RE: [Asterisk-Users] GUI client for windows for live monitoring/barge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Riley Sent: Tuesday, 13 January 2004 13:02 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] GUI client for windows for live monitoring/barge I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks, This may be a wacky suggestion, might require more resources, but why don't you try setting up your dialplan so that all calls are in conference s with two members, then you can drop in any time you want... Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Version of SJPhone
I got SJPhone to work at home but not at work. It was the first time I ever ran it, so I don't know if what I found is a new error or it has always been there. It turns out that if your computer does not have a gateway (mine at work does not), then SJPhone does not send the correct VIA string to * (it will be missing the IP address of the computer) and *, thusly, won't allow the phone to register. Run winipcfg or ipconfig /all from a DOS prompt to ensure you have a gateway IP address defined on your Windoze box. Lemme know if this was the problem - I'm interested. I reported this bug to SJ yesterday. --On Monday, January 12, 2004 2:40 PM -0500 Darren Nickerson [EMAIL PROTECTED] wrote: Unless I'm missing something, that's true. I also made the mistake of upgrading ... -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: admin To: [EMAIL PROTECTED] Sent: Sunday, January 11, 2004 10:54 AM Subject: [Asterisk-Users] New Version of SJPhone I just installed the new version of SJPhone and it appears that it cannot work with * anymore? /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-Client for Handheld PC
Hi, Yes Telesym, xten and one more I can't remember the name of it, they are all for PPC-only. :( /HHA From: Ray Burkholder [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC Date: Mon, 12 Jan 2004 14:33:39 -0500 What are the ones you found for PocketPC? I guess you've looked at the Telesym site? They have a SIP flavor coming out shortly for some PDA's. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik Andresen Sent: January 12, 2004 05:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-Client for Handheld PC Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _ Let the new MSN Premium Internet Software make the most of your high-speed experience. http://join.msn.com/?pgmarket=en-uspage=byoa/premST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie to asterisk
Dear Sir / Madam, I am a newbie in using Asterisk. I am interested in its SIP. Before I start to use it, I would like to know whether the system can work between two Linuxbox without any FXO and FXS card and just using microphone which connect tothe regularsound card? I am looking into others applicationsand all of them are using at least one FXS card. Sorry for such beginner problem and please help. Thank you very much for your time. Max
RE: [Asterisk-Users] More words for Allison
knot yet. :) cameron. - I like puns. On Mon, 12 Jan 2004, Sean Cheesman wrote: knot n. A unit of speed, one nautical mile per hour thanks to our good friends at reference.com. Are we done yet? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, January 12, 2004 10:10 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] More words for Allison -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 11, 2004 8:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] More words for Allison [...] snip knots per hour I'm a land-lubber, but I think knots is a speed unit (like Miles Per Hour), so I think you want knots here, not knots per hour, if you are talking wind speed. [...] Then stick to being a land lubber. Because you're wrong. A knot is a unit of linear measurement. Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This newbie gives up for now - sadly
As Robert's colleague that owns 7960s I can go on about the superiority of the Cisco phone. The most immediate difference is the look and feel. Everyone that has seen or held my phone says that it is nice. Everyone that picks up a Grandstream phone or looks at one says they are cheap. Grandstream should really consider putting some lead weights in the handset. Hell, the free USB phone from Voiceglo feels better than the Grandstream phone... and that is just the exterior... As soon as they get their problems with SIP functionality and stability sorted, they should spend some time and effort on product design. I understand they are trying to be competitive but people expect a phone to look and feel a certain way. cameron. On Tue, 6 Jan 2004, Robert Hajime Lanning wrote: John, Jared is right. I have a co-worker who has coughed up the money for the Cisco 7960 SIP phones. These have a soft button for Supervised Transfer. And, it works. I only have the Grandstream BT101 phones, and their Transfer button only implements Blind Transfer. So, to get it to work, you will need to upgrade to non-budget phones. Not ideal, but Asterisk does support the feature, just Grandstream does not. quote who=Jared Smith On Tue, 2004-01-06 at 06:20, John Coll wrote: Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. It sounds to me like this is a problem with the Grandstream phones in particular, and not Asterisk. Supervised transfers work *GREAT* with the Cisco 7960 phones... I use them almost every day. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
GotoIfTime application. I'll own up to writing the broken code. The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c 11 Jan 2004 09:19:16 - 1.92
+++ pbx.c 13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
[Asterisk-Users] Nufone.net net wackiness?
I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. My accounts still seem to work, but I wonder WTH is going on? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: problem with safe_asterisk
Hi, Pat Boyle wrote: I have no problems lauching asterisk from the command line . . . asterisk -c However, I'm trying to autostart on boot up, so I'm trying safe_asterisk When I do this, I get: Asterisk ended with exit status 127. Asterisk died with code 127. Aborting. I've looked through the script but can't find what the problem is. I'm running on RedHat Fedora. Could You please have a look in the logfile. Maybe there are some information about the abort. I don't use Fedora but on Debian the log is under /var/log/asterisk/messages HTH, Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone.net net wackiness?
On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. My accounts still seem to work, but I wonder WTH is going on? looks like Jeremy maybe forgot to renew the registration. Looks like it was updated today. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3.3v PCI board - TE410P photo
Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RFC3389 messages with ATA 186
Walt Reed wrote: I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to turn off the RFC3389 support on the ata 186? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Change audiomode to 0x00140014 Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone.net net wackiness?
No he renewed it... but Gododaddy did the transaction over the phone manually and they never posted the payment so they shut it down. It should be back by now. Switch-1 ip is 66.225.202.72 bkw On Tue, 13 Jan 2004, Steven Critchfield wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. My accounts still seem to work, but I wonder WTH is going on? looks like Jeremy maybe forgot to renew the registration. Looks like it was updated today. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone.net net wackiness?
Looks like they went off the air just after my PayPal payment was processed. I gues we wait a couple days to see if Nufone has gone belly up/bankrupt/gone or if this is just a domain name screw up. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
GotoIfTime application. I'll own up to writing the broken code. The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c 11 Jan 2004 09:19:16 - 1.92
+++ pbx.c 13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is critical?
