Re: [asterisk-users] Verizon-Vonage Lawsuit
J. Oquendo wrote: Stepping back into reality for a moment, emotions aside, I side with the law if Vonage infringed on VZ's patent. Don't misconstrue what I typed, re-read it clearly. If you created something, patented it, and someone else used it without permission or compensation, if you can honestly tell this list you would sit back while said person or company made millions and do nothing, you would be lying to youself. Other companies in the industry used VZ's patents under licensing without incident so why couldn't Vonage. This isn't David versus Goliath here in fact Vonage tried to get things in order with Verizon AFTER the fact. So kudos to the judge in this case. Personally I don't like Verizon, and I'm glad I deal with them on a minimal level nowdays. However, the law is the law. This sounds like you really don't know what these legal proceedings are about. I googled this a little a week or two ago, when it appeared on engadget of all places. It appears that VZ sued Vonage for infringement of seven patents, including three for billing methods. IIRC, the billing issues were thrown out in a first round, I assume, because it's one of those how else you're gonna bill customers? deals. The one bit that did keep coming up in all my reading was that VZ apparently patented some sort of mechanism to interconnect a packet network (VOIP) to a circuit switched network (PSTN). They seemed to attempt to gain an injunction barring Vonage from using this technology or method, essentially cutting off Vonage's customers from the PSTN, and rendering Vonage service useless. Judging by how surprisingly little information was available on this, the conclusion would be that Verizon owns some patent for the VOIP/PSTN interface -- that, in turn, would mean that all digital PBX systems currently in operation infringe on this patent in some manner. (Again, this is interpolated from the small amount of information I found when searching two weeks ago). If Verizon's patent claim is indeed so broad as to prevent Vonage's PSTN interconnect, then Verizon would still have to show that the patent is non-obvious and a truly new invention (this may be difficult, because packet-based and circuit switched networks have been around for longer than Verizon has, and there is an obvious way of connecting those two); Verizon would also have to show that they had sufficient interest to develop the patent (similar to the Cisco/Apple controversy over the iPhone trademark). That latter part is hindered by the fact that Verizon didn't start going after Vonage until they had allegedly lost over a million customers to Vonage -- it appears a reciprocal action to protect VZ's business interests and not their IP. That last point could be quite a big one against VZ -- Vonage is gaining customers not because they stole Verizon's doubtful IP, but because they offer a better deal. In my area, Vonage is cheaper than a Verizon dialtone alone -- and I'd still pay for each outgoing call if I had Verizon. That said, this is going to be interesting to watch for all us asterisk users. If Vonage loses this one, VZ is going to go after the next VOIP provider... and sooner or later, anti-trust regulation will kick in. Fun world. ___ --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] Wireless Bridge for SNOM360
Due to our house layout I am unable to run some CAT5 cable from one room to another. Therefore I purchased a Belkin wireless ethernet bridge, but to my amazement it does not work :( Though, if I plug the adapter into a PC ethernet port it works a treat. Connects to the AP with a strong signal. I unplug it from the PC, and plug it back into the SNOM and the wireless connection is dropped. Any ideas ? -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 845 869 2749 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] ipv6 patch
On 22:23, Fri 06 Apr 07, Hans Witvliet wrote: On Tue, 2007-04-03 at 05:30 -0700, Jason Kim wrote: Is it exists? If not, how could they have done this: http://opensourcepbx.tmcnet.com/topics/applications/articles/5450-industry-forum-hails-successful-voip-over-asterisk-ipv6.htm (But i'll guess it's not mainstream code yet...) hw The ipv6 code lives in svn on the digium svn repos: http://svn.digium.com/view/asterisk/team/blanchet/v6/ Read http://svn.digium.com/view/asterisk/team/blanchet/v6/README-IPV6.txt?view=markup before running this code. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail from GTalk says from an unknown caller
Am Freitag, den 06.04.2007, 18:23 -0700 schrieb Am Turnip:
When I listen to voicemail from my Google Talk buddy, the envelope says,
from an unknown caller. But the voicemail correctly records the caller
ID of calls that arrive via Zapata into the same context that receives
Google Talk calls. How can I configure the voicemail to include the
caller's Google Talk identity in the envelope?
