Re: [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else
On Wed, Nov 01, 2006 at 05:47:35PM -0500, Zeeshan Zakaria wrote: Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. Which Tormenta card exactly? I believe tha there are several companies that produce Tormenta2-based cards. I know that at least some of them have some modifications to the tor2 driver, but those modifications never made it into the main tree (and sometimes not even availble for download). That said, I have no experince with either of those cards. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Call Statistics
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... of course you can always use http://cacti.net/download_cacti.php Hi Moisies! I heard that Munin can (or they are working on that) log how many simultaneous call on each interface Asterisk has. Can Cacti do the same? I have tried Cacti once and I liked it weary much. It's easy to configure and has nice interface. I definitely need to install it again! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Call Statistics
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... http://www.areski.net/asterisk-stat-v2/about.php Hi Doug! I don't recommend anybody using Asterisk stat. Last version is V2.0.1 (07 March 2005). It's obsolete. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Thomson ST2030 and Asterisk BLF
Hi everybody.
I know there have been some posts in the past about this subject.
However it seems I cannot get the Thomson ST2030 phone
to work with BLF and call pickup.
Firmware on the phones is 1.5t3
I've applied then patch to chan_sip.c which adds the else condition
} else if (strstr(p-useragent, THOMSON)) {
p-subscribed = DIALOG_INFO_XML;
somewhere in the handle_request_subscribe() function.
The hints are properly configured as well as the subscribecontext in
sip.conf/extensions.conf
In fact, the busy lamp is working (I can see
busy lines and ringing lines on the phone),
however call pickup is not.
When a line key is flashing (i.e. the associated sip phone is ringing),
if I press that key, the phone sends a SUBSCRIBE sip message
to asterisk.
I don't understand exatctly what the phone is expecting back
from asterisk or how asterisk handles the SUBSCRIBE message,
however the call is *not* picked up, and the status line key
gets fast-blinking, and remains in that status,
being unusable, until I reboot the phone.
Any hint?
Thanks.
Alberto.
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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.
Disabling Hyperthreading helped us to work around one way audio calls.Up to 1 call out of 3, were touched by this trouble.When we switched hyperthreading off, we never missed a single call anymore.Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAPtel channel dance
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a TDM[124]xxP, all channels
change their numbers if you do a /etc/initd/zaptel restart. This is
slightly confusing. (I'd file a bug if there were a bug tracking system
that allowed users to submit bugs).
Yours, Florian
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Re: [asterisk-users] IAX problem
On 2 Nov 2006, at 02:38, Itamar Lavender wrote: Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register = username:[EMAIL PROTECTED] What does the rest of iax.conf look like ? Auth is a 2 way thing - you have sucessfully registered with them, but when they send you a 'new' your box fails to authenticate them as it can't find a matching user/friend entry in iax.conf. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
On 1 Nov 2006, at 20:28, Zeeshan Zakaria wrote: I think I will agree with folks here, it must be something else on the network, not the phones themselves. I am not going to replace all of the phones, its too expensive, but for trial, want to try something better. PoE is also important to me at this point. I am thinking of trying Linksys 942. I was thinking of Polycom, but there its LCD is not backlit. I keep all LCDs backlit so that is important for me. As for good Aastra phones, there in no external power adapter. Snoms are expensive. Take a look at elmeg - they make last years snoms under license but are quite a bit cheaper - I'm not sure if they have a PoE version. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else
On 1 Nov 2006, at 22:47, Zeeshan Zakaria wrote: Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with 512MB RAM. Serial ports, parallel port and other unnecessary things on the motherboard are disabled. People in the office don't talk much with each other over the extensions, so can't say the performance of the system over the local network, but incoming and outgoing calls start giving trouble few times a day and people do complain about it. I was thinking if it was because of the linux kernel, which is 2.4.21-32.EL on CensOS 3, or because of the Tormenta PRI Interface card. There is no major use of the Internet in the office. Server is strong enough to handle calls. And there are never more than 3,4 calls at a time. In fact breaking of sound can happen when only one person is in the office and only one phone is being used. Please help before I start buying new stuff to replace some of the existing stuff, just to find out that it didn't help anyways. Thanks Sounds like a clock slip problem on your PRI interface - what is in your zaptel.conf ? Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto dial out and auto answer
Hello list, We have 2 asterisk servers (without firewall and NAT), and We want to do : From the first server, we have a .call file which dial out to the second server. The second server automatically answers and Play a music during X seconds, then it hangs up. Is it possible? Thanks you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
On Thu, Nov 02, 2006 at 10:10:01AM +0100, Florian Hars wrote:
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a TDM[124]xxP, all channels
change their numbers if you do a /etc/initd/zaptel restart. This is
slightly confusing.
The order of the channels is the order in which the spans register to
Zaptel, which is basically the order in which the modules load.
On Debian, load the modules through /etc/modules . Otherwise they will
be loaded through hotplug/udev in an unpredictable order (by the order
of PCI slots) which may or may not be the order that you like.
Gentoo has an equivalent file, whose name I forgot.
Redhats seem to lack such a mechanism, and I'm not sure whther or not
those cards do get hotplugged/coldplugged. Thus the tsrange need to load
them in the zaptel startup script.
Anyway, the order in which you happened to load them right now is not
guaranteed to be the order in which you load them next time unless you
explicitly
(I'd file a bug if there were a bug tracking system
that allowed users to submit bugs).
Users are surely allowed. Just register.
Also, bug reports to xpp/genzaptelconf are welcomed. It should be able
to write such module loading lists that should provide predictable order
in both Debian and Redhats.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] echo with spa-3000
On Thu, 2006-11-02 at 16:08 +1100, James Harper wrote: More an echo algorithm question than a purely asterisk one... Well, being the other side of the world my solution may not work for you. I had echo on my SPA3000, I was sure I'd selected the correct impedance for here in France, then one day I saw a setting Global in the dropdown for the FXO since then no one has complained about echo on any of the SPA3000 units I've got installed. Give it a try it might or might not help. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extending a call limited by L in Dial app
Hi, If I use L(x[:y][:z]) in Dial app the call is limited to x milliseconds, Is it possible for the callee to extend the call past x milliseconds? raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.
Hi Matthew, Matthew Mackes (Webmail) schrieb: Zulty WIP 2- THESE PHONES ARE AWESOME!!! AWESOME!!! WiFi SIP phones- is it possible to provide a phonebook to this phones (via LDAP, TFTP, XML-file or anything else)? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Example Polycom function key config
FYI - Polycom have confirmed to me that you can only send one digit via the programmable feature keys. Idiots. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Heckford Sent: 01 November 2006 09:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Example Polycom function key config Hi Jamie - Hi Noah, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example Here's the keys line that I use for one of my clients: keys key.scrolling.timeout=1 key.IP_500.37.function.prim=DialpadPound key.IP_500.31.function.prim=DialpadStar key.IP_600.37.function.prim=DialpadPound key.IP_600.30.function.prim=DialpadStar/ Thanks for that, I have something similar but what I can't work out is how to send multiple digits. For example 2x 'DialpadPound'. I have tried putting it in twice etc. to no avail. Anyone know how to get this to work? I'm trying to get our transfer key (##) programmed to one of the function keys basically. Thanks, Jamie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else