___
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RE: [Asterisk-Users] FS/OS Telephony Summit 2004
Hi I am attending the tutorial day, i am looking forward to it. See you there. Craig. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: 13 January 2004 10:31 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] FS/OS Telephony Summit 2004 Hello * world, i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen from the 16th til 20th january. Together with Christian Richter i will be speaking about * on monday. And we will give an * tutorial on tuesday. I will be presenting some ISDN stuff there, including the quadBRI cards. If you will be there too and want to meet, just let me know. :) Details on the summit can be found at: http://www.guug.de/veranstaltungen/telephony-summit-2004/ best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax
Hi All I have just a quick question regarding app_txfax for Asterisk. When I send a fax from asterisk to a traditional fax machine connected to asterisk via the digium analog card everything works perfectly. However the same fax machine on the public telephoine network results in errors (looks like some sort of training error). My asterisk box is connected to the pstn using an ISDN card. I don't mind trying to fix this myself but I am puzzled by the different behavior experienced when the fax machine is on the digium card and when it is connected to our public carrier, and therefore have no idea where to start. Would someone (Steve Underwood ;-) )mind at least putting me on the right track so I can address this issue? Thanks in advance Steve Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip and x-lite
Hi, Ing Isianto Istiadi wrote: Thanks for the Info, and It worked. But I have a couple of questions: 1. There's an echo. How to get rid of the echo? 2. Is there any way to call from x-lite just the extention number? (say that in my extention.conf, I have extention 32 to connect to my fxs card (TDM). If I just call 32, it will time out. The work around that I did is to add the user that I want to call to phone book with the extention like [EMAIL PROTECTED], is there any way to do this? Did I miss something in the configurations? AFAIK this is a problem of x-lite. If You enter any numerical value without @-part, the phone interpretes the number as an IP-Adress or so. I remember looking at funny sniffer traces, when doing things like this. I currently have no solution (beside from Your phonebook solution). Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie to asterisk
KH Chow wrote: Dear Sir / Madam, I am a newbie in using Asterisk. I am interested in its SIP. Before I start to use it, I would like to know whether the system can work between two Linux box without any FXO and FXS card and just using microphone which connect to the regular sound card? I am looking into others applications and all of them are using at least one FXS card. Sorry for such beginner problem and please help. Thank you very much for your time. Max Here are 2 apps you could try: KPhone: http://www.wirlab.net/kphone/what.html LinPhone: http://www.linphone.org/?lang=usrubrique=1 If you want to try with H.323 (v4), you can give a try to GnomeMeeting, a great communication software: http://www.gnomemeeting.org/ JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] KPhone working
Hi, If anyone else had a problem I got kphone to work with Asterisk. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 128 kbs satelite link
On Wednesday 17 December 2003 09:48 am, Senad Jordanovic wrote: Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? There's only 500ms lag over satelite. The rest is either misconfiguration or slow connections. (We've been connected to S.W. indies with 500ms.) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This newbie gives up for now - sadly
Though slightly off-topic, I was wondering if anyone would have any ideas to the following regarding our Cisco 7960's. To keep this short - the plan facts: - With phone configured for NAT, works fine with Pulver FWD service from any location (home, various peoples offices etc...) BUT - ... phone does not work in my office. Cannot log on to system. This is with both Real IP AND NATed IP. - Yes I turned off NAT when testing phone with the real IP. Still didn't work - We have two incoming ISP lines in our office. Both have real IP's. No combination works with both lines. - Zultys Zip2 phones however seem to work fine with Real IP's from our office (ZIP doesn't support NAT), on both lines - MSN messenger also works fine For some reasons , our Cisco phones are just cursed when used in our office... I have no explanation for its erratic behaviour at all. Perhaps I should call in a Feng Shui expert? Terence (yes - it's a good looking phone though) As Robert's colleague that owns 7960s I can go on about the superiority of the Cisco phone. The most immediate difference is the look and feel. Everyone that has seen or held my phone says that it is nice. Everyone that picks up a Grandstream phone or looks at one says they are cheap. Grandstream should really consider putting some lead weights in the handset. Hell, the free USB phone from Voiceglo feels better than the Grandstream phone... and that is just the exterior... As soon as they get their problems with SIP functionality and stability sorted, they should spend some time and effort on product design. I understand they are trying to be competitive but people expect a phone to look and feel a certain way. cameron. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] KPhone working
Steve wrote: Hi, If anyone else had a problem I got kphone to work with Asterisk. I have problems with kphone + Asterisk. KPhone does not seem to ACK invites, ie. KPhone --- sends INVITE -- Asterisk KPhone -- sends 101 Trying --- Asterisk KPhone -- sends 202 OK --- Asterisk KPhone --- does not send ACK With other sip phones(linphone, selfmade), Asterisk behaved as if it does not notice ACKs: XPhone --- sends INVITE -- Asterisk XPhone -- sends 101 Trying --- Asterisk XPhone -- sends 202 OK --- Asterisk XPhone --- sends ACK-- Asterisk Asterisk timeouts on OK retransmission as if it has not noticed ACK. Any hints? Or should I send traces? Maciej Kaminski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P without q931?
Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3.3v PCI board - TE410P photo
Hello, Everything I've read says that 3.3v 32bit cards will work in 64 bit slots, and the cards do fit, they just have some extra space left on the slot. MATT--- -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 3:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GUI client for windows for live monitoring/b arge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: January 12, 2004 11:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Riley Sent: Tuesday, 13 January 2004 13:02 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] GUI client for windows for live monitoring/barge I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks, This may be a wacky suggestion, might require more resources, but why don't you try setting up your dialplan so that all calls are in conference s with two members, then you can drop in any time you want... Cheers, Woody Thanks for the idea I'll look at doing that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Linux Distribution
Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3.3v PCI board - TE410P photo
mattf wrote: Hello, Everything I've read says that 3.3v 32bit cards will work in 64 bit slots, and the cards do fit, they just have some extra space left on the slot. MATT--- Yes you are 100% correct.. A 3.3v 32bit card will just have a shorter connector on the bottom that will not extend into the 64bit area.. Unfortunately if you want 3.3v 32bit you have to get a motherboard with 64bit slots becasue I don't know of any motherboard that has 3.3v 32bit.. Also you probably will have to go Xeon.. Later.. -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 3:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
[EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? Hi Mark, I am working on a distro called SAX built to optimize * and routing. It works with RPMs and its HFS is RedHat like. I built all packages by hand and created RPMs packages. It is in beta version by now. More few days and I will release an ISO image. Daniel thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
[EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? thanks mark You better dusck down cos here comes the war about who's distro is better.. :) Use the one you are most comforatable with is the easiest and most logical answer.. IMO thats all that matters since all Linux distros essentially use the same software packages to make up the distro.. What may be cool is to have a Asterisk-Linux specifically constructed and optimised for Asterisk.. and maybe even small enough to run on a CF disk.. and not based on any current distro so there are no fights.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
the one you feel most confortable with. as far as I know, asterisk is developed under RedHat, but really, I run it with RH, debian, slack. Many with suse and so on... so is up to you. matteo. Il mar, 2004-01-13 alle 12:48, [EMAIL PROTECTED] ha scritto: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
cool idea :) Il mar, 2004-01-13 alle 13:10, Daniel Bichara ha scritto: [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? Hi Mark, I am working on a distro called SAX built to optimize * and routing. It works with RPMs and its HFS is RedHat like. I built all packages by hand and created RPMs packages. It is in beta version by now. More few days and I will release an ISO image. Daniel thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone.net net wackiness?