Just an idea... have you checked CALLERID(num) versus CALLERID(name)?
Try a NOOP statement to see what those contain right before jumping into
voicemail; if only one contains the proper info, that might cause the
unknown caller.
exten=123,n,NOOP(NAME: ${CALLERID(name)} NUM: ${CALLERID(num)})
or the like.
BR
Anselm
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Re: [asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Output of lspci -vvv
-- cut
02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Unknown device b560
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 22
Region 0: I/O ports at ddb8 [size=8]
Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
-- cut
Just a comment: the CHANGES file has the item fixed detection of
miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c
in -d and in -e .
Problem solved...
If anyone else is interested, here is what I changed to make it work
with a BeroNet HFC-4S rev 01 card:
Patch file:
cut
297a298,299
} else if (qoztmp-type == 0xb560) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
1584c1586
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
---
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
1638a1641
case 0xb560:
cut -
I don't know how to make it into a correct patchfile, so if someone else
knows that, it could be done and maybe placed where everybody could get it.
---
Best regards,
Henrik Woffinden
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[asterisk-users] CRM integration with Asterisk
Hi I have seen discussion about Asterisk integration with SugarCRM and Salesforce.com CRM in mailing list archives. I just want to add here that Star Outlook Dialer (Free Edition) has built in integration (through StarJunction) with SugarCRM as well as with Salesforce CRM. It is available for free download at http://www.starutilities.com/staroldialer.htm Regards, Kamran ___ --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] BeroNet HFC-4S card is now detected as only 2 ports
On Sat, Apr 07, 2007 at 12:17:03PM +0200, Henrik Woffinden wrote:
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Output of lspci -vvv
-- cut
02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Unknown device b560
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 22
Region 0: I/O ports at ddb8 [size=8]
Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
-- cut
Just a comment: the CHANGES file has the item fixed detection of
miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c
in -d and in -e .
Problem solved...
If anyone else is interested, here is what I changed to make it work
with a BeroNet HFC-4S rev 01 card:
Patch file:
cut
297a298,299
} else if (qoztmp-type == 0xb560) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
1584c1586
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
---
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
1638a1641
case 0xb560:
cut -
I don't know how to make it into a correct patchfile, so if someone else
knows that, it could be done and maybe placed where everybody could get it.
Simply use 'diff -u' instead of 'diff' .
Any idea if this can break anything?
--
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] Different devices for asterisk!!!
Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind of device user has used to register with my asterisk server? Thanx in advance -- Regards Rizwan Hisham Software Engineer ___ --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] CRM integration with Asterisk
S. A. Kamran wrote: I just want to add here that Star Outlook Dialer (Free Edition) has built in integration (through StarJunction) with SugarCRM as well as with Salesforce CRM. It is available for free download at http://www.starutilities.com/staroldialer.htm Can someone tell me how to decompress staroldialer.exe? Tried to make it executable but that doesn't work either: bash: ./staroldialer.exe: cannot execute binary file Is Star Utilities aware of the problem? ;) Happy Easter! Philipp ___ --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-users Digest, Vol 33, Issue 27
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] hox to connecte two asterisk server
hi lee,
i have changed my config in iax, and now i can call make calls without
problem.
this is a description of my architecture:
i have two offices with SIP users having the extension _037XXX in the first
office and _022XXX in the second office.
i want to connect my two offices with IAX for making possible communication
betwen my two offices users.
i have forgot some details in my iax.cong that's why i was having the
problem westerday.
thanks for hep :)
2007/4/7, Yuan LIU [EMAIL PROTECTED]:
From: hind habaoui [EMAIL PROTECTED]
Date: Fri, 6 Apr 2007 18:01:11 +
hi lee.
I see your problem with trunk iax, probably i don't have the solution but
i
don't knew if you can help me to solve mine.