I don't know wh is the manufacturer of these cards. Here is my zapata.conf ;; Zapata telephony interface;; Configuration file [trunkgroups] [channels]switchtype=nationalcontext=from-pstnsignalling=pri_cpegroup=1language=en;rxwink=300; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines ;;usedistinctiveringdetection=yes usecallerid=yeshidecallerid=nocallwaiting=yescallerid=xyz 1234567890callerid=asreceivedusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yes callreturn=yesechocancel=yesechocancelwhenbridged=noechotraining=800rxgain=0.0txgain=0.0group=0callgroup=1pickupgroup=1immediate=nofacilityenable=yes ;faxdetect=bothfaxdetect=incoming;faxdetect=outgoing;faxdetect=no ;channel = 1-23,25-47,49-71,73-95channel = 1-23,25-47 ;Include genzaptelconf configs#include zapata-auto.conf And here is zaptel.conf # Span 1: Tor2/0/1 Tormenta 2 (PCI) Quad T1 Card 0 Span 1 B8ZS/ESF BLUE RED # ??: 1 Tor2/0/1/1 Clear# ??: 2 Tor2/0/1/2 Clear# ??: 3 Tor2/0/1/3 Clear# ??: 4 Tor2/0/1/4 Clear# ??: 5 Tor2/0/1/5 Clear # ??: 6 Tor2/0/1/6 Clear# ??: 7 Tor2/0/1/7 Clear# ??: 8 Tor2/0/1/8 Clear# ??: 9 Tor2/0/1/9 Clear# ??: 10 Tor2/0/1/10 Clear# ??: 11 Tor2/0/1/11 Clear# ??: 12 Tor2/0/1/12 Clear# ??: 13 Tor2/0/1/13 Clear # ??: 14 Tor2/0/1/14 Clear# ??: 15 Tor2/0/1/15 Clear# ??: 16 Tor2/0/1/16 Clear# ??: 17 Tor2/0/1/17 Clear# ??: 18 Tor2/0/1/18 Clear# ??: 19 Tor2/0/1/19 Clear# ??: 20 Tor2/0/1/20 Clear# ??: 21 Tor2/0/1/21 Clear # ??: 22 Tor2/0/1/22 Clear# ??: 23 Tor2/0/1/23 Clear# ??: 24 Tor2/0/1/24 HDLCFCS # Span 2: Tor2/0/2 Tormenta 2 (PCI) Quad T1 Card 0 Span 2 B8ZS/ESF ClockSource # ??: 25 Tor2/0/2/1 Clear# ??: 26 Tor2/0/2/2 Clear# ??: 27 Tor2/0/2/3 Clear# ??: 28 Tor2/0/2/4 Clear# ??: 29 Tor2/0/2/5 Clear # ??: 30 Tor2/0/2/6 Clear# ??: 31 Tor2/0/2/7 Clear# ??: 32 Tor2/0/2/8 Clear# ??: 33 Tor2/0/2/9 Clear# ??: 34 Tor2/0/2/10 Clear# ??: 35 Tor2/0/2/11 Clear# ??: 36 Tor2/0/2/12 Clear# ??: 37 Tor2/0/2/13 Clear # ??: 38 Tor2/0/2/14 Clear# ??: 39 Tor2/0/2/15 Clear# ??: 40 Tor2/0/2/16 Clear# ??: 41 Tor2/0/2/17 Clear# ??: 42 Tor2/0/2/18 Clear# ??: 43 Tor2/0/2/19 Clear# ??: 44 Tor2/0/2/20 Clear# ??: 45 Tor2/0/2/21 Clear # ??: 46 Tor2/0/2/22 Clear# ??: 47 Tor2/0/2/23 Clear# ??: 48 Tor2/0/2/24 HDLCFCS # Span 3: Tor2/0/3 Tormenta 2 (PCI) Quad T1 Card 0 Span 3 B8ZS/ESF BLUE RED # ??: 49 Tor2/0/3/1 Clear# ??: 50 Tor2/0/3/2 Clear# ??: 51 Tor2/0/3/3 Clear# ??: 52 Tor2/0/3/4 Clear# ??: 53 Tor2/0/3/5 Clear # ??: 54 Tor2/0/3/6 Clear# ??: 55 Tor2/0/3/7 Clear# ??: 56 Tor2/0/3/8 Clear# ??: 57 Tor2/0/3/9 Clear# ??: 58 Tor2/0/3/10 Clear# ??: 59 Tor2/0/3/11 Clear# ??: 60 Tor2/0/3/12 Clear# ??: 61 Tor2/0/3/13 Clear # ??: 62 Tor2/0/3/14 Clear# ??: 63 Tor2/0/3/15 Clear# ??: 64 Tor2/0/3/16 Clear# ??: 65 Tor2/0/3/17 Clear# ??: 66 Tor2/0/3/18 Clear# ??: 67 Tor2/0/3/19 Clear# ??: 68 Tor2/0/3/20 Clear# ??: 69 Tor2/0/3/21 Clear # ??: 70 Tor2/0/3/22 Clear# ??: 71 Tor2/0/3/23 Clear# ??: 72 Tor2/0/3/24 HDLCFCS # Span 4: Tor2/0/4 Tormenta 2 (PCI) Quad T1 Card 0 Span 4 B8ZS/ESF BLUE RED # ??: 73 Tor2/0/4/1 Clear# ??: 74 Tor2/0/4/2 Clear# ??: 75 Tor2/0/4/3 Clear# ??: 76 Tor2/0/4/4 Clear# ??: 77 Tor2/0/4/5 Clear # ??: 78 Tor2/0/4/6 Clear# ??: 79 Tor2/0/4/7 Clear# ??: 80 Tor2/0/4/8 Clear# ??: 81 Tor2/0/4/9 Clear# ??: 82 Tor2/0/4/10 Clear# ??: 83 Tor2/0/4/11 Clear# ??: 84 Tor2/0/4/12 Clear# ??: 85 Tor2/0/4/13 Clear # ??: 86 Tor2/0/4/14 Clear# ??: 87 Tor2/0/4/15 Clear# ??: 88 Tor2/0/4/16 Clear# ??: 89 Tor2/0/4/17 Clear# ??: 90 Tor2/0/4/18 Clear# ??: 91 Tor2/0/4/19 Clear# ??: 92 Tor2/0/4/20 Clear# ??: 93 Tor2/0/4/21 Clear # ??: 94 Tor2/0/4/22 Clear# ??: 95 Tor2/0/4/23 Clear# ??: 96 Tor2/0/4/24 HDLCFCS # Global dataspan=1,1,0,esf,b8zsspan=2,2,0,esf,b8zsspan=3,3,0,esf,b8zsspan=4,4,0,esf,b8zs bchan=1-23,25-47,49-71,73-95dchan=24,48,72,96 loadzone= usdefaultzone= us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] echo with spa-3000
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 2 November 2006 20:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] echo with spa-3000 On Thu, 2006-11-02 at 16:08 +1100, James Harper wrote: More an echo algorithm question than a purely asterisk one... Well, being the other side of the world my solution may not work for you. I had echo on my SPA3000, I was sure I'd selected the correct impedance for here in France, then one day I saw a setting Global in the dropdown for the FXO since then no one has complained about echo on any of the SPA3000 units I've got installed. Give it a try it might or might not help. Just tried that but no change. I think I've tried all of the combinations. I think the signal reflection is happening somewhere beyond where the impedance of my phone comes into affect. Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sound-files not playing?
Hi all! In my extensions I have the following: exten = 999,1,Answer() exten = 999,2,PlayBack(beeperr) In /var/lib/asterisk/sounds/ I have both beeperr.gsm beeperr.ulaw, both with '-rw-r--r--' permissions. when I dial extension 999 I get: -- Executing Answer(SIP/asterisk.domain.com-081477a0, ) in new stack -- Executing Playback(SIP/asterisk.domain.com-081477a0, beeperr) in new stack Nov 2 10:57:11 WARNING[17300]: file.c:512 ast_openstream_full: File beeperr does not exist in any format Nov 2 10:57:11 WARNING[17300]: file.c:824 ast_streamfile: Unable to open beeperr (format ulaw): Permission denied Nov 2 10:57:11 WARNING[17300]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/asterisk.domain.com-081477a0 for beeperr == Auto fallthrough, channel 'SIP/asterisk.domain.com-081477a0' status is 'UNKNOWN' (the name of the box is here asterisk.domain.com ) What am I doing wrong? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wait for an extension and dial. Why does this not work?
From my extensions.conf:
exten = 888,1,Answer()
exten = 888,n,WaitExten(20|m)
exten = 888,n,Dial(SIP/[EMAIL PROTECTED],60,tr)
This should:
* answer
* wait 20 seconds for an extension with music on the background
* pass the call to that extension on ${SERADDRESS}
What am I doing wrong here? I don't even get the background music while
WaitExten is active. I doubt that it is active anyway, since I get
disconnected before the 20 seconds have passed... :-/
Greetings,
Evert
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[asterisk-users] Macro variables and redirects
Hi, I have a dialplan that works like this: ; Arg 1 is the phone, Arg2 is the timeout (optional), Arg 3 is the voicemailbox(optional) exten = 20,1,Macro(dialexten,SIP/1234,15,1234) exten = 21,1,Macro(dialexten,SIP/1235,15) ; Arg 1 is phones, Arg 2 is timeout, Arg 3 is voicemail exten = 30,1,Macro(huntgroup,SIP/1234SIP/1235,25,1234) If I call extension 30, answer it, then redirect to extension 21 via an attended transfer, and no one answers the voicemail will time out to mailbox 1234, when it should not (the macro makes it play busy instead). I can just set a value and test for that (like off), but should the macro arguments for the huntgroup macro be remembered in the dialexten macro like this? The redirect calls the second macro correctly, but if you NoOp the arguments in the macro, you can see that they are inherited from the previous macro) I am using 1.2.10. With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] blindtransfer and initiator hangup
Good afternoon. The asterisk has two kinds transfer, attended and blind, me interests as to set for blindtransfer performance what or commands on exten = h for the one who this transfer initiated. I.e. now in the console it is visible Hangup the initiator but as on this Hangup to hang up performance of a command, for me a riddle. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
2006/11/1, Armin Schindler [EMAIL PROTECTED]: On Wed, 1 Nov 2006, Pedro Silva wrote: As you can see in the log below, the called number is just '0': CalledPartyNumber = 810 It seems DDI 0 of your line was called. So just do exten = 0,n,Dial... Armin Is that right! Thanks! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lucent TNT Help
I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to merge these all on to one TNT. I found out how to do dnis-or-voip for the call type on the voip line which allows me to set based on dialed number if its going to go to a modem or voip call, however, I'm trying to figure out how to set up the TNT to have voip origination use a certain PRI in the pool as the primary and then fail over to the other PRI's. I think it will probally involve setting up trunk groups, but I'm not entirely sure how I would set the trunk group for origination. Can anyone give me some friendly advice to try to figure this out? -Corey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
On Thu, 2006-11-02 at 11:32 +0200, Tzafrir Cohen wrote:
On Thu, Nov 02, 2006 at 10:10:01AM +0100, Florian Hars wrote:
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a TDM[124]xxP, all channels
change their numbers if you do a /etc/initd/zaptel restart. This is
slightly confusing.
The order of the channels is the order in which the spans register to
Zaptel, which is basically the order in which the modules load.
On Debian, load the modules through /etc/modules . Otherwise they will
be loaded through hotplug/udev in an unpredictable order (by the order
of PCI slots) which may or may not be the order that you like.
Gentoo has an equivalent file, whose name I forgot.
Redhats seem to lack such a mechanism, and I'm not sure whther or not
those cards do get hotplugged/coldplugged. Thus the tsrange need to load
them in the zaptel startup script.
Anyway, the order in which you happened to load them right now is not
guaranteed to be the order in which you load them next time unless you
explicitly
(I'd file a bug if there were a bug tracking system
that allowed users to submit bugs).