only domain name screwed up. mmh.. my registrar allows me an autorenew for all domain names... pretty useful :) matteo. Il mar, 2004-01-13 alle 09:24, Chris Albertson ha scritto: Looks like they went off the air just after my PayPal payment was processed. I gues we wait a couple days to see if Nufone has gone belly up/bankrupt/gone or if this is just a domain name screw up. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward call with response required to accept
Hi! I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Consider using a queue and agents. Read more on the Wiki. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail issue
Hi! I get to voicemail either way. It just doesn't playback the unavail on the IAX call. Plays back fine on the SIP call. Both calls show up as playing voicemail/company/6711/unavail on the console. Sounds like a codec problem - check which codecs are being used during the IAX connection and look at the disallow/allow statements. Also make sure that you don't have any too much lag that swallows the first bit of playback because the connection hasn't been fully established yet - a look into /var/log/asterisk/messages can help to diagnose that - insert a Wait(2) to avoid this. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3.3v PCI board - TE410P photo
No need to go Xeon, I have one of these: http://www.tyan.com/products/html/thunderk7x.html Dual AMD Athlon MP with one 3.3v 64bit PCI slot MATT--- -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 7:07 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo mattf wrote: Hello, Everything I've read says that 3.3v 32bit cards will work in 64 bit slots, and the cards do fit, they just have some extra space left on the slot. MATT--- Yes you are 100% correct.. A 3.3v 32bit card will just have a shorter connector on the bottom that will not extend into the 64bit area.. Unfortunately if you want 3.3v 32bit you have to get a motherboard with 64bit slots becasue I don't know of any motherboard that has 3.3v 32bit.. Also you probably will have to go Xeon.. Later.. -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 3:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7910 phone
Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok Cisco's site shows SIP drivers for 7960, 7940, 7912, 7905 only. If you want to run 7910 in Skinny mode, that may work. I'll leave that up to the chan_sccp and chan_skinny people. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
I use Fedora FC1. Best is a matter of opinion. Whatever you know is best for you. Michael On Tue, 13 Jan 2004 12:48:09 +0100, [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] The future is here. It's just not evenly distributed yet. - William Gibson ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LCR / Trollphone Rate Engine
Hi! | firstperiod | int(10) unsigned | | | 0 || (Explain?) How long is the first billing interval. The first 60 seconds might be billed at $.04 per minute which then changes... | startcost| int(10) unsigned | | | 0 || (Connection fee?) Sounds reasonable. Be careful: That could also be minimum fee, which is not identical to connection fee. It could also describe the fee for the firstperiod. But most likely this is the connection fee. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe issues?
Hi, Sorry Chris, actually, I cannot help you regarding your problem! But I would like to know how allow an user to change of conferences (go to an other room) !?! Regards, Aresk On Tue, 2004-01-13 at 02:47, Christopher Arnold wrote: Hi all, i have a setup with chatrooms, several MeetMe conferences wich users can change inbetween. 10 users maximum in each room. It seems like when i have more than 40-45 users on the system at the same time asterisk drops abt 20 and continnues buisness as usual. Is there anyone else who have run inte this problem? Any solutions? It would me neat to hear about peoples experiences with MeetMe, how many simultanius users is a practical maximum? On what platform? Also i have nooted an new option d - dynamic add conference, what is the usage for this? /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] KPhone working
On 13-01 12:17, Maciek Kaminski wrote: Steve wrote: Hi, If anyone else had a problem I got kphone to work with Asterisk. I have problems with kphone + Asterisk. KPhone does not seem to ACK invites, ie. KPhone --- sends INVITE -- Asterisk KPhone -- sends 101 Trying --- Asterisk KPhone -- sends 202 OK --- Asterisk KPhone --- does not send ACK Could you, please, send me SIP message dumps of this ? Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
On 13/01/04 11:48, [EMAIL PROTECTED] wrote: which is the best distribution to work with asterisk? They're all just Linux. There is no best. This question is asked so frequently it almost looks like a troll to me. :) I've therefore updated the FAQ on the wiki: - http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Which Linux distribution should I choose for Asterisk? -- There is no best distribution. There are no fundamental differences in functionality or behaviour between Linux distributions like there are between versions of Windows. Pick whichever one you feel most comfortable with. M'kay? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Festival (* dies with no info)
Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the console. Asterisk is built from CVS and has been configured per the Wiki as well, including the test extension (555). I start Festival with the festival_server script, then start Asterisk. (snippet from extensions.conf) exten = 555,1,Answer exten = 555,2,Festival(mary had a little lamb) exten = 555,3,Hangup Here's what Asterisk says with -v, calling from SIP 81001 to 555: Asterisk Ready. -- Executing Answer(SIP/81001-e87b, ) in new stack -- Executing Festival(SIP/81001-e87b, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (from-sip, 555, 2) exited non-zero on 'SIP/81001-e87b' ...at this point Asterisk is dead. No segfault, no error message. # cat /var/log/asterisk/messages Jan 7 15:36:49 WARNING[1074416352]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now # cat /var/log/asterisk/event_log Jan 7 15:36:47 asterisk[5038]: Started Asterisk Event Logger (I capture stderr to asterisk.err) # cat /var/log/asterisk/asterisk.err Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I'm guessing the ouch comes from mpg123 being surprised that Asterisk is gone. Debug info in syslog seems pretty unhelpful if I use -d: Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 4024 (check_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 5098 (handle_request): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 1002 (find_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 3417 (build_route): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 304 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 361 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 363 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 379 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 400 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 410 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 123 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 130 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File chan_sip.c, Line 1081 (sip_hangup): Festival's info is very minimal, but seems to indicate success: # cat festival_server.log Load server start ./festival_server.scm festival port=1314 wrapper Wed Jan 7 15:36:40 EST 2004 : USING DEFAULT CONFIGURATION wrapper Wed Jan 7 15:36:41 EST 2004 : waiting serverWed Jan 7 15:36:41 2004 : Festival server started on port 1314 client(1) Wed Jan 7 15:37:00 2004 : accepted from localhost client(1) Wed Jan 7 15:37:00 2004 : disconnected ...a process listing after the * crash shows a zombie festival, although Festival will happily take new connections: 5024 ?S 0:00 /bin/sh /usr/local/festival/bin/festival_server 5030 ?S 0:00 festival --server ./festival_server.scm 5065 ?Z 0:00 [festival defunct] I can restart Asterisk again, and do this over and over and over. If I use the -g option to generate a core dump, I never see one generated. Any thoughts on what might be happening here? What am I doing wrong? -- Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3.3v PCI board - TE410P photo
mattf wrote: No need to go Xeon, I have one of these: http://www.tyan.com/products/html/thunderk7x.html Dual AMD Athlon MP with one 3.3v 64bit PCI slot MATT--- Athon MP or Xeon IMO are the same thing.. They are just the high end version of either the AMD or Intel proc respectively.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Sync clarification
If you've got spans from different providers...you're in for an adventure. You'll be able to do one of the following (which one is telco and luck dependant): So what you're saying is that the TE410P is not capable of *independently* clocking each of the T1s. Hell even the venerable old AS5248 can handle that. This is going to be fun... Is it possible to accept clock from the telco for one span and *generate* clock on the other three spans (i.e. for internal channel banks and whatnot) ? Will I run into problems there? I don't forsee it but I also didn't forsee the problem being discussed in this thread... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Symbol NetVision Phone