Can't seem to see what the problem you have? Errors? Incorrect result?
(What is expected and what is the result?) Also, you need to clarify the
settings on two servers more clearly - your sip-calls context seems to
suggest that server B uses IAX with its users, but uses SIP to connect to
server A? The iax.conf seems to suggest SIP rather than IAX.
Yuan Liu
i want to connecte two asterisk server: server A and server B. i want
make
possible calls betwen all asterisk users.: users in server A with sip
number
022100 can phone another sip user in server B with number 037100.
this is my config:
*
iax.conf for server A:
**
register = serveur_rabat:[EMAIL PROTECTED]
[serveur_casa]
type=peer
host=dynamic
username=serveur_casa
secret=casa
disallow=all
allow=ulaw
allow=gsm
;context=sip-calls
[serveur_casa]
type=user
host=dynamic
username=serveur_casa
secret=casa
disallow=all
allow=ulaw
allow=gsm
my extension.conf
**
[sip-calls]
exten=_022[1-8]XX,1,macro(Bienvenu)
exten=_022[1-8]XX,2,SetGlobalVar(BOITE=${CDR(src)})
exten=_022[1-8]XX,3,Dial(SIP/sip-${EXTEN},${TP_MAX_APPEL})
exten=_022[1-8]XX,4,macro(BoiteVocale,${BOITE})
exten=_022[1-8]XX,5,hangUp()
;
;
;lecture des boites vocales
exten=_[1-8]XX,1,macro(lecture_boite)
exten=_[1-8]XX,2,PlaBack(vm-num-i-have)
exten=_[1-8]XX,3,HangUp()
;
;
; on donne accès au service du standard
exten=022999,1,Wait(5)
exten=022999,2,Dial(${TEL1},,t)
exten=022999,3,HangUp()
include=parkedcalls
include=iax-calls
[iax-calls]
exten=_037XXX,1,macro(Bienvenu)
exten=_037XXX,2,Dial(IAX2/serveur_rabat/${EXTEN},${TP_MAX_APPEL},r)
exten=_037XXX,3,macro(BoiteVocale)
**
file config for the second server looks like the server's A file.
thank you in advance
hind
GTR 2007
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--
hind habaoui
Ingenieur Réseaux et Télecommunication
2007
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Re: [asterisk-users] Different devices for asterisk!!!
If your device is connecting to asterisk as a peer or a friend, the the sip show peers user will show a user agent field. For example I have a linksys phone in my home office that connects as a friend and so if I type sip show peer 1000 - the phones username. I get the following entry Useragent: Linksys/SPA942-5.1.5 Which tells me brand, model and firmware. This field stores the user agent string sent by the device, so each manufacture and even device may give different information. Here is another just to show some of the detail. Useragent: Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 On 4/7/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind of device user has used to register with my asterisk server? Thanx in advance -- Regards Rizwan Hisham Software Engineer ___ --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 -- Bruce Reeves Nortex Networks ___ --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] CRM integration with Asterisk
Will an Outlook dialer run on Linux ?? Works fine on XP, might check your OS :) On 4/7/07, Philipp Kempgen [EMAIL PROTECTED] wrote: S. A. Kamran wrote: I just want to add here that Star Outlook Dialer (Free Edition) has built in integration (through StarJunction) with SugarCRM as well as with Salesforce CRM. It is available for free download at http://www.starutilities.com/staroldialer.htm Can someone tell me how to decompress staroldialer.exe? Tried to make it executable but that doesn't work either: bash: ./staroldialer.exe: cannot execute binary file Is Star Utilities aware of the problem? ;) Happy Easter! Philipp ___ --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 -- Bruce Reeves Nortex Networks ___ --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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:53 PM: On Fri, Apr 06, 2007 at 12:37:47PM -0700, Lee Howard wrote: It's usually built and left in the zaptel source directory where you extracted and built zaptel. If it doesn't get built for you from zttest.c then check the Makefile that it has zttest in BINS like this from mine: BINS=ztcfg torisatool makefw ztmonitor ztspeed zttest fxotune A simpler method: make -C /path/to/zaptel/source zttest which will generate zttest for you. You can use zttest from whatever version of Zaptel you might have. Repeated warnings about clock skew detected. joe a. ___ --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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. joe a. ___ --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] Different devices for asterisk!!!