Users are surely allowed. Just register.
Also, bug reports to xpp/genzaptelconf are welcomed. It should be able
to write such module loading lists that should provide predictable order
in both Debian and Redhats.
For Redhat, Fedora, CentOS and other derivatives:
You can play tricks in /etc/modprobe.conf using the install directive.
The man page for modprobe.conf gives an example.
You could also force their loading and presumably their order in initrd
or rc.modules which runs as part of rc.sysinit.
rc.modules is the cleanest approach (IMHO), as initrd gets rebuilt by
some updates (e.g. kernel).
Bob...
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[asterisk-users] How to clear trixbox configuration
Hello all, To test some configs i forgot the trixbox web config (freepbx) and i made changes directly in asterisk config files (sip.conf, extensions.conf, etc). Result: asterisk is working ok but the the web config is totaly confused and, if i made a change via freepbx this not works ok. Only now i know that this changes will be made in file_custom.conf... problem of newbie... :). I also updated the asterisk for version 1.2.12.1, independently for the trixbox updating system. My trixbox version is 1.2.2. So i need to clear all configuration and start again only with the web config in freepbx. Is possible to clear all web configs and restitute all initial /etc/asterisk/* files to start from zero without re-installing all trixbox box from CD? Thanks in advance! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can some moderator kick this person out of the list
I myself and I am sure hundreds of other users on this mailing list aregetting very much annoyed on receiving follwoing autogenerted message several times a day from [EMAIL PROTECTED] . Is there any moderator on the list who can take care of this. It comes replied to every post and almost to every answer to it. From: [EMAIL PROTECTED] [EMAIL PROTECTED]Date: Nov 1, 2006 3:22 PM Subject: Benachrichtung zum =?unicode-1-1-utf-7?Q?+ANw-bermittlungsstatus (Fehlgeschlagen)?=To: [EMAIL PROTECTED] Dies ist eine automatisch erstellte Benachrichtigung +APw-ber den Zustellstatus.+ANw-bermittlung an folgende Empf+AOQ-nger fehlgeschlagen. *@ prebit.netFinal-Recipient: rfc822;*@ prebit.netAction: failedStatus: 5.1.1 -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VM Language
What is the best way to have the voicemail system and system do more than one language I know I have to have all wav, gsm files on the system. -- Best regards, Al Bochter Bochter Services (Voip PBX) Free World DialUp: 780217 EXT: 250 WebSite: http://www.freeworlddialup.com/ http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
Switching is what you want. NAT is Network Address Translation that allows the router to map IP addresses between router interfaces. You may wish to verify that all of the ports on your network, if automatically negotiated, did what you want. Probably, 100Mb, Full-Duplex. If not then force the config. Improper negotiation tends to drop packets. Everything appears to work, but slowly. Depending on your network infrastructure, you may also look into QOS. Bob... On Wed, 2006-11-01 at 16:15 -0500, Zeeshan Zakaria wrote: All the phones already have the latest firmware. They keep updating themselves automatically. In my setup of Grandstream phones, all the computers of the network go through the phones, i.e. I am using the builtin phones as swithces. They all have 2 ethernet ports. Does this has to do anything with the voice quality, or do I need to change something in the phones' setup, like switching it from switch to router in basic settings? What is this NAT/Router setting anyways and how should it be setup? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: ${CALLERIDNUM}
SP == Scott Pinhorne [EMAIL PROTECTED] writes:
SP I am setting up my phones so that if the callerID is 3 digits the
SP phones ring one way if it is more than 3 digits it rings another
SP i.e. internal calls and external calls.
SP exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
SP This will tell it to jump to 5 if callerID if but how do i
SP tell it do jump based on length of callerID?
There has been lots of answers to this one, but how about simply:
exten = /XXX,1,Goto(threedigits)
exten = /XX,1,Goto(twodigits)
exten = /.,1, ...
/Benny
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RE: [asterisk-users] Videoconferencing solutions with Asterisk-
Hi Maxx, Think you are referring to the bounty I put together with a few of my business friends. You're right nothing came out of it and I had a custom developed solution built using Adobe FMS (now no longer available as they phased out the 10 seat license and the minimum is now 100 seats). I've never heard of Adiance, any info? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Thursday, 2 November 2006 12:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Videoconferencing solutions with Asterisk- Does anyone have any experience with this? We're looking to deploy a pretty robust HiDef Video Conferencing solution, and if it were built around Asterisk, that'd be a huge bonus. It looks like a bounty was offered on it for a while with no results, and now an Indian company - Adiance - claims to have a solution, but I can't find any real feedback on it from end users. What do you guys use for Video Conferencing? Any recommendations? --Maxx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Example Polycom function key config
Jamie Heckford wrote: FYI - Polycom have confirmed to me that you can only send one digit via the programmable feature keys. Search the archives for the last month. If I recall correctly, if you are using firmware 2.0.1 it will allow you to map a speed dial and the speed dial can be programmed to send multiple digits. I'll be looking at this on Saturday. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally arrived and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so cheap calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lucent TNT Help
Hell we sold ours off about 2 1/2 years ago...so I am a little rusty on em now... But this was always a pretty easy manual to look at: http://www.hal-pc.org/~ascend/MaxTNT/ - Original Message - From: Corey Frang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 02, 2006 6:27 AM Subject: [asterisk-users] Lucent TNT Help I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to merge these all on to one TNT. I found out how to do dnis-or-voip for the call type on the voip line which allows me to set based on dialed number if its going to go to a modem or voip call, however, I'm trying to figure out how to set up the TNT to have voip origination use a certain PRI in the pool as the primary and then fail over to the other PRI's. I think it will probally involve setting up trunk groups, but I'm not entirely sure how I would set the trunk group for origination. Can anyone give me some friendly advice to try to figure this out? -Corey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.13.21/511 - Release Date: 11/1/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
On Thu, Nov 02, 2006 at 06:34:03AM -0500, Bob Chiodini wrote:
On Thu, 2006-11-02 at 11:32 +0200, Tzafrir Cohen wrote:
On Thu, Nov 02, 2006 at 10:10:01AM +0100, Florian Hars wrote:
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a TDM[124]xxP, all channels
change their numbers if you do a /etc/initd/zaptel restart. This is
slightly confusing.
The order of the channels is the order in which the spans register to
Zaptel, which is basically the order in which the modules load.
On Debian, load the modules through /etc/modules . Otherwise they will
be loaded through hotplug/udev in an unpredictable order (by the order
of PCI slots) which may or may not be the order that you like.
Gentoo has an equivalent file, whose name I forgot.
Redhats seem to lack such a mechanism, and I'm not sure whther or not
those cards do get hotplugged/coldplugged. Thus the tsrange need to load
them in the zaptel startup script.
Anyway, the order in which you happened to load them right now is not
guaranteed to be the order in which you load them next time unless you
explicitly
(I'd file a bug if there were a bug tracking system
that allowed users to submit bugs).
Users are surely allowed. Just register.
Also, bug reports to xpp/genzaptelconf are welcomed. It should be able
to write such module loading lists that should provide predictable order
in both Debian and Redhats.
For Redhat, Fedora, CentOS and other derivatives:
You can play tricks in /etc/modprobe.conf using the install directive.
The man page for modprobe.conf gives an example.
This is not the proper place: those are not real dependencies. You may
actually want to load those modules separately one day.
You could also force their loading and presumably their order in initrd
or rc.modules which runs as part of rc.sysinit.
Hmmm... sounds nice, however the text I read there is:
# Load modules (for backward compatibility with VARs)
if [ -f /etc/rc.modules ]; then
/etc/rc.modules
fi
Is it guranateed to remain there?
rc.modules is the cleanest approach (IMHO), as initrd gets rebuilt by
some updates (e.g. kernel).