HiList ! I received an unit of the Symbol NetVision Phone and i will test it with asteriskusing H.323 or Skinny , somebody tested thisphone with asterisk and can share experience? Miklos
[Asterisk-Users] Asterisk 0.7.0
Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! Mark p.s. there was no 0.6.0 release. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 0.7.0 Release Mirrors
Here are a list of mirrors for the 0.7.0 tarball. http://66.225.202.82/downloads/asterisk-0.7.0.tar.gz http://parc.styx.org/asterisk/asterisk-0.7.0.tar.gz http://www.bkw.org/asterisk-0.7.0.tar.gz http://www.moctel.com/asterisk/asterisk-0.7.0.tar.gz http://matrix.gs/asterisk-0.7.0.tar.gz http://www.cancunsystems.com/asterisk-0.7.0.tar.gz Thanks, bkw_ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FS/OS Telephony Summit 2004
Hello * world, i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen from the 16th til 20th january. Together with Christian Richter i will be speaking about * on monday. And we will give an * tutorial on tuesday. I will be presenting some ISDN stuff there, including the quadBRI cards. If you will be there too and want to meet, just let me know. :) Details on the summit can be found at: http://www.guug.de/veranstaltungen/telephony-summit-2004/ best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thank You All
I'd be happy to give my docs to the project. I just noticed that it was in progress after I posted but I'd be happy to help. Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Monday, January 12, 2004 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Thank You All On Mon, 2004-01-12 at 09:31, Lane Hoskins wrote: [snip] The only snags we ran into were during basic configuration due to some things that were written about contexts but not clearly explained. As such we are working on a basic guide/manual similar to the 'Getting Started' pages on the wiki for those who want another perspective on installing and configuring this great system. [snip] This will not be a huge project but should be around 20-30 printed pages letting the noob (like us) get up and running smoothly and pointing to the correct places for help. Again, Thanks to the entire community and I hope that our documentation will be of help. Would you mind contributing your writing to the Asterisk Documentation Project at http://www.asteriskdocs.org/? We could certainly use your help in writing good solid Asterisk documentation. If you use IRC, we're usually hanging out in #asterisk-doc on freenode.net. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RFC3389 messages with ATA 186
Thanks to everyone that replied! I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Change audiomode to 0x00140014 The above setting did it - the other info people provided gave me the background on this to understand it more. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Documentation! (WAS: More words for Allison)
On Mon, 2004-01-12 at 19:20, Rich Adamson wrote: That was my thought too... I sent him a few bucks, but then noticed that everyone else seems to be sending him a lot more. Maybe if I offer to write some Asterisk documentation (which I am doing, by the way) people will send me money! hmmm... where should I send $1,000? :-) Actually, we could use help a lot more than we could money right now. (But if you're serious about the $1000, I'm sure we could set up a PayPal account for the Asterisk Documentation team to use for necessary expenses, kind of like some other open source teams do.) I know I've mentioned this on the list a lot in the past few days and people are probably sick of me, so I promise this will be the last plug for a while... Please join us in #asterisk-doc or on the asterisk-doc mailing list or just check out what we've done so far at http://www.asteriskdocs.org/. The sooner we get some solid written documentation done, the better it will be for all of us. (And no, I'm not trying to compete with the Wiki or any of the other great resources out there. I'd just like some good solid documentation in a nice easy format that someday might be published into a book.) So please, if you're interested, come give us a hand... Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3.3v PCI board - TE410P photo
There's a BIG difference in price, depending upon what you consider the equivalent, the Xeon's are about twice as expensive as the Athlon MP's MATT--- -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 8:19 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo mattf wrote: No need to go Xeon, I have one of these: http://www.tyan.com/products/html/thunderk7x.html Dual AMD Athlon MP with one 3.3v 64bit PCI slot MATT--- Athon MP or Xeon IMO are the same thing.. They are just the high end version of either the AMD or Intel proc respectively.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 Sync clarification
On Tuesday, January 13, 2004 7:36 AM, Andrew Kohlsmith [SMTP:[EMAIL PROTECTED] wrote: If you've got spans from different providers...you're in for an adventure. You'll be able to do one of the following (which one is telco and luck dependant): If all providers are referenced back to a stratum 1 clock (which they should be) then all provider spans should have very very very close timing. Close enough that only a few frame slips a year may occur. So, in general spans from different providers should not be a problem. So what you're saying is that the TE410P is not capable of *independently* clocking each of the T1s. Hell even the venerable old AS5248 can handle that. This is going to be fun... That is correct. Is it possible to accept clock from the telco for one span and *generate* clock on the other three spans (i.e. for internal channel banks and whatnot) ? Will I run into problems there? I don't forsee it but I also didn't forsee the problem being discussed in this thread... Yes it is possible to receive clock from one span and provide it for the other three. That is how I am running. Regards, Andrew Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse
I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3.3v PCI board - TE410P photo
mattf wrote: No need to go Xeon, I have one of these: http://www.tyan.com/products/html/thunderk7x.html Dual AMD Athlon MP with one 3.3v 64bit PCI slot MATT--- Athon MP or Xeon IMO are the same thing.. They are just the high end version of either the AMD or Intel proc respectively.. Later.. Ok, so I'm a compaq kind of guy but I can't even remember off the top of my head any of their servers that don't include 64-bit 3.3v slots, even the lower end, older G2, Pentium III based servers. ie. ML350/G2 PIII 1.26ghz includes... 64-bit/33MHz,PCI(5 available) 3.3 Volt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inbound call routing problem
I have come to a stumbling block. We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Lane Hoskins, MCP Network Engineer 540.767.7626 image001.gif
Re: [Asterisk-Users] Voicepulse
same here... with nufone too... i was just getting everyone is busy at the moment message in CLI... it was working fine before.. was it them or was something wrong with my network? will check tomm. cm - Original Message - From: Burak Balasaygun [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 8:21 PM Subject: [Asterisk-Users] Voicepulse I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3.3v PCI board - TE410P photo
Yes, the card works nicely in the 64-bit slots, it just doesn't use all of the pins. Example, the Tyan S2723 works fine. The 3.3v key helps to hold it snugly. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, January 13, 2004 8:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pick up remote call
Hi, I,m trying to pickup remote call using the SIP protocol and *8# from my phone but with no success. I just installed * 0.7.0 and my Phones are connected to one ATA 186 with image 2.16.1. I set in the sip.conf the follow parameter: callgroup=1 pickupgroup=1 for each phone. Someone can help me ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3.3v PCI board - TE410P photo
Yes, they will - I've tried it. 64-bit, 3.3v slots Regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Tuesday, January 13, 2004 11:39 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] 3.3v PCI board - TE410P photo Hello, Everything I've read says that 3.3v 32bit cards will work in 64 bit slots, and the cards do fit, they just have some extra space left on the slot. MATT--- -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 3:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3.3v PCI board - TE410P photo Scott Stingel wrote: Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector layout matches. CHECK YOUR BOARD CAREFULLY, not only the spec's, but a picture of the PCI slots. I've seen some of the boards posted here do not in actuality have compatible slots. Looking at your photo the card is a 3.3v 32 bit card.. Unfortunately it does not liik like anyone has a board with a 3.3v 32bir, they all go straight to the 64bit slots which i guess is understandable.. Here is a page with some photos.. http://hsi.web.cern.ch/HSI/s-link/devices/s32pci64/slottypes.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P without q931?
Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve Well, not quite PRI nor quite what you're describing, but would SS7 be what you're after? Are you asking for a PRI with no D-Channel? Or a group of PRI's that share a single D-Channel? If the former, I'm uncertain. If the latter, that's called NFAS (Non-Facility Associated Switching) and I'd love to see Asterisk support NFAS, for both immediate reasons (same-card and multi-card spanning NFAS groups) and future reasons (my hopes that someone will come up with a channelized DS-3 driver for Zap PRI interfaces, which would almost certainly require NFAS.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Specifying a codec to be used in /etc/sip.conf
Hi all! Is it possible to tell * to allow connecting an incoming (SIP-) call with the G711 codec (a simple fax). I have not found any setting in sip.conf that would refer to this problem. I am using * and the spandsp library to receive faxes from a SIP gateway. Everything works for now except the final transmission of the fax. It seems that the sender and *, the receiver, do not negotiate the correct codec, which must definitely be G711. Any ideas? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 24x7x365 asterisk support available?
Does anyone know of companies or individuals who provide 24x7 asterisk support options? -j -- Jeffrey Paul - [EMAIL PROTECTED] - (877) 748-3467 Senior Network Administrator, Diamond Financial Products An expert is a man who has made all the mistakes which can be made in a very narrow field. -- Niels Bohr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Installation problem
Also, I have an error with make make install under asterisk: /bin/sh: line 1: ./mkdep: Permission denied make: *** [.depend] Error 126 Any idea ? Tnanks, Marin Blu --- C. Maj [EMAIL PROTECTED] wrote: On Mon, 12 Jan 2004, marin blu waxed: I'm trying to install * on Mandrake 9.2/P4, but under asterisk - make clean;make install there is the following error: How about: make then: make install -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttool and errors
John Brown (CV) wrote: It appears that zttool doesn't actually report T1 span errors. If I inject BPV's, crc errors, framing errors, etc into a T1 span, the counters on zttool don't change. It works OK for me with Tormenta 2 and TE410P boards. Both zttool and the /proc/zaptel/x files seem to agree on the error counts too. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inbound call routing problem
Lane Hoskins wrote: I have come to a stumbling block. We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Lane Hoskins, MCP Network Engineer 540.767.7626 I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Sync clarification
If you've got spans from different providers...you're in for an adventure. You'll be able to do one of the following (which one is telco and luck dependant): So what you're saying is that the TE410P is not capable of *independently* clocking each of the T1s. Hell even the venerable old AS5248 can handle that. This is going to be fun... Is it possible to accept clock from the telco for one span and *generate* clock on the other three spans (i.e. for internal channel banks and whatnot) ? Will I run into problems there? I don't forsee it but I also didn't forsee the problem being discussed in this thread... Think we're trying to make this more difficult then what it really is. Every T1 card has a clock, period. The card, regardless of whether it is in a channel bank or a PC, runs at some frequency determined by the engineer that designed the card. A specific card's clock might run at 15.44 mega-units/sec, however the exact frequency at any point in time might be 15.43999 or 15.44001, or some other variation. Letting the clock slide around over time is not a cool thing in high speed digital communications. Therefore, the person implementing the card usually has to choose a source from which to sync his card's clock. There isn't any need to attempt to sync the card's clock from multiple sources simultaneously. The telephone company engineers have had to make the exact same engineering decisions for each central switching office, however since many of these offices have digital facilities from several external companies, they simply coordinate with these other companies as to who is going to be the source (for clock syncing) verses who will simply listen. Those decisions are based on a rather well understood hierarchical arrangement that usually starts with a large carrier and an atomic clock. (The telco will also engineer for a primary and one or more failover secondaries, etc.) Since the digium card has a clock, you simply pick one source to sync from. If you just happen to have multiple T1's coming from different companies, you can only hope/expect those companies have participated in the effort to follow the hierarchical, historically well understood, syncing arrangements. If one of them happens to be a fly-by-night organization that hasn't understood the international sync requirements, your only option is to either encourage them to participate or find a different provider. Period. Once you've chosen a sync source, your card's clock should now be in sync with master atomic clock via layers of this well understood hierarchy. If you connect channel banks to this same card, the digital signals transmitted by your card to the channel bank is going to be derived from your card's in-sync clock. That says your channel banks should then be configured to sync from that card. If you don't do that, then you are breaking the hierarchical structure within your network. If you are large enough to have many asterisk boxes all interconnected via T1's in some sort of full mesh configuration, then as an engineer you have to design your systems in such a way as to pick a clock source to sync with (call it your Master), and design each component in your network to sync from that Master via your own hierarchy. Its not that hard, but it really needs to be done. Just like the telephone company engineers, you should think about what happens if your primary source of sync fails. If you enjoy T1's from multiple external sources, then pick a secondary (backup) for syncing. However you choose to do that is based on your exact network configuration, and not on how the digium card was designed, etc. If your asterisk box interconnects with a traditional pbx that has T1 connections to the pstn, then whoever engineered that pbx had to make the same sync decisions (even though they didn't tell you about it). In this case, your asterisk machine should sync from the traditional pbx. If you have a T1 from your pstn telco terminating on your asterisk, and another T1 going from asterisk to your traditional pbx, then configure asterisk to sync from the telco and the traditional pbx to sync from your asterisk. To complete this rather lengthy topic... what happens if you ignore all of this and just slap a bunch of systems together with no regard to a master sync source? The quality and stability of your network will likely not be as good as what it could be. If your clocks (in each device) happen to be running very very close to what is expected, your network might run just fine. But, if one of the clock's frequency drifts around, it could impact quality via frame slippage and other unwanted events, and if off by a large amount could even be the source of failures. (Your milage will vary directly with the stability of your clocks.) Hope that helps someone Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
On Tue, 13 Jan 2004, Jonathan Moore wrote: LSRB = Loop Start with Reverse Battery I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. Try LSRB - it may work. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up remote call
is just *8 see ya. matteo. Il mar, 2004-01-13 alle 16:03, massimo ha scritto: Hi, I,m trying to pickup remote call using the SIP protocol and *8# from my phone but with no success. I just installed * 0.7.0 and my Phones are connected to one ATA 186 with image 2.16.1. I set in the sip.conf the follow parameter: callgroup=1 pickupgroup=1 for each phone. Someone can help me ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS problem
Hi, Is there a problem with the cvs.digium.com ? I can not download the asterisk repository. Thanks, Marin Blu __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it battery drop disconnect). The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Festival (* dies with no info)