Rizwan Hisham wrote: Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind of device user has used to register with my asterisk server? Have a look at these functions (core show function ...): SIPPEER(useragent), SIPCHANINFO(useragent), SIP_HEADER(User-Agent) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] FAX thru TDM400p
On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. The second one must have given some output. If the first one gives no errors, you probably have at least a semi-functioning timing source. -- 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
Re: [asterisk-users] Open Source VoIP client (on a webpage)
On 6 Apr 2007, at 21:21, Matthew Rubenstein wrote: On Fri, 2007-04-06 at 12:00 -0700, [EMAIL PROTECTED] wrote: Date: Fri, 6 Apr 2007 16:13:29 +0100 From: Tim Panton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage) To: Jason Wolfe [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed On 6 Apr 2007, at 00:59, Jason Wolfe wrote: I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Our SDK isn't open source, but it is an IAX applet - javascript/DHTML friendly and lightweight. Is that applet available unbundled from the rest of your software and service package? At a flat (ie not per-instance) price? Yes, it is available separately and the price is per-server. Anyone interested should contact me off-list as this is getting dangerously commercial! Tim. Tim Panton www.mexuar.net www.westhawk.co.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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM: On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. The second one must have given some output. If the first one gives no errors, you probably have at least a semi-functioning timing source. Yes, sorry, left out time, output follows: foo:~ # time head -c 8192 /dev/zap/pseudo real0m1.029s user0m0.000s sys 0m0.004s foo:~ # joe a ___ --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] Different devices for asterisk!!!
Thanx a lot...it does it..i was in need of it very badly On 4/7/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind of device user has used to register with my asterisk server? Have a look at these functions (core show function ...): SIPPEER(useragent), SIPCHANINFO(useragent), SIP_HEADER(User-Agent) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] Verizon-Vonage Lawsuit
On 7 Apr 2007, at 07:02, Jay Milk wrote: If Verizon's patent claim is indeed so broad as to prevent Vonage's PSTN interconnect, then Verizon would still have to show that the patent is non-obvious and a truly new invention Sadly not - if they have a patent granted, then the onus of proof is the other way around, Vonage and the rest of us have to prove it is obvious or cite prior-art. That last point could be quite a big one against VZ -- Vonage is gaining customers not because they stole Verizon's doubtful IP, but because they offer a better deal. In my area, Vonage is cheaper than a Verizon dialtone alone -- and I'd still pay for each outgoing call if I had Verizon. That said, this is going to be interesting to watch for all us asterisk users. If Vonage loses this one, VZ is going to go after the next VOIP provider... and sooner or later, anti-trust regulation will kick in. Fun world. If I were Vonage I'd have a delegation in HongKong now, moving all my Telco interconnects to somewhere where the US patent system is treated with the contempt it is starting to earn. Tim Panton www.mexuar.net www.westhawk.co.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] Verizon-Vonage Lawsuit
On Sat, 07 Apr 2007, Jay Milk wrote: (my comments inline) This sounds like you really don't know what these legal proceedings are about. I googled this a little a week or two ago, when it appeared on engadget of all places. It appears that VZ sued Vonage for infringement of seven patents, including three for billing methods. IIRC, the billing issues were thrown out in a first round, I assume, because it's one of those how else you're gonna bill customers? deals. The jury found that three of five disputed patents were infringed and all are valid, while rejecting Verizon's claim that the infringement was willful. The patents cover a method of translating calls between the Internet and standard phones, call-waiting features and wireless handsets. http://www.bloomberg.com/apps/news?pid=20601087sid=aj9gCo9Pr58grefer=home No one is mentioning which of the three are the main cause for the rulings so you're ASSuming its largely based on on the interconnection of PSTN's. Take that out of the mix for a moment, and you still have two