs. XO/Allegiance
I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ 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] MCI vs. XO/Allegiance
I prefer MCI since we use their pri and internet. MCI's support is very pro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Coulson Sent: June 13, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance Wiley Siler wrote: Anyone out there using ISDN PRI from either MCI or XO/Allegiance? We have a DS-3 full of PRI from X/O. They work great, mostly, but their tech support sucks. They screw up number ports all the time and about every week there is some local number I can't dial to via XO which once I open a ticket mysteriously gets fixed without a good explanation. Eventually everything works, but you have to beat on them continously to get things done. Better than dealing with SBC though. David ___ 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] snom 190: dial tone without registration?
You should use DHCP to enforce IP address to MAC binding when the phones boot. And then let the phones register and use host access (deny/permit) permissions in peer section to restrict by IP address/mask. alan wrote: Gavin Hamill [EMAIL PROTECTED] wrote: On Monday 13 June 2005 16:42, alan wrote: I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. snipped enforce SIP channel IP restrictions in Asterisk without host=ipaddr, or get the snom 190 to stop complaining when it's not registered /snipped How about tackling this with iptables and matching specific IP addresses on specific MAC addresses? This solves part, but not all, of the problem. This ensures that only authorized devices can connect to asterisk, and that their IP addresses are also correct. But it doesn't force each device to use only its assigned sip channel. (That is: with dynamic IP registration, a valid IP/MAC could be configured with another device's SIP registration information, and steal calls which should be going to the other device.) I suppose iptables in combination with sip secrets should be enough. But realistically, I can already do what I want the way I want to do it, with the SPA-841. I mostly need to decide: if this feature is lacking, is it enough for me to prefer the Sipura over the snom? Thanks again, Alan Ferrency pair Networks, Inc. [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] MCI vs. XO/Allegiance
Well, the fact that two negatives for XO and a positive for MCI all came at once says a lot to me. Interestingly enough their SLA reads... * 24/7/365 Network Monitoring and Service. If for some reason your network is having problems, the chances are XO will know about it before you do and respond before any problems become critical. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue, Jr. Sent: Monday, June 13, 2005 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ 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] MCI vs. XO/Allegiance
Let me throw another complaint against XO on the table. They actually shut off the wrong T1 and they transferred all of the DIDs to the T1 they shut off! how screwed up is that? We are now about 2 years later and their billing department still calls us every month for nonpayment of the T1 that they turned off. We also have 6 T1s through MCI, all long distance. They have been much better to deal with. MATT--- -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Monday, June 13, 2005 7:34 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MCI vs. XO/Allegiance Well, the fact that two negatives for XO and a positive for MCI all came at once says a lot to me. Interestingly enough their SLA reads... * 24/7/365 Network Monitoring and Service. If for some reason your network is having problems, the chances are XO will know about it before you do and respond before any problems become critical. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue, Jr. Sent: Monday, June 13, 2005 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ 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] Should I choose DSL @ 1.5 or a full T1?
You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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 ___ 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] problem with pf and asterisk
current setup SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address or #2 asterisk trying to get back to me as 192.168 on public internet.. got canreinvite=yes and no. nat=yes qualify=1000 externaladdr=IP of (em1) localnet=172.16.0.0/12 i would need help form someone who did a sismilar setup.. i do run carp and pfsync also on the FW. mirrored to FW2 down ATM... anyhelp appreciated.. banging head on the wall for 2 weeks now.. ___ 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] Should I choose DSL @ 1.5 or a full T1?
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
The only technical difference between a T1 and SDSL is how it's physically
delivered to the customer, what usually happens is that a T1 is not oversold,
while an SDSL is oversold anywhere from 8:1 to 3:1.
ADSL is full duplex as well, if you don't know how to do QOS then it will feel
like it's half duplex, but it's not. I have 1000/320 ADSL that I can use full
bandwidth both ways.
Marcelo Pacheco
Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a shared
path on sdsl.
The s in sdsl means symmetrical, not duplex, that is that the signaling
rate is the same in either direction, but still half duplex.
For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
nature of the traffic, unlike most internet that is download-centric.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
Barton Fisher wrote:
I'm looking to expand my bandwidth for my Asterisk PBX. Why should I
choose a T1 over DSL for my asterisk server? I found someone
offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
Loops. Is this a good deal?
Thanks
Bart
-
-
--
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Where are you located? What CLEC gives you a T-1 for $290?
FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
getting a break for having a voice and a data circuit broken out from
one fiber drop, but that's what I'm paying here in Orange County. Also,
I had a business cable modem before, which was *allegedly* not shared
for business customers (suspicious) and the throughput was a roller
coaster, as was the latency. The DS-1 cleared all that up.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Are you sure? Everything I have seen says SDSL = Full Duplex. That being achieved by dropping the pair that provided voice and using it for signalling. Where ADSL utilizes unoccupied frequencies and averts conflict with analog voice frequencies, SDSL takes over the whole line. SDSL eliminates analog voice capabilities in favor of full-duplex data transmission. No splitter, no analog voice-nothing but data. As a decent alternative to T1, SDSL has gotten a fair amount of attention from Competitive Local Exchange Carriers. Excerpt from http://www.isp-select.com/SDSL.htm Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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 ___ 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] Should I choose DSL @ 1.5 or a full T1?
Damon, I have no idea where you are getting your information from, but what you said makes no sense. DSL based lines, be it ADSL or SDSL, are based upon a connection technology in the ATM family. As a result, the upstream and downstream of the connection can be controlled seperately. If someone offers you a 1.5 SDSL connection, it doesn't actually mean that you have 2x768kbps, it may actually mean that you have 2x1.5Mbps. However, that speed is only towards your internet provider, what you get beyond that point would be bound to your ISP's SLA and contract. Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). Now, lets do a little math (23+1)*64 = 1536kbps = 1.536Mbps, hence the speed for a single T1 circuit. Now, if you have a T1 installed, and you are currently using 512kbps of upload, it means that you are physically using 8 lines out of the 23 data lines for uploading. You can then use the rest to what ever purpose you want, but while those lines are in play, you won't be upload another 512kbps on the same lines. The reason for that is that each of these lines operates on a seperate Time Slot within the physical layer. Once a Time Slot is taken for a specific data flow, it can't be used for another data flow. This actually means that a T1 will give you a shared 1.5Mbps towards your ISP, with speed that vary on the upload and download, according to your usage. While when using a DSL, your quality of service for the connection to the ISP is described by the policy of connection. In many countries (eg: Israel, Turkey, China, UK), DSL lines are actually ADSL lines, where the downstream is around 1.5Mbps while the uplink is around 128kbps (just enough to do a little VoIP). Last time I was in the UK, about 4 weeks ago, I noticed they are now selling 8Mbps ADSL connection to your house, however, the uplink is 512kbps. I would suggest that you get all the information from your providers regarding the type of services rendered on the SDSL line, and make sure that it's the right one for you. Nir S Damon Estep wrote: You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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] Should I choose DSL @ 1.5 or a full T1?
Marcelo Pacheco wrote:
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or
asymetrics of the actual physical line.
The only technical difference between a T1 and SDSL is how it's physically
delivered to the customer, what usually happens is that a T1 is not oversold,
while an SDSL is oversold anywhere from 8:1 to 3:1.
That is correct in the genereal idea, however, as xDSL technologies are
switched technologies, unlike
cable (DOCSIS) technologies, the fact that you are overloaded 8:1 or 3:1
will not really matter. As long
as your equipment supports QoS correctly, you shouldn't have a problem.
ADSL is full duplex as well, if you don't know how to do QOS then it will feel
like it's half duplex, but it's not. I have 1000/320 ADSL that I can use full
bandwidth both ways.
ADSL appears to be half-duplex only due to the fact that most ISP's
misconfigure the modems and routers.
As a rule of thumb, the modem/router can be re-configured to utilize
both channels to the fullest, but again, this
must rely on the fact that your ISP's equipment supports QoS at the
switch level correctly.
Nir S
Marcelo Pacheco
Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a shared
path on sdsl.
The s in sdsl means symmetrical, not duplex, that is that the signaling
rate is the same in either direction, but still half duplex.
For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
nature of the traffic, unlike most internet that is download-centric.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
Barton Fisher wrote:
I'm looking to expand my bandwidth for my Asterisk PBX. Why should I
choose a T1 over DSL for my asterisk server? I found someone
offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
Loops. Is this a good deal?
Thanks
Bart
-
-
--
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Where are you located? What CLEC gives you a T-1 for $290?
FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
getting a break for having a voice and a data circuit broken out from
one fiber drop, but that's what I'm paying here in Orange County. Also,
I had a business cable modem before, which was *allegedly* not shared
for business customers (suspicious) and the throughput was a roller
coaster, as was the latency. The DS-1 cleared all that up.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance we have been using XO/Allegiance for over 3 years and have had no problems. I can't compare to MCI but we also had a sprint t1 that we had to get remove due to them being bad in billing and also not very reliable for faxing. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 13, 2005 6:59 PM Subject: [Asterisk-Users] MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Should I choose DSL @ 1.5 or a full T1?