It may not be you, I think the Festival driver is buggy. Specifically, I've found that the the way in which you pass the text to Festival matters. If I use the Festival () suntax then it won't work. If I use the wrong sort of quotation mark instead of ' there are problems. Asterisk will consume vast amounts of processor resources. However, if I specify the command in a way the Festival app likes then all is OK. Try variants like: exten = 555,4,Festival,'mary had a little lamb' Iain --On Tuesday, January 13, 2004 8:11 am -0500 Doug Raum [EMAIL PROTECTED] wrote: Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the console. Asterisk is built from CVS and has been configured per the Wiki as well, including the test extension (555). I start Festival with the festival_server script, then start Asterisk. (snippet from extensions.conf) exten = 555,1,Answer exten = 555,2,Festival(mary had a little lamb) exten = 555,3,Hangup Here's what Asterisk says with -v, calling from SIP 81001 to 555: Asterisk Ready. -- Executing Answer(SIP/81001-e87b, ) in new stack -- Executing Festival(SIP/81001-e87b, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (from-sip, 555, 2) exited non-zero on 'SIP/81001-e87b' ...at this point Asterisk is dead. No segfault, no error message. # cat /var/log/asterisk/messages Jan 7 15:36:49 WARNING[1074416352]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now # cat /var/log/asterisk/event_log Jan 7 15:36:47 asterisk[5038]: Started Asterisk Event Logger (I capture stderr to asterisk.err) # cat /var/log/asterisk/asterisk.err Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I'm guessing the ouch comes from mpg123 being surprised that Asterisk is gone. Debug info in syslog seems pretty unhelpful if I use -d: Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 4024 (check_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 5098 (handle_request): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 1002 (find_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 3417 (build_route): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 304 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 361 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 363 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 379 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 400 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 410 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 123 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 130 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File chan_sip.c, Line 1081 (sip_hangup): Festival's info is very minimal, but seems to indicate success: # cat festival_server.log Load server start ./festival_server.scm festival port=1314 wrapper Wed Jan 7 15:36:40 EST 2004 : USING DEFAULT CONFIGURATION wrapper Wed Jan 7 15:36:41 EST 2004 : waiting serverWed Jan 7 15:36:41 2004 : Festival server started on port 1314 client(1) Wed Jan 7 15:37:00 2004 : accepted from localhost client(1) Wed Jan 7 15:37:00 2004 : disconnected ...a process listing after the * crash shows a zombie festival, although Festival will happily take new connections: 5024 ?S 0:00 /bin/sh /usr/local/festival/bin/festival_server 5030 ?S 0:00 festival --server ./festival_server.scm 5065 ?Z 0:00 [festival defunct] I can restart Asterisk again, and do this over and over and over. If I use the -g option to generate a core dump, I never see one generated. Any thoughts on what might be happening here? What am I doing wrong? -- Doug
Re: [Asterisk-Users] Best Linux Distribution
Il personally use Mandrake 9.2 and it works perfectly. On Debian, we've never got the FritzCard USB2 ISDN card working, but nothing to do directly with Asterisk. The only performance issue I've got was while running X (many comments around this issue). JC [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound call routing problem
On Tue, 2004-01-13 at 07:52, Lane Hoskins wrote: We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. You need to understand more about contexts. If you put lines 1 and 2 in a context (let's call it [everyone]) and each of the other lines in it's own context (let's say [line3], [line4], etc.), then you can control what happens in each context. If you haven't figured out where to assign a context to each line, it's in your /etc/asterisk/zapata.conf file. After setting those in zapata.conf, your (very simplified) extensions.conf file will look something like this: [everyone] ; ring everyone exten=s,1,Answer() exten=s,2,Dial(SIP/JohnSIP/MarySIP/FredSIP/Bob) [line3] exten=s,1,Answer() exten=s,2,Dial(SIP/John,20,r) exten=s,3,Dial(John's cellphone goes here,10,r) exten=s,4,VoiceMailMain(John's mailbox) exten=s,5,Hangup() exten=s,103,Dial(John's cellphone goes here,10,r) exten=s,104,VoiceMailMain(John's mailbox) exten=s,105,Hangup() exten=s,204,VoiceMailMain(John's mailbox) exten=s,205,Hangup() [line4] exten=s,1,Answer() exten=s,2,Dial(SIP/Mary,20,r) exten=s,3,Dial(Mary's cellphone goes here,10,r) exten=s,4,VoiceMailMain(Mary's mailbox) exten=s,5,Hangup() exten=s,103,Dial(Mary's cellphone goes here,10,r) exten=s,104,VoiceMailMain(Mary's mailbox) exten=s,105,Hangup() exten=s,204,VoiceMailMain(Mary's mailbox) exten=s,205,Hangup() ... etc., etc. ... Hope that gets you started... While this should work, I take no responsibility for typos and or stupid mistakes I may have made while writing this in a hurry... Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
On Tuesday 13 January 2004 03:42, Jason Penton wrote: I have just a quick question regarding app_txfax for Asterisk. When I send a fax from asterisk to a traditional fax machine connected to asterisk via the digium analog card everything works perfectly. However the same fax machine on the public telephoine network results in errors (looks like some sort of training error). My asterisk box is connected to the pstn using an ISDN card. I don't mind trying to fix this myself but I am puzzled by the different behavior experienced when the fax machine is on the digium card and when it is connected to our public carrier, and therefore have no idea where to start. Would someone (Steve Underwood ;-) )mind at least putting me on the right track so I can address this issue? If you read the caveat for the TxFax and RxFax applications, you'll note that they will only work with Zaptel devices. Your ISDN card is not a Zaptel device. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
On Tuesday 13 January 2004 02:27, WipeOut wrote: Tilghman Lesher wrote: On Tuesday 13 January 2004 00:10, Mark Spencer wrote: Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! There is one bug so far and it's critical. It breaks includes and the GotoIfTime application. I'll own up to writing the broken code. The fix is very simple, though (attached). Why not quickly patch the source an release 0.7.1 if the bug is critical? We're planning to do that, but there's going to be a lag between planning a release and getting a release out. For people who want to use 0.7.0 right away, it's better to release news of the discovery of the bug right away, not wait for a new release. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 24x7x365 asterisk support available?
On Tue, 13 Jan 2004, Jeffrey Paul wrote: Does anyone know of companies or individuals who provide 24x7 asterisk support options? My company does, http://www.internetsolver.com/ dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Specifying a codec to be used in /etc/sip.conf
Follow-up question, what does * use for fax? T38 or passthrough? - Original Message - From: Peter Bittner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:12 AM Subject: [Asterisk-Users] Specifying a codec to be used in /etc/sip.conf Hi all! Is it possible to tell * to allow connecting an incoming (SIP-) call with the G711 codec (a simple fax). I have not found any setting in sip.conf that would refer to this problem. I am using * and the spandsp library to receive faxes from a SIP gateway. Everything works for now except the final transmission of the fax. It seems that the sender and *, the receiver, do not negotiate the correct codec, which must definitely be G711. Any ideas? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P without q931?