violations so your argument makes little sense. The one bit that did keep coming up in all my reading was that VZ apparently patented some sort of mechanism to interconnect a packet network (VOIP) to a circuit switched network (PSTN). They seemed to attempt to gain an injunction barring Vonage from using this technology or method, essentially cutting off Vonage's customers from the PSTN, and rendering Vonage service useless. This one bit that keeps coming up is obviously under the microscope from every VoIP company in existence including my employer, our vendors and anyone with some ties to VoIP no matter how great or how small. Take this out of the focus for a moment, and you STILL have two other violations. Judging by how surprisingly little information was available on this, the conclusion would be that Verizon owns some patent for the VOIP/PSTN interface -- that, in turn, would mean that all digital PBX systems currently in operation infringe on this patent in some manner. (Again, this is interpolated from the small amount of information I found when searching two weeks ago). Should I reiterate the need to take this one infringement out of the mix and focus on the other two? If Verizon's patent claim is indeed so broad as to prevent Vonage's PSTN interconnect, then Verizon would still have to show that the patent is non-obvious and a truly new invention (this may be difficult, because packet-based and circuit switched networks have been around for longer than Verizon has, and there is an obvious way of connecting those two); Verizon would also have to show that they had sufficient interest to develop the patent (similar to the Cisco/Apple controversy over the iPhone trademark). That latter part is hindered by the fact that Verizon didn't start going after Vonage until they had allegedly lost over a million customers to Vonage -- it appears a reciprocal action to protect VZ's business interests and not their IP. Non-obvious? You say tomaytoe I say tomahtoe. A patent is a patent is a patent. Blame the government and legal shmoes for their bastardization of words, intents and ambiguities when dealing with these matters. No matter how you want to cut this though, there are 3 infringements and this is what I look at. That last point could be quite a big one against VZ -- Vonage is gaining customers not because they stole Verizon's doubtful IP, but because they offer a better deal. In my area, Vonage is cheaper than a Verizon dialtone alone -- and I'd still pay for each outgoing call if I had Verizon. Its obvious VZ went after Vonage due to Vonage slowly taking away VZ's customer base, but what are you going to do, VZ holds the cards no matter how much you dislike it, no matter how much you want to play the butchery game on patents, intents, and what your notion of what is going on is. That said, this is going to be interesting to watch for all us asterisk users. If Vonage loses this one, VZ is going to go after the next VOIP provider... and sooner or later, anti-trust regulation will kick in. As stated previously, I think anyone and everyone in the world of VoIP is watching. As for your dire apocalyptic prediction of anti-trust regulation, too many players would bury VZ. Cisco, ATT, 3Com, I could name hundreds that own patents that would play the same game of Patent Ambiguities to render any arguments as worthless should a case go to court on VZ's merits. Strangely and for no other reason than to understand the context of it all, I will dig up the court transcripts (since its all public information) to see what patent numbers were infringed and try to understand it for my own sake. But to summarize, one's own interpretations of these events are that and that alone: One's own interpretation. You say: But everyone interconnects to the PSTN with FXS/FXO, foo bar foo. I say: Well there are 3 infringments. Take away the
Re: [asterisk-users] Re: G729 'disappears' randomly
On Fri, 2007-03-23 at 03:11 +0530, Rajeev Natarajan wrote: It happened again this evening and when I checked the host-id in /var/log/asterisk/messages the time when it did not register, it showed a host-id Mar 22 18:14:48 VERBOSE[2586] logger.c: == G.729 Host-ID: 90:23:3a:b7:dc:46:88:fc:cf:bb:78:a2:b8:00:75:97:34:xx:xx:xx (removing the last 6 for security) and it did not load the g729 Mar 22 18:43:18 VERBOSE[2580] logger.c: == G.729 Host-ID: 05:e5:4b:6c:0d:8b:66:fd:7a:b5:8e:a6:23:73:0b:b1:66:xx:xx:xx WORKS perfectly Any clues on why the host-id changes? IDEA: I also notice that sometimes eth0,eth1 and eth2 (Yes: i have three network interfaces) interchange on reboot. Are they related? Quite possibly, the registration program for that codec will bind to eth0 and use it as the host ID, if you change ethernetcards or re-number interfaces you will need to re-register the codec. As for the re-ordering of your network cards I would suggest you look into running udev with some rules to keep the order of the cards consistent over reboots. -- Nikolai Lusan # # # Weblog: http://lusan.id.au/~nikolai/blog # Website:http://lusan.id.au/~nikolai # # ___ --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] HPEC audio clipping