And can't easily be re-run.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] connecting internal line with external line
Hi, I'm new to asterisk. I want asterisk to connect a external line with an internal line: the PC dials a number and connects this call to a internal telephone (telephone switchboard, based on ISDN, 4 analogue telephones) of my office. Can somebody here give me keyword how to search (e.g. with google) to realise it? tia Ekkard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error installing asterisk, module zaptel not found
After deciding to move a semi working asterisk setup to another box, installing and recompiling asterisk, addons and zaptel, modprobe zaptel says, module not found. Following various tales of how to modify udev stuff, still get that error. lspci does show the board in the list. All the LED's on the back of the board are dark. I have a TDM400p (tdm22b). I did not actually install the board, until after asterisk and add ons were complied. Just before the steps to compile zaptel. After installing board and playing doing the udev hack dance, did recompile with same results, as stated. What could be the probem(s)? phoneman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
Tzafrir Cohen wrote: Anyway, the order in which you happened to load them right now is not guaranteed to be the order in which you load them next time unless you explicitly Yes, I know. And I managed to fix it. The problem is that the distribution (zaptel-1.2.10) comes with two different explicit orderings, zaptel.sysconfig has tor2 wct4xxp wct1xxp wcte11xp wctdm24xxp wcfxo wctdm wcfxs wcusb wcfxsusb torisa ztdummy xpp_usb while the Makefile calls genmodconf with tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy This loads wct24xxp before wct1xxp (and wct4xxp is missing, as per http://bugs.digium.com/view.php?id=8071). So you get different orderings on boot and after /etc/init.d/zaptel restart Users are surely allowed. Just register. I've long since given up registering to bug trackers, there are far too many of them, and I don't want to remember a username/password pair for every program I use. Yours, Florian. -- Dr. Florian Hars | BIK ASCHPURWIS + BEHRENS GmbH | Büro, papierloses (n): Feldbrunnenstr. 7, 20148 Hamburg |Büro, in dem große Haufen Papier (040) 41 47 87 -21, Fax: -15 |lose herumliegen (FdI#321) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Subscriptions and call back on busy problems with Snom phones
Hello guys, as davidded wrote on voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+phone+snom), currently the Call completion feature of the SNOM phones interferes with subscriptions to Asterisk. To make subscriptions work correctly, you have to disable the call_completion feature of the snom phones. Without disabling this feature, after a call the subscriptions of the monitored extensions get lost. Unfortunately call_completion is needed for call back on busy. So you have decide between subscriptions or call back on busy :-(... Has anybody else experinced these problems? Maybe anyone got a patch :-)? Thanks, Andreas Haardt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
But how do you deal with the cable co blocking the ports you need for SIP? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally "arrived" and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so "cheap" calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Panasonic KX Model
I have to do this configuration with a panasonic KX-TD1232 model. You need some other information about the panasonic system?.Thanks. G. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to determine which version is running
Hi, Is it possible to see which version of libpri and zaptel that's currently running/loaded for example in the * CLI? _ Få de bedste søgeresultater med MSN Search: http://search.msn.dk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Java Web Phone
Be aware that any web phone will still be running on the client, so NAT and firewall issues may be harder to manage from a web phone. Unless someone has a service that pipes the audio over port 80 and converts it at the server. -- -- Steven http://www.glimasoutheast.org Vladimir Montealegre Estailes [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar más rápido en Internet.Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP v IAX2
FCC if you are in the USA. Simple. Otherwise find another broadband provider. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 But how do you deal with the cable co blocking the ports you need for SIP? Best regards,Al BochterBochter Serviceshttp://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers.Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security itemshttp://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon,Well Skype was one of the reasons I started my Asterisk based business.I first came across a VoIP demo about 12 years ago in a teleco carrier inAltanta GA.At that time the technology was very primitive (most people still had dialup lines). Anyway, to cut a long story short it wasn't until I many yearslater that I tried Skype, then I knew the technology had finally arrivedand was good enough for business communications. Here in Canada, longdistance is realitvely inexpensive so cheap calls are not very important Most of my clients are sold on the feature set in Asterisk and theability to have extensions in multiple sites/offices without any linecosts.Henry L.Coleman CEO*VoIP-PBX* 1-866-415-5355Toronto OntarioCanada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and badpoints In the end, sales/marketing perception will always win regardlessof better technologies. That will be Skype then ;-)--Jon FarmerTelford, Shropshire, UK ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersInbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto dial out and auto answer
Yes it is possible. - Original Message - From: Michel [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 02, 2006 11:20 AM Subject: [asterisk-users] Auto dial out and auto answer Hello list, We have 2 asterisk servers (without firewall and NAT), and We want to do : From the first server, we have a .call file which dial out to the second server. The second server automatically answers and Play a music during X seconds, then it hangs up. Is it possible? Thanks you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Java Web Phone
I created the xtn file how do I use it ? Any demo or I have to buy it from them ? - Original Message - From: Carlos Rojas To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 02, 2006 3:39 AM Subject: Re: [asterisk-users] Java Web Phone Hello,LookX-web litehttp://www.asterisk-es.org/modules/mydownloads/visit.php?cid=6lid=12Regards On 11/1/06, Vladimir Montealegre Estailes [EMAIL PROTECTED] wrote: Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar más rápido en Internet.Tutopia es Internet para todos. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to determine which version is running
René Christensen wrote: Hi, Is it possible to see which version of libpri and zaptel that's currently running/loaded for example in the * CLI? no. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Example Polycom function key config
Jamie Heckford wrote: FYI - Polycom have confirmed to me that you can only send one digit via the programmable feature keys. Search the archives for the last month. If I recall correctly, if you are using firmware 2.0.1 it will allow you to map a speed dial and the speed dial can be programmed to send multiple digits. I'll be looking at this on Saturday. Doug Hi Doug, AFAIK (from looking through the archives) this will only allow you to send the digits onhook, not during a call. If it works during a call then excellent, I'll try have a play tomorrow and let you know how it goes as well. Jamie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Java Web Phone
Hi Steven, Feel free to give me a call at www.cognation.net/contact anytime between 8am to 8pm New York time. I think you'll find you don't have nat issues as long as you have port 4569 udp unblocked. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven Sent: Thursday, 2 November 2006 8:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Java Web Phone Be aware that any web phone will still be running on the client, so NAT and firewall issues may be harder to manage from a web phone. Unless someone has a service that pipes the audio over port 80 and converts it at the server. -- -- Steven http://www.glimasoutheast.org Vladimir Montealegre Estailes [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar más rápido en Internet.Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
VOIP is NOT telephone so the FCC don't have anything to say about VOIP. Well not right now. But in CAN there are cable co. that block the SIP ports and there is an up charge for them to unblock SIP. Ask Vonage.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Dean Collins wrote: FCC if you are in the USA. Simple. Otherwise find another broadband provider. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 But how do you deal with the cable co blocking the ports you need for SIP? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally "arrived" and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so "cheap" calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 9:44:20 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to determine which version is running
On Thu, Nov 02, 2006 at 02:39:14PM +0100, René Christensen wrote: Hi, Is it possible to see which version of libpri and zaptel that's currently running/loaded for example in the * CLI? You can tell the versions of the modules that are currently in your filesystem (not necessarily those that are currently loaded) using modinfo: modinfo zaptel /sbin/modinfo zaptel The same information for the running module (at least for more recent kernels) : cat /sys/modules/zaptel/version Not sure about Zaptel's userspace tools and for libpri. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.4.4 Released!
Hello everyone, I have released AstLinux 0.4.4. Thanks to all of the testers on astlinux-users, AstLinux 0.4.4 now includes mISDN support (again). Grab AstLinux at http://www.astlinux.org. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] light web user interface
Curt Shaffer ([EMAIL PROTECTED]) wrote: Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc. I would also like it open enough that I can add features to it. I'm not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I can't see where you can lock them down to only see information on a particular extension. probably VoiceOne (http://www.voiceone.it/) is wath you need. -- Jonathan Alberto Rivera Gomez Grupo de Usuarios de GNU/Linux - UANL http://linuxuanl.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Example Polycom function key config
Jamie Heckford wrote: Jamie Heckford wrote: If it works during a call then excellent, I'll try have a play tomorrow and let you know how it goes as well. Thanks! It would save me some time. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Bandwidth questions
Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself. So, to clarify - Vonage has to have the necessary bandwidth to handle whatever amount of simultaneous calls. I can imagine that one vonage user calling another vonage user would use some sort of sip re-invite and perhaps even calls to other huge providers (packet8) are direct client to client. (Last time I read about this it seems that even calls to other large voip providers go through the PSTN though). Barring voip to voip calls, everything must run through their bandwidth right? If I'm right on this, I guess we need to come up with some sort of viable business model to do sell our own service. I want to concentrate on smb clients to whom we can then provide an asterisk box which would leave our bandwidth free, but my boss isn't particularly keen on this route. Anyways, Thanks for any insight and advice on this question, sorry if I'm asking this in the wrong place Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX problem
You are correct, I don't have entry for them to be able and authenticate! how should I define it? Should it be a peer or a user? Could you please add an example? Thanks a million... Itamar Lavender IT Manager Direct: +1 646 485 1828 __ Traiana, Inc 51 E. 42nd St., 10th Fl New York, NY 10017 Main: +1 212 404 1714 Fax:+1 656 536 4900 www.traiana.com The information contained in this e-mail is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others explicitly authorized to receive it. If you have received this e-mail in error, please destroy it and delete it from your computer. Any disclosure, copying or distribution of the information is strictly prohibited and may be unlawful. No responsibility can be accepted to any end users for any action taken on the basis of the information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Thursday, November 02, 2006 04:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX problem On 2 Nov 2006, at 02:38, Itamar Lavender wrote: Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register = username:[EMAIL PROTECTED] What does the rest of iax.conf look like ? Auth is a 2 way thing - you have sucessfully registered with them, but when they send you a 'new' your box fails to authenticate them as it can't find a matching user/friend entry in iax.conf. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP v IAX2
Hi Al, You might want to check out http://www.eweek.com/article2/0,1895,1773983,00.asp (this was last year and the first one that popped up in google-I didnt look very far) But what the hell do I know. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 VOIP is NOT telephone so the FCC don't have anything to say about VOIP. Well not right now. But in CAN there are cable co. that block the SIP ports and there is an up charge for them to unblock SIP. Ask Vonage.. Best regards,Al BochterBochter Serviceshttp://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers.Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security itemshttp://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Dean Collins wrote: FCC if you are in the USA. Simple. Otherwise find another broadband provider. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 But how do you deal with the cable co blocking the ports you need for SIP? Best regards,Al BochterBochter Serviceshttp://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers.Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security itemshttp://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon,Well Skype was one of the reasons I started my Asterisk based business.I first came across a VoIP demo about 12 years ago in a teleco carrier inAltanta GA.At that time the technology was very primitive (most people still had dialup lines). Anyway, to cut a long story short it wasn't until I many yearslater that I tried Skype, then I knew the technology had finally arrivedand was good enough for business communications. Here in Canada, longdistance is realitvely inexpensive so cheap calls are not very important Most of my clients are sold on the feature set in Asterisk and theability to have extensions in multiple sites/offices without any linecosts.Henry L.Coleman CEO*VoIP-PBX* 1-866-415-5355Toronto OntarioCanada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and badpoints In the end, sales/marketing perception will always win regardlessof better technologies. That will be Skype then ;-)--Jon FarmerTelford, Shropshire, UK ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersInbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 9:44:20 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream HandyTone-488 with Asterisk ?
Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running asterisk with 'sudo'
Hi guys, I'm using RedHat and am trying to configure my sudo to enable user 'testuser' to run Asterisk. However whenever I try to run 'sudo asterisk' as 'testuser' I get prompted for password. This is the line in my sudoers configuration file that I thought should do the trick, but it doesn't: testuser ALL=NOPASSWD: /usr/sbin/asterisk Does anyone know how to configure the sudo so that 'testuser' will be able to run the asterisk? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Problems
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten = _X.,1,Set(TIMEOUT(absolute)=0)
exten = _X.,2,NoOp(${EXTEN})
exten = _X.,3,DEADAGI(live-full.php)
exten = _X.,4,Wait,2
exten = _X.,5,Hangup
The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
works flawlessly in all but one aspect which I believe is related to
asterisk rather than the script itself.
As the script is launched using DEADAGI I expect it to carry on after
the channel has been hungup (to save the results of the user input in
this case) which works unless the users are leaving a voice message at
the time. The script uses record_file and records ok if the user ends
the call with a keypress (#) but if the user hangs up once they have
finished leaving their message the script exits immediately rather than
carrying on:
Nov 2 11:45:57 VERBOSE[24262] logger.c: AGI Rx RECORD FILE
/ivr/recordedtemp/1162467957 wav # 12 0 BEEP s=5 Nov 2 11:45:57
DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to read format slin
Nov 2 11:45:57 DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to
write format ulaw
Nov 2 11:45:57 VERBOSE[24262] logger.c: -- Playing 'beep' (language
'en')
Nov 2 11:45:58 DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to
write format ulaw Nov 2 11:46:01 DEBUG[3658] chan_iax2.c: Immediately
destroying 1, having received hangup Nov 2 11:46:01 VERBOSE[24262]
logger.c: AGI Tx 200 result=0 (hangup) endpos=22560 Nov 2 11:46:01
DEBUG[24262] pbx.c: Spawn extension (live-full,70,3) exited non-zero
on 'IAX2/AQL IAX-1'
Does anyone know why the channel is closing down immediately rather than
waiting for the script to exit?
Thanks
Jordan
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Re: [asterisk-users] Snom or Cisco Phones?
CP,I've never heard complaints about the Snom 320 speakerphone hardware, nor do I think they sound bad from using the phones myself. I believe Snom did make a significant improvement to their speakerphone hardware not to long ago.Of course, there is never any guarantee on the "quality" of something, since I don't think something can please everyone. Maybe other users of this phone can post their feedback here as well and based upon that you can get a good idea of what your results should be. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 5:24 PM, cp wrote: Do the speakerphone’s work well on Snom 320’s? I have a Linksys 841 and could never get the speakerphone working well. -CP From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, November 01, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom or Cisco Phones? Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on the keypad for some non-tehcy end users). But they have terrific functionality and great audio quality in most office environments, and are very easy to set up and install. Everyone seems to really love them. The Cisco phones are nice as well, but IF you decide to go with Cisco, READ what you are buying and what you are getting before just blindly buying it (in fact, do this anyways, it's common sense to do this before buying any product, anywhere). Cisco products normally don't come with half of the items you need, and unfortunately most resellers (and Cisco) don't make this too easy to read and understand. DO NOT buy refurbished Cisco if you want support, especially since there has been some bogus Cisco voice equipment shipping lately from some of the certified Cisco resellers/distributors. Network World had an article on this recently: http://www.networkworld.com/news/2006/102306counterfeit.html Cisco may have a great look to their phones and have the design very well thought out (not to mention the big Cisco name - which is good enough for some), but they are normally harder to install and configure and are VERY proprietary. If you buy Cisco, Cisco wants you to ONLY buy Cisco (for support and marketing reasons). Snom 320's are a great choice just because these phones mainly support everything the Snom 360's support (i.e. sidecars) Only main differences between these two models is that the Snom 360 has the larger LCD screen as well as newly added XML support. We have about 50 stations here, some management, some support, some sales and have pretty much decided as a company to completely use Snom phones for all of our employees. Keep in mind, each phone out there will have their specific pro's and con's, as well as quarks ... seems there is no real "perfect phone" out there yet. But Snom in my mind, is pretty dang close. This all of course is just personal opinion from past experience. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 8:35 AM, Tom Vile wrote: I love the Snom phones as well. The function keys are great and easy to use. On 10/31/06, mitcheloc [EMAIL PROTECTED] wrote:My vote is definitely for Snom, I've worked with Cisco phones for years, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy. On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote: On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360)* Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the "upgrade" path from sccp to sip a totally unnecessary annoyance.
Re: [asterisk-users] Asterisk architecure
Guillaume, merci beaucoup - thanks a lot. The links were very useful, especially tech-inivite, the owner of the site did a great job. I got three books last night as well: Asterisk, Understanding SIP (Alan Johnston) and Practical VOIP Security. From the books and the website I can only understand that Asterisk comes in as a registrar/location server. In other contexts, such as a p2p environment, I assume that no proxies are used and no location server is required either. So my question is again, where does Asterisk fit in? I can perhaps see that in an incoming call screening scenario (as depicted here: http://www.tech-invite.com/Ti-sip-service-14.html) Asterisk acts as the announcement server. But in other cases, such as 'transfer unattended' or 'call forwarding - busy' (http://www.tech-invite.com/Ti-sip-service-8.html), there is no apparent need for Asterisk, is that correct? I would tremendously appreciate a little more detail on scenarios where Asterisk is required/used. Jez --- G(P)L [EMAIL PROTECTED] wrote: jez . a écrit : Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy. I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? Thank you Hi, You can find some interesting diagram here : http://www.tech-invite.com/Ti-sip-dialog.html Other diagrams more architecture ortiented : http://lehmann.free.fr/divers/SIP%20tutorial.pdf slides 32 and after. The document is not mine :) If you want something more specific to Asterisk's architecture, I recommand you this book : http://www.eyrolles.com/Informatique/Livre/9780596009625/livre-asterisk.php Bye Guillaume Lehmann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Everyone is raving about the all-new Yahoo! Mail (http://advision.webevents.yahoo.com/mailbeta/) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Static Realtime Select from Database
I did an ngrep trace of what Asterisk realtime static does when it queries the database. Here's what I saw SELECT category, var_name, var_val, cat_metric FROM rt_pbx1_sip_vw WHERE filename='sip.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id; Firstly, why does it order in DESCENDING order by cat_metric? Shouldn't it be ASCENDING order? The docs imply that you use cat_metric to specify the order that you want your contexts to appear. If Asterisk sorts in descending order, they will be reversed! Also, is cat_metric really required? For example, in the following set of data, rows for dundipbx1 are separated by rows for other contexts. Is Asterisk smart enough to know it has to combine this data into the one context? I realise that var_metric is useful in situations like deny/permit. +---++--++ | category | var_name | var_val | cat_metric | +---++--++ | dundipbx1 | context| syst_DUNDiPhoneMap | 0 | | dundipbx1 | deny | 0.0.0.0/0.0.0.0 | 0 | | dundipbx1 | secret | X| 0 | | dundipbx1 | type | user | 0 | | dundipbx1 | username | dundipbx1| 0 | | dundipbx2 | deny | 0.0.0.0/0.0.0.0 | 0 | | dundipbx2 | fromuser | dundipbx2| 0 | | dundipbx2 | host | pbx2.ipt.XXX.com | 0 | | dundipbx2 | secret | FBoAsFz3S8hFQ| 0 | | dundipbx2 | type | peer | 0 | | dundipbx2 | username | dundipbx2| 0 | | dundipbx3 | deny | 0.0.0.0/0.0.0.0 | 0 | | dundipbx3 | fromuser | dundipbx3| 0 | | dundipbx3 | host | pbx3.ipt.XXX.com | 0 | | dundipbx3 | permit | xxx.yyy.142.201 | 0 | | dundipbx3 | permit | xxx.yyy.142.203 | 0 | | dundipbx3 | permit | xxx.yyy.142.204 | 0 | | dundipbx3 | secret | X| 0 | | dundipbx3 | type | peer | 0 | | dundipbx3 | username | dundipbx3| 0 | | general | allowguest | no | 0 | | general | autodomain | no | 0 | | general | bindaddr | xxx.yyy.142.203 | 0 | | general | bindport | 5060 | 0 | | dundipbx1 | permit | xxx.yyy.142.204 | 0 | | dundipbx2 | permit | xxx.yyy.142.203 | 0 | | general | allow | ulaw | 0 | +---++--++ 81 rows in set (0.00 sec) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Phone can not find TFTP server
David Are you using a separate V-Lan for voice? Your phone may require CDP to be enabled. Clint This email message is intended only for the addressee(s) and contains information that may be confidential and/or copyright. If you are not the intended recipient please notify the sender by reply email and immediately delete this email. Use, disclosure or reproduction of this email by anyone other than the intended recipient(s) is strictly prohibited. No representation is made that this email or any attachments are free of viruses. Virus scanning is recommended and is the responsibility of the recipient. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Thursday, November 02, 2006 12:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom 601 Phone can not find TFTP server Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying Could not contact boot server, using existing configuration. I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Phone can not find TFTP server
This may help,http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-24.htmSounds like you don't have a configuration file on your TFTP server. After that message does it say anything about "Error loading (mac address).cfg!" and will then reboot?If so, get the MAC.cfg file in there. The article should explain this better. Not a guaranteed fix, but should help you locate and fix the problem.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 2, 2006, at 12:51 AM, Klaverstyn, David C wrote: Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying “Could not contact boot server, using existing configuration”. I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Java Web Phone