On Tue, 13 Jan 2004, John Todd wrote: does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Well, not quite PRI nor quite what you're describing, but would SS7 be what you're after? yes basically! i'm thinking of using a dedicated ss7 signalling box but dropping the voice into something that can do the conversion to voip, play things to the caller etc Are you asking for a PRI with no D-Channel? Or a group of PRI's that share a single D-Channel? If the former, I'm uncertain. If the latter, that's called No, I'm just being (rather badly) loose with my terms.. I'm not talking ISDN I'm talking purely of pushing PCM down a B channel and connecting the channel to whatever.. [ss7 is what i need for my network interface but internally i want to use our own customised systems, asterisk will do nicely for some of that..] Steve NFAS (Non-Facility Associated Switching) and I'd love to see Asterisk support NFAS, for both immediate reasons (same-card and multi-card spanning NFAS groups) and future reasons (my hopes that someone will come up with a channelized DS-3 driver for Zap PRI interfaces, which would almost certainly require NFAS.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and AGI crash...
Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten = 125,1,Ringing exten = 125,2,Wait(3) exten = 125,3,Answer exten = 125,4,Wait(2) exten = 125,5,AGI(agi-sayani.agi) exten = 125,6,Hangup I can crash the asterisk server by hanging up during the call, if I leave the call to complete and let * hang up then everything seems fine. Asterisk does not crash if I am running from the console, only if asterisk has been started in the background (it does still crash if I am attached via asterisk -r at the time the call is hung up). Using the agi test script (on extension 126, same config as above) I get the following... *CLI -- Executing Ringing(SIP/-08135e80, ) in new stack -- Executing Wait(SIP/-08135e80, 3) in new stack -- Executing Answer(SIP/-08135e80, ) in new stack -- Executing Wait(SIP/-08135e80, 2) in new stack -- Executing AGI(SIP/-08135e80, agi-test2.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test2.agi AGI Environment Dump: -- accountcode = -- callerid = SNOM Phone 1543 8552 -- channel = SIP/-08135e80 -- context = sip-gw -- dnid = unknown -- enhanced = 0.0 -- extension = 126 -- language = en -- priority = 5 -- rdnis = unknown -- request = agi-test2.agi -- type = SIP -- uniqueid = 1074011198.0 1. Testing 'sendfile'...PASS (0) 2. Testing 'sendtext'...PASS (0) 3. Testing 'sendimage'...PASS (0) -- Playing 'digits/1' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/90' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/million' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/30' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/thousand' (language 'en') Jan 13 16:26:50 WARNING[1116941120]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) == Spawn extension (sip-gw, 126, 5) exited non-zero on 'SIP/-08135e80' -- Executing Hangup(SIP/-08135e80, ) in new stack == Spawn extension (sip-gw, h, 1) exited non-zero on 'SIP/-08135e80' PASS (-1) 5. Testing 'waitdtmf'...FAIL (unexpected result '') 6. Testing 'record'...FAIL (unexpected result '') 6a. Testing 'record' playback...FAIL (unexpected result '') == Complete == 7 tests completed, 4 passed, 3 failed == The test seems to stop half way through. I am not entirely sure that these two issues are actually related though as I don't see any of the warning from chan_sip if I hang up during a call to the say-ani script. I don't seem to be getting a core dump, are there any known issues with AGI at the moment? Voicemail, SayUnixTime and everything else is working fine. -- Tristan 'Minty' Colgate [EMAIL PROTECTED] | ICQ #154577755 --- I don't mean to sound bitter, cold, or cruel, but I am, so that's how it comes out - Bill Hicks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI. used beyond own phone network?
- Original Message - From: C. Maj [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 12, 2004 10:14 PM Subject: Re: [Asterisk-Users] ADSI. used beyond own phone network? What kind of security implications would this have? Probably the same as using DTMF when you call the bank to check on your credit cards. As long as everybody can be trusted at the telco, you're probably fine. I was referring to security of your systems after having downloaded unknown scripts from various parties. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Multiple Products H.323 Protocol Denial of Service Vulnerabilities
FYI for those that might have an interest... TITLE: Cisco Multiple Products H.323 Protocol Denial of Service Vulnerabilities SECUNIA ADVISORY ID: SA10610 VERIFY ADVISORY: http://www.secunia.com/advisories/10610/ CRITICAL: Moderately critical IMPACT: DoS WHERE: From remote OPERATING SYSTEM: Cisco ATA 180 Series Analog Telephone Adaptors Cisco BTS 10200 Softswitch Cisco IOS 11.x Cisco IOS 12.x Cisco IOS R11.x Cisco IOS R12.x SOFTWARE: Cisco CallManager 3.x Cisco Conference Connection (CCC) 1.x Cisco Internet Service Node (ISN) 2.x Cisco IP Phone 7900 Series DESCRIPTION: Multiple Cisco products contain vulnerabilities in the H.323 protocol implementation, which can be exploited by malicious people to cause a DoS (Denial of Service). The vulnerabilities are caused due to various errors in the processing of H.225.0 and Q.931 messages over TCP, which can be exploited by sending specially crafted messages to an affected system (default port 1720/tcp). Successful exploitation may crash or reboot vulnerable devices and applications or cause them to consume 100% CPU resources. The vulnerabilities affect the following products with H.323 support: * Cisco IOS 11.3T and later versions * Cisco CallManager versions 3.0 through 3.3 * Cisco Conference Connection (CCC) * Cisco Internet Service Node (ISN) * Cisco BTS 10200 Softswitch * Cisco 7905 IP Phone H.323 Software Version 1.00 * Cisco ATA 18x series products running H.323/SIP loads with versions earlier than 2.16.1 SOLUTION: See patch matrices and workarounds in original advisory: http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml#software http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml#workarounds PROVIDED AND/OR DISCOVERED BY: NISCC ORIGINAL ADVISORY: Cisco: http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml NISCC: http://www.uniras.gov.uk/vuls/2004/006489/h323.htm -- About: This Advisory was delivered by Secunia as a free service to help everybody keeping their systems up to date against the latest vulnerabilities. Subscribe: http://www.secunia.com/secunia_security_advisories/ Definitions: (Criticality, Where etc.) http://www.secunia.com/about_secunia_advisories/ Please Note: Secunia recommends that you verify all advisories you receive by clicking the link. Secunia NEVER sends attached files with advisories. Secunia does not advise people to install third party patches, only use those supplied by the vendor. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Again: 7920 Cisco IP Phone Skinny SIP
hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: -- Phone auth's with AP Phone gets IP from DHCP TFTP Server Phone loads OS7920.TXT Phone loads SEPmacaddr.CNF.XML Phone loads xmlDefault.conf.xml Phone registeres to Asterisk Phone gets registered Phone gets Info/Dial/Stuff from Asterisk Phone gets Line Info SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY SoftKeySetReqMessage SKINNY SoftKeySetResMessage SKINNY OffHookMessage SKINNY SetSpeakerModeMessage SKINNY OnHookMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage But if you look at the Support of the 7920 in Callmanager Express, you get a file named cmterm_7920.3.3-01-02-021.bin so i was investigating further. so i wrote cmterm_7920.3.3-01-02-021 in OS7920.TXT and suddenly the Cisco 7920 shows Upgrading Firmware :-) Unfortunately for some reason it did not accept the firmware, but it still tries to load it. Some additional info: - The 7920 is requesting cmterm_7920.3.3-01-02-021^J.bin (so with an Ctrl-J in it), so you have to rename the file. I also got the information from documents that the 7920 is running in 7960 emulation mode, so draw your own conclusions in regards of SIP possiblity :-) I tried to use some 7960 images, but did not succeed :-( Would appreciate some help in this issue :-) --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound call routing problem