Eric ManxPower Wieling wrote: I am experiencing the same thing. I assumed that I just didn't have a fast enough CPU (2.4 Ghz Celeron Ghz, also tried on a 1.8 Ghz Pentium 4). I am using a T400P with an Adtran TA750 Channel Bank rather than the Digium analog cards. I'm not doing any VoIP on this system, strictly analog and I get echo on calls. I will be working on improving Zaptel next week to allow us to do proper debugging of this issue with the provider of the HPEC code; stay tuned for updates, and those of you are experiencing this problem please ensure that Digium Support is aware of you and can contact you when we have the new Zaptel code available for testing. Thanks! ___ --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] FAX thru TDM400p
On Sat, Apr 07, 2007 at 08:42:59AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM: On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. The second one must have given some output. If the first one gives no errors, you probably have at least a semi-functioning timing source. Yes, sorry, left out time, output follows: foo:~ # time head -c 8192 /dev/zap/pseudo real0m1.029s Approximately 1 second. user0m0.000s sys 0m0.004s foo:~ # So your timing source is basically working. -- 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] Verizon Vonage 101
I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant messenger US 7,054,308 B1 Method and apparatus for estimating the call grade of service and offered traffic for voice over internet protocol calls at a PSTN-IP network gateway According to Google: They've listed 118 patents assigned to Verizon Results 1 - 10 of about 118 from uspto.gov for +assigned to +verizon One dealing with PSTN Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon +pstn Two matches dealing with VoIP but only one is a patent. And that is related to the above search Results 1 - 2 of 2 from uspto.gov for +assigned to +verizon voip Three matches dealing with Voice and IP but only one is a patent. And that too is related to the above search Results 1 - 3 of 3 from uspto.gov for +assigned to +verizon voice IP Nine matches dealing with telephone and IP but only one is a patent. And that too is related to the above search Results 1 - 9 of 9 from uspto.gov for +assigned to +verizon telephone IP. One patent related to voicemail Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon voicemail My thoughts, the voicemail one is broad, and could be circumvented easily. If I were a juror, I would laugh but an infringment is an infringement is an infringement. I would make Vonage stop using the technology. The VoIP patent however is a bit more detailed, and although it can be construed as broad, that too would make me side with Verizon, but not to the degree of shutting down Vonage. On the flip side of things, Vonage is no stranger to infringing on patents. Of course, turnabout is fair play, and Klausner Technologies Inc. filed suit against Vonage for infringing its patent number 5,572,576, which concerns the retrieval of VoIP voicemail on a cell phone or handheld device. http://www.cedmagazine.com/article/CA6351074.html -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --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] Verizon Vonage 101
J. Oquendo wrote: So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... I am looking at Verizon Services Corp. v. Vonage Holdings Corp., Slip Copy, 2007 WL 528749, E.D.Va.,2007, which is the result of the Markman hearing. That is the court interpreting the claim language, and here are the patents discussed: 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 I do not know which of these Vonage was found to have infringed. Patrick ___ --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] Verizon Vonage 101