On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote: Hello list partners you know about a softphone made in java attachable in a web page? GNU! I'm using JIAXClient [1] to permit to any user to join one meetme room [2] with the IAX2 protocol, works very great for me, and is very easy to install and modify to your needs. [1] http://www.hem.za.org/jiaxclient/ [2] http://www.rmsenecuador.info/jiaxclient/index.html Thaks in advance! __ Visita www.tutopia.com y comienza a navegar más rápido en Internet.Tutopia es Internet para todos. Web Bug from http://www.tutopia.com/bannerserving/banman.asp?ZoneID=0BannerID=3494AdvertiserID=776CampaignID=2739Task=GetMode=TEXT ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime, DUNDi and regexten
We've been using DUNDi, Realtime, and regexten extensively for months now, and it's been working great since we got it running. We don't use the NoOp's for anything other than extension discovery since creating the NoOp's in already configured contexts isn't very stable. Aaron On Thu, 2006-11-02 at 00:29 -0500, Andrew Joakimsen wrote: I can't even get regexten to work with config files On 11/1/06, Douglas Garstang [EMAIL PROTECTED] wrote: It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html Someone else posted what appears to be the same issue to the bug tracker 3 years ago: http://bugs.digium.com/view.php?id=3053 Mark Spencer closed the bug, saying it was a configuration issue, and to use 'includes'. Not sure what he means by that. Has anyone got this to work? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
Dean, I am going ways offtopic on this one, but I must say that although I am not from the US, I am finding the repetitive bad communications and VoIP legislation being voted in the US disturbing. Having in mind the CALEA Act (as an outrageous example of such bad legislation yet not directly related to the link you posted), how indeed could Consumers Put End to VOIP Port Blocking with the COPE Act (http://www.commoncause.org/atf/cf/%7BFB3C17E2-CDD1-4DF6-92BE-BD4429893665%7D/HR5252_COPE.PDF) pending final approval by the US authorities? I am not willing to start a discussion on net neutrallity here on this list (anyways not in this topic), just putting my two cents in. Cheers On 11/2/06, Dean Collins [EMAIL PROTECTED] wrote: Hi Al, You might want to check out http://www.eweek.com/article2/0,1895,1773983,00.asp (this was last year and the first one that popped up in google-I didn't look very far) But what the hell do I know. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 VOIP is NOT telephone so the FCC don't have anything to say about VOIP. Well not right now. But in CAN there are cable co. that block the SIP ports and there is an up charge for them to unblock SIP. Ask Vonage.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Dean Collins wrote: FCC if you are in the USA. Simple. Otherwise find another broadband provider. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 But how do you deal with the cable co blocking the ports you need for SIP? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally arrived and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so cheap calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006
Re: [asterisk-users] Running asterisk with 'sudo'
On Thu, 2006-11-02 at 17:12 +0100, Asterisk wrote: Hi guys, I'm using RedHat and am trying to configure my sudo to enable user 'testuser' to run Asterisk. However whenever I try to run 'sudo asterisk' as 'testuser' I get prompted for password. This is the line in my sudoers configuration file that I thought should do the trick, but it doesn't: testuser ALL=NOPASSWD: /usr/sbin/asterisk Does anyone know how to configure the sudo so that 'testuser' will be able to run the asterisk? Use the visudo to make changes to /etc/sudoers , to make the sudo stop asking for a password you need a line at /etc/sudoers like (take note on the space after the = ): testuser ALL= NOPASSWD: /usr/sbin/asterisk Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
On 2 Nov 2006, at 14:54, Al Bochter wrote: VOIP is NOT telephone so the FCC don't have anything to say about VOIP. Well not right now. But in CAN there are cable co. that block the SIP ports and there is an up charge for them to unblock SIP. Ask Vonage.. Yet another advantage of us using IAX - although if it gets mega popular they will block that too. Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
On Wed, 1 Nov 2006, Henry.L.Coleman wrote: I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Set the Early Dial option - it's on a per-line basis, then as soon as Asterisk gets a number it can dial, it will. No need to wait the 4 seconds or press the send button... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Grandstream HandyTone-488 with Asterisk ?
On 2006-11-02 07:51:15 -0800, Noc Phibee [EMAIL PROTECTED] said:
Hi
anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ?
Actually my HandyTone 488 are connected to:
wan port to my lan
line FXO port are connected to my local analogic line
i want that when a call in by my analog line, it's sent to my asterisk
for other voip post can answer ..
it's possible ?
It's possible, but the FXO port on that device is more of a power out
fail over to PSTN in my opinion.
The FXO can be made to work, but it always had issues with my setup.
1) Echo problems. I have a long loop.
2) If calls are picked up on the first ring, they go into la la land.
3) General reliability and stability issues.
The dial plan looked like this (where 2003 is the extension of the FXO).
exten = _NXX,1,dial(SIP/@2003,60,D(ww${EXTEN}))
This dials 7 digit number through the extension 2003.
As far as incoming calls, you need to set the HT-488 up from it's
config. screens and and it's a regular SIP extension in asterisk.
Good Luck,
Marty
PS If you make this work great, please do let us know how!
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[asterisk-users] Re: VOIP Bandwidth questions
On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said: snip My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself.snip For one thing, I suppose they use codecs that compress the voice data as much as possible. Probably g729, or ilbc or some such. Also, it's not true that all the traffic need to flow through there servers. Once the connections are setup in a well designed system, the data could flow directly. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with MYSQL commands in dialplan
Hi,
I have been
using MYSQL successfully from the dialplan in the past, but I am stumped this
morning trying to use it for some other table. I am basically cut and
pasting from the wiki. Here is myCLI
output:
-- Executing
NoOp("SIP/11.11.11.11", "701") in new
stack -- Executing MYSQL("SIP/11.11.11.11", "Connect connid
localhostusernamepassworddatabase_name") in new stack ---(sameinput
works at other places) -- Executing
MYSQL("SIP/11.11.11.11", "Query resultid 1
SELECT redirection_data from redirections where accountcode=514555 and exten=701 and active=true") in new
stack
-- Executing MYSQL("SIP/11.11.11.11", "Fetch fetchid 2
redirection_data") in new stackNov 2 12:51:30 WARNING[24892]:
app_addon_sql_mysql.c:115 find_identifier: Identifier 2, identifier_type 2 not
found in identifier listNov 2 12:51:30 WARNING[24892]:
app_addon_sql_mysql.c:330 aMYSQL_fetch: aMYSQL_fetch: Invalid result identifier
2 passed -- Executing MYSQL("SIP/11.11.11.11", "Clear 2") in new
stackNov 2 12:33:44 WARNING[24450]: app_addon_sql_mysql.c:115
find_identifier: Identifier 2, identifier_type 2 not found in identifier
listNov 2 12:33:44 WARNING[24450]: app_addon_sql_mysql.c:348
aMYSQL_clear: Invalid result identifier 2 passed in
aMYSQL_clear -- Executing MYSQL("SIP/11.11.11.11", "Disconnect 1") in new
stack -- Executing NoOp("SIP/11.11.11.11", "") in new
stack
I am running
1.2.4. Not even sure what the warning means (WARNING[24892]:
app_addon_sql_mysql.c:115 find_identifier: Identifier 2, identifier_type 2 not
found in identifier list).
Any help is
appreciated.
Mike
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Re: AW: [asterisk-users] Snom or Cisco Phones?
I concur, We run about 50 7960's and they work quite well. Sound quality is pretty good. Like Aaron said, unless you're running call manager you can't program soft keys, etc. We're looking at going to a different phone that would give us some more customization options. Also.. Cisco's come with skinny firmware. You'd have to acquire the SIP firmware from somewhere. And NO cisco will NOT 'give' it to you If you need some, I have some for sale! :) :) That and Cisco won't give you the time of day if you don't use their stuff ;) We have about 1600 of the Cisco's on campus, and unless you run them on the call manager, you're not gonna have nearly as many features as any other phone that's designed with SIP in mind. That said, if you need a phone with dialtone, a pretty screen, and limited xml services, then I will say that the cisco's are extremely easy to provision once you figure out the upgrade paths. (Oh, and we're running 7940's and 7960's... if you're looking at the 7912's, etc, good luck, they're a _complete_ pain to work with) Aaron On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote: I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo with spa-3000
check the impedance for the fxo port. I bet it's set to 600. change it to 900 and see what happens James Harper wrote: More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000 (only when the call is routed via SIP of course), but I think it's more noticeable on the PAP2 due to the increased latency. The echo occurs even when I call an automated voice service or something, so I'm thinking that possibly there is a gross impedance mismatch on my side of the telco switch. I believe the following is happening (latency measures are guesses): Handset -(0ms)- PAP2 -(40ms)- Asterisk -(40ms)- SPA3000 -(0ms)- Telco Does an EC algorithm need a measurable delay to work? The EC would have to cope with an almost unmeasurably small echo delay (the delay only creeps in on the other side of the IP link. Is there another way I should be solving this problem, especially as I can keep Asterisk in the loop if I want to? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- One day at a time, one second if that's what it takes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running asterisk with 'sudo'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 testuser ALL=(ALL) NOPASSWD: /usr/sbin/asterisk * testuser ALL=NOPASSWD: /usr/sbin/asterisk - -- Regards, Tod Detre Technical Lead Global Information Technology CampusEAI Consortium 1940 East 6th Street, 11th Floor Cleveland, OH 44114 Tel: 216.589.9626 x151 Fax: 216.589.9639 www.campuseai.org -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFSjST2ThzE/IuQ3sRAvo4AJ97R1ci0Wdo0MEi9E5dYVEm915GcwCfbclg R/s0KY2TQ9KOGrNqu5Kiwxo= =s9UD -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail issues
I put my voicemail groups into different contexts so that I can use Dial
by name and escape.