Nir Simionovich wrote: Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ 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] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance Sprint nevermore... I switched voer to Sprint a few years ago and they literally dropped service form under us. It was during that Sprint ION fiasco. They sold to me, installed, and literally terminated the service10 days later. The whole time they were working the install for me, the other side of the company was going belly up. Poo poo on sprint... w From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel BatistaSent: Monday, June 13, 2005 5:31 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] MCI vs. XO/Allegiance we have been using XO/Allegiance for over 3 years and have had no problems. I can't compare to MCI but we also had a sprint t1 that we had to get remove due to them being bad in billing and also not very reliable for faxing. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 13, 2005 6:59 PM Subject: [Asterisk-Users] MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Should I choose DSL @ 1.5 or a full T1?
BTW - The Speakeasy SDSL connection I originally posted about is
delivered via Covad.
The SLA (some of it at least)
Average Network Delivery and Delay2 - Further proof that Covad has
confidence in the performance of our network.
Delivery - 99.9% successful delivery of all data packets sent from your
location over the Covad network, or you will be eligible for a credit of
up to 10% of your monthly service fee.
Delay - 110 millisecond average for the round trip of a message sent
from your location to a test point on the Covad network, or you will be
eligible for a credit of up to 10% of your monthly service fee.
SLA can be found here...
http://www.covad.com/products/access/telespeed/details.shtml#sla
Being only 4000 feet from the Central Office, this works very well for
me.
I have not been able to figure if QoS is possible yet.
Haven't figured out the examples from the Wiki for QoS via HFB (I think)
and no answer from techs yet.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: Monday, June 13, 2005 6:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Marcelo Pacheco wrote:
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of
that is half-duplex is 802.11{b,g}.
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or asymetrics of the actual physical line.
The only technical difference between a T1 and SDSL is how it's
physically delivered to the customer, what usually happens is that a T1
is not oversold, while an SDSL is oversold anywhere from 8:1 to 3:1.
That is correct in the genereal idea, however, as xDSL technologies are
switched technologies, unlike cable (DOCSIS) technologies, the fact that
you are overloaded 8:1 or 3:1 will not really matter. As long as your
equipment supports QoS correctly, you shouldn't have a problem.
ADSL is full duplex as well, if you don't know how to do QOS then it
will feel like it's half duplex, but it's not. I have 1000/320 ADSL
that I can use full bandwidth both ways.
ADSL appears to be half-duplex only due to the fact that most ISP's
misconfigure the modems and routers.
As a rule of thumb, the modem/router can be re-configured to utilize
both channels to the fullest, but again, this must rely on the fact that
your ISP's equipment supports QoS at the switch level correctly.
Nir S
Marcelo Pacheco
Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a
shared path on sdsl.
The s in sdsl means symmetrical, not duplex, that is that the
signaling rate is the same in either direction, but still half duplex.
For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
nature of the traffic, unlike most internet that is download-centric.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
Barton Fisher wrote:
I'm looking to expand my bandwidth for my Asterisk PBX. Why should
I
choose a T1 over DSL for my asterisk server? I found someone
offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
Loops. Is this a good deal?
Thanks
Bart
-
-
--
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Where are you located? What CLEC gives you a T-1 for $290?
FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
getting a break for having a voice and a data circuit broken out from
one fiber drop, but that's what I'm paying here in Orange County.
Also, I had a business cable modem before, which was *allegedly* not
shared for business customers (suspicious) and the throughput was a
roller coaster, as was the latency.
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Hi David, You are correct, I always get those 2 confused. Thanks for the clearing. Nir S David Coulson wrote: Nir Simionovich wrote: Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David ___ 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] Should I choose DSL @ 1.5 or a full T1?
Not really true about T1 description. When you apply for T1, you need tell vendor if it's channelized or non-ch. If you are going to use it for 1.5M network, you need use unchannelized T1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: June 13, 2005 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Hi David, You are correct, I always get those 2 confused. Thanks for the clearing. Nir S David Coulson wrote: Nir Simionovich wrote: Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David ___ 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] Should I choose DSL @ 1.5 or a full T1?
Leon Sun wrote: Not really true about T1 description. When you apply for T1, you need tell vendor if it's channelized or non-ch. If you are going to use it for 1.5M network, you need use unchannelized T1. T1 is T1. How you use the DS0s delivered across it is up to you. You can mux them out to POTS lines, use them all for data or mix it up and run voice and data over the same T1. Telco vendors don't care what you do with it, unless it's terminating for data/voice in their equipment. Even when you use all 24 channels for data, they still function as 24 distinct DS0 channels as far as timing is concerned. Unlike OC-nc circuits (Where you save some overhead for the sake of being unable to channelize the STS channels) , there is no overhead variation when channelizing a DS-1 versus using a full DS-1 for data. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ 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