They could be suing for patents completely unrelated to VoIP as a technology. There are cases on the book where people like Katz have been running around suing contact center operators because he has a patent on authenticating yourself to a phone service using a pin number and using that information to access account records. Contact center operators get hit by that crap all the time and Katz is a master of letting them operate for years before he comes banging on their door looking for a check. With VoIP, the news is always talking about subscriber numbers and industry growth projection. This is like putting blood in the water. It was only a matter of time until the guys like Verizon, Katz, etc start pulling their polished patent infringement weapons out on the naive VoIP operators. This is just how business is done in America. You want to see a whacked out patent? Take a look at Katz's patent on Methods and apparatus for intelligent selection of goods and services in telephonic and Electronic Commerce. This guy has patents on paying by phone or web for products using a credit card. There are 267 different methods this clown has patented and he actively sues companies for using these methods in common business channels. http://www.google.com/patents?vid=USPAT6055513id=VGQEEBAJ At my former employer, when VoIP was starting to get hot - they had me apply for a patent on IP contact center technologies which took a lot of what Katz had produced and expanded it to VoIP. We did this for purely defensive disclosure purposes, but there are clowns out there who do this to generate revenue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo Sent: Saturday, April 07, 2007 11:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Verizon Vonage 101 I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant messenger US 7,054,308 B1 Method and apparatus for estimating the call grade of service and offered traffic for voice over internet protocol calls at a PSTN-IP network gateway According to Google: They've listed 118 patents assigned to Verizon Results 1 - 10 of about 118 from uspto.gov for +assigned to +verizon One dealing with PSTN Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon +pstn Two matches dealing with VoIP but only one is a patent. And that is related to the above search Results 1 - 2 of 2 from uspto.gov for +assigned to +verizon voip Three matches dealing with Voice and IP but only one is a patent. And that too is related to the above search Results 1 - 3 of 3 from uspto.gov for +assigned to +verizon voice IP Nine matches dealing with telephone and IP but only one is a patent. And that too is related to the above search Results 1 - 9 of 9 from uspto.gov for +assigned to +verizon telephone IP. One patent related to voicemail Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon voicemail My thoughts, the voicemail one is broad, and could be circumvented easily. If I were a juror, I would laugh but an infringment is an infringement is an infringement. I would make Vonage stop using the technology. The VoIP patent however is a bit more detailed, and although it can be construed as broad, that too would make me side with Verizon, but not to the degree of shutting down Vonage. On the flip side of things, Vonage is no stranger to infringing on patents. Of course, turnabout is fair play, and Klausner Technologies Inc. filed suit against Vonage for infringing its patent number 5,572,576, which concerns the retrieval of VoIP voicemail on a cell phone or handheld device. http://www.cedmagazine.com/article/CA6351074.html -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --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:
Re: [asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports
Tzafrir Cohen wrote:
On Sat, Apr 07, 2007 at 12:17:03PM +0200, Henrik Woffinden wrote:
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Output of lspci -vvv
-- cut
02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Unknown device b560
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 22
Region 0: I/O ports at ddb8 [size=8]
Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
-- cut
Just a comment: the CHANGES file has the item fixed detection of
miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c
in -d and in -e .
Problem solved...
If anyone else is interested, here is what I changed to make it work
with a BeroNet HFC-4S rev 01 card:
Patch file:
cut
297a298,299
} else if (qoztmp-type == 0xb560) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
1584c1586
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
---
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
1638a1641
case 0xb560:
cut -
I don't know how to make it into a correct patchfile, so if someone else
knows that, it could be done and maybe placed where everybody could get it.
Simply use 'diff -u' instead of 'diff' .
Any idea if this can break anything?