I had set ext 500 as
exten = 500,1,VoiceMailMain(${CALLERID(number)[EMAIL PROTECTED]|s)
but now that the contexts are different. this does not work
#1 how do I have everyone use an ext to get the voicemail regardless of
context.
#2 can I get the mail buttons to work on my polycom 501s and swissphones
#3 where do I put the i ext to allow the caller to go from the
voicemail back to a ext in the dialplan
Thanks
Jason
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Re: [asterisk-users] AstLinux 0.4.4 Released!
Hi Kristian What else is in the 0.4.4 release ?? Any news on the Sangoma A200 or Faxing ? Thanks Barry Kristian Kielhofner wrote: Hello everyone, I have released AstLinux 0.4.4. Thanks to all of the testers on astlinux-users, AstLinux 0.4.4 now includes mISDN support (again). Grab AstLinux at http://www.astlinux.org. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Phone can not find TFTP server
Are you sure you have the phone setup to use TFTP and not FTP? - Original Message - From: Klaverstyn, David C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 02, 2006 1:51 AM Subject: [asterisk-users] Polycom 601 Phone can not find TFTP server Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying Could not contact boot server, using existing configuration. I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] testing
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Re: [asterisk-users] Asterisk architecure
where exactly Asterisk 'fits' in my VOIP architecture.I think you should tell what do you want to do ? May be guys here can tell you whether Asterisk is the right choice for you or not. Anyways, talking about SIP and Asterisk, it can do SIP to PSTN (PRI/FXO) and vice versa. Means it can act as a media/voip gateway. Check this from you referece: http://www.tech-invite.com/Ti-sip-CF3666.htmlRemember, Asterisk is a PBX which supports SIP, other protocols and hardwares to interface with PSTN. Hope this helps little !.Vikki. On 11/2/06, je . [EMAIL PROTECTED] wrote: Guillaume, merci beaucoup - thanks a lot.The links were very useful, especially tech-inivite,the owner of the site did a great job. I got threebooks last night as well: Asterisk, Understanding SIP(Alan Johnston) and Practical VOIP Security. From the books and the website I can only understandthat Asterisk comes in as a registrar/location server.In other contexts, such as a p2p environment, I assumethat no proxies are used and no location server is required either.So my question is again, where does Asterisk fit in? Ican perhaps see that in an incoming call screeningscenario (as depicted here: http://www.tech-invite.com/Ti-sip-service-14.html)Asterisk acts as the announcement server. But in othercases, such as 'transfer unattended' or 'callforwarding - busy'( http://www.tech-invite.com/Ti-sip-service-8.html),there is no apparent need for Asterisk, is thatcorrect?I would tremendously appreciate a little more detailon scenarios where Asterisk is required/used. Jez--- G(P)L [EMAIL PROTECTED] wrote: jez . a écrit : Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy. I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? Thank you Hi,You can find some interesting diagram here : http://www.tech-invite.com/Ti-sip-dialog.html Other diagrams more architecture ortiented : http://lehmann.free.fr/divers/SIP%20tutorial.pdf slides 32 and after. The document is not mine :) If you want something more specific to Asterisk's architecture, I recommand you this book : http://www.eyrolles.com/Informatique/Livre/9780596009625/livre-asterisk.php Bye Guillaume Lehmann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Everyone is raving about the all-new Yahoo! Mail(http://advision.webevents.yahoo.com/mailbeta/)___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP v IAX2
Hi Maka, I'm not from the USA either but thanks for the well researched answer. Yes I could have talked about net neutrality and the various legislations globally protecting it but I knew the first answer in google was enough. Besides my preferred answer was go find another broadband provider and stop whining like a sissy :) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of maka Sent: Thursday, 2 November 2006 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 Dean, I am going ways offtopic on this one, but I must say that although I am not from the US, I am finding the repetitive bad communications and VoIP legislation being voted in the US disturbing. Having in mind the CALEA Act (as an outrageous example of such bad legislation yet not directly related to the link you posted), how indeed could Consumers Put End to VOIP Port Blocking with the COPE Act (http://www.commoncause.org/atf/cf/%7BFB3C17E2-CDD1-4DF6-92BE- BD4429893665%7D/HR5252_COPE.PDF) pending final approval by the US authorities? I am not willing to start a discussion on net neutrallity here on this list (anyways not in this topic), just putting my two cents in. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Out Dial Interface for Asterisk
Hi All, I sent this a while back but never received any replies. My deadline is fast approaching so I thought I'd throw it out there again in hope of some advice. I need the ability to automatically out-dial and play a dynamically generated message. I then need the ability for the answering party to give feedback via touch tone. I am a .Net Programmer and I have looked at the Asterisk.NET examples, but all I see there is creating calls and sending them to system phones, etc. I don't see anyway of capturing responses back from the answering party, or how to play dynamically generated messages. Does anyone know if this is possible with the Asterisk.NET interface? Or does anyone know of another way to accomplish my needs? Any advice is greatly appreciated. --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] light web user interface
This looks a lot closer to what I need than anything else at this point. Thanks for the link, I'm gonna add start looking at adding functionality to this today! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Rivera Sent: Thursday, November 02, 2006 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] light web user interface Curt Shaffer ([EMAIL PROTECTED]) wrote: Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc. I would also like it open enough that I can add features to it. I'm not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I can't see where you can lock them down to only see information on a particular extension. probably VoiceOne (http://www.voiceone.it/) is wath you need. -- Jonathan Alberto Rivera Gomez Grupo de Usuarios de GNU/Linux - UANL http://linuxuanl.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regexten regcontext broken for SIP?
I am having the same issues. Did you ever file a bug report?On 10/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote:Hi ho,is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1being created upon SIP client registration, show dialplan xxx reveals no change.And yes, I have also read and checked bug 7144; if I go down that routeand no SIP client is registered I get a CLI warning that my standardcontext tries to include an empty context - go figure... http://bugs.digium.com/view.php?id=7144So, do I need to file a bug report, or is it working OK for others?Cheers, PhilippP.S.: Of course I am aware of this Wiki page: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still no CLI in 1.4 branch (OSX)
On 1 Nov 2006, at 19:51, Joshua Colp wrote: Martin Joseph wrote: I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it? Thanks, Marty I fixed this as of revision 46780 in the 1.4 branch. Give it a go. That works for me now. However the http manager port is non-responsive. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problems
Jordan Kirby wrote:
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten = _X.,1,Set(TIMEOUT(absolute)=0)
exten = _X.,2,NoOp(${EXTEN})
exten = h,1,DEADAGI(live-full.php)
___
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[asterisk-users] Tampa Bay Asterisk Users Meetup on Monday
Hello, We will be having another Tampa Bay Area Asterisk Users Meetup on Monday, November 6th at 7:30 PM. Asterisk users from gurus to new users are welcome. Along with user discussions, we will be talking about Astricon and Asterisk 1.4 at this meeting. We will also have free items from Digium to be given away. go to the site for more info: http://asteriskpbx.meetup.com/1/calendar/5178348/ See you there, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: VOIP Bandwidth questions
Martin Joseph wrote: On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said: snip My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself.snip Also, it's not true that all the traffic need to flow through there servers. Once the connections are setup in a well designed system, the data could flow directly. I'm not sure what you mean here. What connections? If they're terminating to the PSTN Either they're paying someone to do it or they're doing it themselves right? If they're doing it themselves they have to handle the bandwidth requirements. As I said before - if both endpoints are on vonage the data might go from device to device ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom latest version
Hi, Where should I go toget the Polycom`s latest official (non-beta) version? I am registered on the Polycomcustomer website but that doesn't seem accessible. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Bandwidth questions
This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. For example: Telco A has 100 subscribers to his phone service in a city (home and business), so he needs to ask himself a- Will the telco buy a switch that can handle 100 calls simultaneously? So he can provide service to his subscribers 100% of the time at any time of the day even during riots,panic,flood,etc? b- Or will the telco go for a balance and guess that at the peak time of the day he will have 75 simultaneous call, so he goes out and buy a switch that handles 75-80 calls at the same time? c- how many trunks will the Telco have to talk to other telcos? So telco in City A can communicate with Telco in city B (or even in the same city)? International voice providers suffer from this kind of problem. Some sell plastic cards with a local phone number and a pin so you call them to call to other cities/countries but that cheap voice provider has, let's say, ten thousand long distance lines and ten thousand local phone numbers, but they sell 100k plastic cards a month with a peak usage 3 times every ten days of 12thousand lines? obviously 2 thousand callers wont get connected (only 3 times every ten days in a specific time range) but the other 7 days the peak usage is 10thousand calls? Every ten days the provider try to connect 106k calls but fail to connect 6k calls, that's 6% failure rate every ten days (100% in a 7 days period and 98% in those 3 days). Can you live with that failure ratio? that's up to you. I don't work for a Telco, but a Telco may apply the dialup-internet rule (and they live happy with it) for 30subscribers-to-1line home users and 10(or 5)subscribers-to-1line for business. (correct me if I'm wrong please it will be nice to know real figures). So apply the same rule to you VoIP hosting. -What codec will you use? let say g711 and let's say it uses 100kilobits per leg. -How many subscribers will you have in a 6 month period? 