On Tue, 13 Jan 2004, Lane Hoskins waxed: We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Here's Rich Adamson's A WORKING EXAMPLE from September: http://lists.digium.com/pipermail/asterisk-users/2003-September/020944.html I see SIP and Voicemail in there, but I haven't tried it myself. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Sync clarification
If you've got spans from different providers...you're in for an adventure. You'll be able to do one of the following (which one is telco and luck dependant): So what you're saying is that the TE410P is not capable of *independently* clocking each of the T1s. Hell even the venerable old AS5248 can handle that. This is going to be fun... Dont think they do.. on the controllers you specify clock source primary/secondary and the box will sync to only one clock. This is true in all telco systems afaik.. taking lines from another telco which is on a different clock source isnt necessarily a big problem but you should expect to see the odd slips on the line where the clocking is slightly mismatched.. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agents and call queueing
Hello, I have been playing around with call queuing very cool. So at the same time I also tried to implement the agent via the agent call back routine. This is causing problems, in the queue.conf if I have a member as Member = Sip/nick It works But if I set up an agent, login using AgentCallBackLogin, the login works, and when a call is entered into the queue then the phone rings but as soon as the sip phone picks up the call then the call is dropped and the call is returned to the queue (whilst listening to on hold music!). Help Thanks Nick
Re: [Asterisk-Users] Best Linux Distribution
I have to agree with the below but only if it is an answer to the limited question of Which is best to use for my Astrisk server. For a server you are using such a small percentage of the Linux distribution that they are effectivly all the same. A server will not make us of any of the graphical interface or Desk top software. Most * servers run with no keyboard or CRT plugged in. BUT, If you are running an Asterisk server you will likley also have a Linux box for development, testing and general e-mail and web serfing. For this purpose it does matter, a little. They all will do the job but differ in terms of the details of exactly what software is included and how the menu system on the desk top is set up. Still none is better but they are differntent enough that people can have strong prefference. The differences between distributions are minor. I doubt an inexperianced user could tell this Solaris 9 box I'm writing this on from a Linux system. Both run gnome and look the same on the surface. But Linux and Solaris are far more different then any two Linuxes. That said, pick a desktop system you like. You can get a free download of any of them or low priced CDs at cheapbytes.com and try them out. Then use the same distribution for your server. They're all just Linux. There is no best. This question is asked so frequently it almost looks like a troll to me. :) I've therefore updated the FAQ on the wiki: - http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Which Linux distribution should I choose for Asterisk? -- There is no best distribution. There are no fundamental differences in functionality or behaviour between Linux distributions like there are between versions of Windows. Pick whichever one you feel most comfortable with. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse
I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Joel Maslak [EMAIL PROTECTED]: On Tue, 13 Jan 2004, Jonathan Moore wrote: LSRB = Loop Start with Reverse Battery I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. Try LSRB - it may work. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New software SIP phone released today
http://shtoom.sourceforge.net/ I haven't tried it yet, but it looks promising. Written in Python. Supposedly works on Linux/FreeBSD, Windows, MacOS X. Written specifically with Asterisk as a server testbed, I believe. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inbound call routing problem
Thanks David, That is exactly what we had to do. We got some help from Digium as well and have it taken care of. Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] inbound call routing problem Lane Hoskins wrote: I have come to a stumbling block. We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Lane Hoskins, MCP Network Engineer 540.767.7626 I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
I think the exchange below shows us that before 0.8.0 comes
out, maybe there should be a 0.8.0-beta then after no problems are
reported in a few week period a 0.8.0-release candidate and
ten 0.8.0 itself.
It's hard to call a realease stable until a number of people outside
the developer's lab have used it for a while.
The other idea is that everyone just knows that x.y.0 == beta
and they all wait for a .1 or .2 realease.
--- WipeOut [EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all
the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and
the
GotoIfTime application. I'll own up to writing the broken code.
The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c11 Jan 2004 09:19:16 - 1.92
+++ pbx.c13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
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Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
KG6OMK
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[Asterisk-Users] E100P works with PCI 3.3V and 5V?
Hi, I just bought the E100P from digium. It has both keys: 3.3V and 5V, so it would fit both, in a 5V-PCI slot and in a 3.3V PCI slot. Is it true, that I can plug it without destroying it in an ordenary 5V PCI slot? Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?
In our last exciting episode, Tilghman Lesher ([EMAIL PROTECTED]) said: I want you to look at the headers of my reply and note that I'm running my mail client on FreeBSD. Now my advice: run your Asterisk server on Linux. First, a disclaimer: this is not mean to be flame-bait nor is it an attempt at trolling. Why this attitude? There are plenty of us out there who do not wish to bring a Linux server into our enterprise for a variety of reasons (lack of familiarity, desire to retain homogeneous environment, etc) that would love to be able to use Asterisk under FreeBSD. I've browsed the archives and perceived what appears to be a slightly hostile attitude towards those who ask about Asterisk support of other free operating systems even without using Digium hardware. Is this Linux-specific bias intentional or accidental? -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. pgp0.pgp Description: PGP signature
RE: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
If you don't have a voltmeter to look at this, try just listening on the line (using an analog telephone) when the far end hangs up. You should hear a distinct click-click on the line a second or two after they hang up. If you hear this, it's likely you are getting the required disconnect supervision from the telco. Note that many (most?) smaller private PBX's do not drop loop current on an analog line when the far end disconnects - but central office class switches usually do. It's not very scientific, but once you've heard one you can recognise it. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sharp Sent: Tuesday, January 13, 2004 3:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600? Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it battery drop disconnect). The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up remote call
is just *8 I've tried but it does not pick up the call and don't show nothing in the consolle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Sync clarification
To complete this rather lengthy topic... what happens if you ignore all of this and just slap a bunch of systems together with no regard to a master sync source? The quality and stability of your network will likely not be as good as what it could be. If your clocks (in each device) happen to be running very very close to what is expected, your network might run just fine. But, if one of the clock's frequency drifts around, it could impact quality via frame slippage and other unwanted events, and if off by a large amount could even be the source of failures. (Your milage will vary directly with the stability of your clocks.) What are the practical effects with in-correct clock sync -like to you hear odd buzzing, or dropped voice or gaps of audio ?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