500 -So to provide all of them with service you will need 48Megabits of bandwith all the time just to connect to your Telco equipments. - But you decide that you analyzed the usage patterns of your service and you will have only 125 subscribers calling other 125 subscribers (this is called On-Net) at peak time every day at 6pm (rush hour). So, go out and buy 24mbits of bandwidth only. - But you suddenly have the option to hire burst IP service where your IP carrier can provide you with more bandwidth if your usage starts to rise in any given time of the day. So you calculate again that your minimum constant usage at any time of the day is 40 users On-Net, so go out and buy 5mbits (for a total of 50 calls) of bandwidth with burst IP enabled from 6pm to 8pm of 48mbits (or 24mbits). This scenario is only subscriberyour_companysubscriber. you also need to calculate subscriber--your_companyother_telcos And the last but most important question is: how much money do you have to burn on this? 100% Uptime full-service, Top Carrier Class performance (and even they get busy sometimes)? or almost perfect service with the once-in-awhile glitch of we're sorry all circuits are busy, please try again. Hope this helps, How many times (at least in my country) haven't you suffered from Im sorry all circuits are busy, please try again during christmas midnight, new years eve, election days or similar behaviors that cause massive amounts of calls being initiated and received? So the answer to your question On 11/2/06, mail-lists [EMAIL PROTECTED] wrote: Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself. So, to clarify - Vonage has to have the necessary bandwidth to handle whatever amount of simultaneous calls. I can imagine that one vonage user calling another vonage user would use some sort of sip re-invite and perhaps even calls to other huge providers (packet8) are direct client to client. (Last time I read about
Re: [asterisk-users] VOIP Bandwidth questions
I forgot to tell that my rant is about a centrally handled servers, with no re-invite and no spider-like interconnects with smaller, geographically located switches. On 11/2/06, Erick Perez [EMAIL PROTECTED] wrote: This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. For example: Telco A has 100 subscribers to his phone service in a city (home and business), so he needs to ask himself a- Will the telco buy a switch that can handle 100 calls simultaneously? So he can provide service to his subscribers 100% of the time at any time of the day even during riots,panic,flood,etc? b- Or will the telco go for a balance and guess that at the peak time of the day he will have 75 simultaneous call, so he goes out and buy a switch that handles 75-80 calls at the same time? c- how many trunks will the Telco have to talk to other telcos? So telco in City A can communicate with Telco in city B (or even in the same city)? International voice providers suffer from this kind of problem. Some sell plastic cards with a local phone number and a pin so you call them to call to other cities/countries but that cheap voice provider has, let's say, ten thousand long distance lines and ten thousand local phone numbers, but they sell 100k plastic cards a month with a peak usage 3 times every ten days of 12thousand lines? obviously 2 thousand callers wont get connected (only 3 times every ten days in a specific time range) but the other 7 days the peak usage is 10thousand calls? Every ten days the provider try to connect 106k calls but fail to connect 6k calls, that's 6% failure rate every ten days (100% in a 7 days period and 98% in those 3 days). Can you live with that failure ratio? that's up to you. I don't work for a Telco, but a Telco may apply the dialup-internet rule (and they live happy with it) for 30subscribers-to-1line home users and 10(or 5)subscribers-to-1line for business. (correct me if I'm wrong please it will be nice to know real figures). So apply the same rule to you VoIP hosting. -What codec will you use? let say g711 and let's say it uses 100kilobits per leg. -How many subscribers will you have in a 6 month period? 500 -So to provide all of them with service you will need 48Megabits of bandwith all the time just to connect to your Telco equipments. - But you decide that you analyzed the usage patterns of your service and you will have only 125 subscribers calling other 125 subscribers (this is called On-Net) at peak time every day at 6pm (rush hour). So, go out and buy 24mbits of bandwidth only. - But you suddenly have the option to hire burst IP service where your IP carrier can provide you with more bandwidth if your usage starts to rise in any given time of the day. So you calculate again that your minimum constant usage at any time of the day is 40 users On-Net, so go out and buy 5mbits (for a total of 50 calls) of bandwidth with burst IP enabled from 6pm to 8pm of 48mbits (or 24mbits). This scenario is only subscriberyour_companysubscriber. you also need to calculate subscriber--your_companyother_telcos And the last but most important question is: how much money do you have to burn on this? 100% Uptime full-service, Top Carrier Class performance (and even they get busy sometimes)? or almost perfect service with the once-in-awhile glitch of we're sorry all circuits are busy, please try again. On 11/2/06, mail-lists [EMAIL PROTECTED] wrote: Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself. So, to clarify - Vonage has to have the necessary bandwidth to handle whatever amount of simultaneous calls. I can imagine that one vonage user calling another vonage user would use some sort of sip re-invite and perhaps even calls to other huge providers (packet8) are direct client to client. (Last time I read about this it seems that even calls to other large voip providers go through
RE: [asterisk-users] Realtime, DUNDi and regexten
We've been using DUNDi, Realtime, and regexten extensively for months now, and it's been working great since we got it running. Could you please tell us a little about the experiences you had in getting it running? Evidently there's some magic involved, otherwise so many wouldn't be struggling like they are. Also, would you mind submitting some sample configs for the community to review? Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: VOIP Bandwidth questions
I think vonage is using g723.1 which requires 6.4kbps voice bandwidth compared to g711 - 64kbps.For SIP to SIP calls, RTP doesn't necessarily goes thru the server. Only Signalling goes to the servers. This means no bandwidht usage for the provider. For SIP to PSTN calls, it has to goes thru a media gateway (owned by the provider) which may be seperate from the sip server. Vikki.On 11/2/06, Martin Joseph [EMAIL PROTECTED] wrote:On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said:snip My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself.snipFor one thing, I suppose they use codecs that compress the voice dataas much as possible.Probably g729, or ilbc or some such.Also,it's not true that all the traffic need to flow through there servers.Once the connections are setup in a well designed system, thedata could flow directly.Marty___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.16 AIX2 - SIP Attended transfer
Need help on this issue, I have a problem, when I receive a call from IAX extension (my external DID, all incoming calls from outside), I cannot transfer my calls using atxfer = *1 That is really weird because all my SIP phones can transfer calls between them. any help? dp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out Dial Interface for Asterisk
Well, as a hopefully helpful pointer: Shawn Kelley wrote: all I see there is creating calls and sending them to system phones, etc. I I assume by system phone you must mean an internal SIP phone for example, like SIP/110 or something. Couldn't you replace that channel name with Zap/phonenumber ? Moj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out Dial Interface for Asterisk
have a look at the call files or AMI originate command. This allows you to generate a call, and specify an extension and context to go to when the call has been answered. In there, you can playback any file you want to the connected caller. Also, you can then use the IVR stuff of asterisk to do the touch tone stuff. Julian Shawn Kelley wrote: Hi All, I sent this a while back but never received any replies. My deadline is fast approaching so I thought I'd throw it out there again in hope of some advice. I need the ability to automatically out-dial and play a dynamically generated message. I then need the ability for the answering party to give feedback via touch tone. I am a .Net Programmer and I have looked at the Asterisk.NET examples, but all I see there is creating calls and sending them to system phones, etc. I don't see anyway of capturing responses back from the answering party, or how to play dynamically generated messages. Does anyone know if this is possible with the Asterisk.NET interface? Or does anyone know of another way to accomplish my needs? Any advice is greatly appreciated. --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 supports now FTPS?
I just noticed that my new Polycom`s, just bought yesterday, now support FTPS for configuration downloads. I have a few questions about this. 1) Is FTPS the same as FTP with SSL? 2) Anybody has a recommendation for a good FTPS server (running on Linux of course). Ideally it would support virtual users (i.e. users not necessarily Linux users) 3) Can I make this work with a self-signed certificate? If so, anything in particular that I need to know? Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error installing asterisk, module zaptel not found
is zaptel.ko anywhere in your system? it should be in /lib/modules/`uname -r`/extra/ On 11/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: After deciding to move a semi working asterisk setup to another box, installing and recompiling asterisk, addons and zaptel, modprobe zaptel says, module not found. Following various tales of how to modify udev stuff, still get that error. lspci does show the board in the list. All the LED's on the back of the board are dark. I have a TDM400p (tdm22b). I did not actually install the board, until after asterisk and add ons were complied. Just before the steps to compile zaptel. After installing board and playing doing the udev hack dance, did recompile with same results, as stated. What could be the probem(s)? phoneman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out Dial Interface for Asterisk
On Thu, 2006-11-02 at 12:31 -0600, Shawn Kelley wrote: Hi All, I sent this a while back but never received any replies. My deadline is fast approaching so I thought I'd throw it out there again in hope of some advice. I need the ability to automatically out-dial and play a dynamically generated message. I then need the ability for the answering party to give feedback via touch tone. I am a .Net Programmer and I have looked at the Asterisk.NET examples, but all I see there is creating calls and sending them to system phones, etc. I don't see anyway of capturing responses back from the answering party, or how to play dynamically generated messages. Does anyone know if this is possible with the Asterisk.NET interface? Or does anyone know of another way to accomplish my needs? I'd drop a call file into asterisks spool dir: call file begin --- Channel: Zap/g1/phonenumber Context: playbackmenu Extension: main Priority: 1 call file end --- in extensions.conf: [playbackmenu] exten = main,1,Background(your-announcement) exten = 1,1,NoOp(User Pressed 1) exten = 2,1,NoOp(User Pressed 2) exten = 3,1,NoOp(User Pressed 3) ... Is that all you need? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