Thanks for the tip. Here's a proper patch-file:
[EMAIL PROTECTED] qozap]# diff -urN qozap.c.orig qozap.c
--- qozap.c.orig2007-04-07 11:49:36.0 +0200
+++ qozap.c 2007-04-07 12:00:52.0 +0200
@@ -295,6 +295,8 @@
} else {
if (qoztmp-type == 0x08b4) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x0);
+ } else if (qoztmp-type == 0xb560) {
+ qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
} else if (qoztmp-type == 0xb550) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
} else if (qoztmp-type == 0xb556) {
@@ -1581,7 +1583,7 @@
if (pcidid == PCI_DEVICE_ID_CCD_M) {
qoztmp-stports = 8;
} else {
- if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
+ if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
qoztmp-stports = 4;
} else {
qoztmp-stports = 2;
@@ -1636,6 +1638,7 @@
if (pcidid == PCI_DEVICE_ID_CCD_M4) {
switch (tmp-subsystem_device) {
case 0x08b4:
+ case 0xb560:
if (ports == -1) ports = 0; /* assume TE mode if
no ports param */
printk(KERN_INFO
qozap: CologneChip HFC-4S evaluation board
configured at io port %#x IRQ %d HZ %d\n,
I don't think it will break anything, as I haven't changed any logic,
just added the device 0xb560 to go to the proper options. But I can't
give u a 100% guarantee as I have no experience in programming these cards.
Shall I upload the patch to somewhere?
Happy Easter.
Best regards,
Henrik Woffinden
___
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 28
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] FAX thru TDM400p
. . . So your timing source is basically working. -- Tzafrir Cohen And this means . . . any FAX-ing issues must be due to other problems? joe a. ___ --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] Remastering asterisk
khaled, I successfull remaster a router CD, and lamppix CD both using knoppix or debian as the base, I am pretty sure... try http://lamppix.tinowagner.com/ http://www.wifi.com.ar/english/cdrouter/ daveC Khaled Chehab wrote: Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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: 7.5.446 / Virus Database: 268.18.24/741 - Release Date: 03/31/2007 08:54 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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] Cannot compile 1.4.2 on Slackware 7
Hi All, I am trying to upgrade an old Asterisk installation to 1.4.2 (it's currently running CVS-08/02/04-15:15:26) but have hit a couple of problems. The first was easily fixed. I got storage size of sin isn't known errors whilst compiling streamplayer.c, but after seeing http://bugs.digium.com/view.php?id=4908#32012 I manually added #include netinet/in.h to the top of streamplayer.c and this compiled OK. (Should I file a bug for this?) However I'm currently stuck here: /usr/src/asterisk-1.4.2# make [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o: In function `ael_yylex': /usr/src/asterisk-1.4.2/include/asterisk/strings.h:167: undefined reference to `__builtin_expect' ast_expr2f.o: In function `ast_expr': /usr/src/asterisk-1.4.2/include/asterisk/strings.h:167: undefined reference to `__builtin_expect' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error 1 make: *** [utils] Error 2 I don't know enough about the internals of Asterisk to know what I should be looking for - is the configure script perhaps not checking for a required library? The machine is quite old, so it is possible I need to upgrade/add something - but what? # uname -a Linux myserver 2.4.25 #5 Wed Jan 26 18:20:35 GMT 2005 i686 unknown # cat /etc/slackware-version 7.0.0 If anyone can point me in the right direction, I'd appreciate it! Cheers, Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein ___ --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] Cannot compile 1.4.2 on Slackware 7
Jonathan Hunter wrote: /usr/src/asterisk-1.4.2# make [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o: In function `ael_yylex': /usr/src/asterisk-1.4.2/include/asterisk/strings.h:167: undefined reference to `__builtin_expect' ast_expr2f.o: In function `ast_expr': /usr/src/asterisk-1.4.2/include/asterisk/strings.h:167: undefined reference to `__builtin_expect' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error 1 make: *** [utils] Error 2 Given the age of your distribution, I suspect your compiler is too old. What version of GCC are you using? ___ --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] Follow Me and Transferring Calls
When my follow me or transferred calls come out to me they appear as if they are coming from one of my lines rather than showing the caller id of the initial caller. I believe there is a way to make it forward the initial caller id information isn't there? Is it just that my voip provider is not allowing me to do this and if so does anyone have any suggestions on some voip providers that will let me provide the caller id info? Thanks, Andy ___ --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] Remastering asterisk
Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Have a look at Kickstart hth ___ --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] Cannot compile 1.4.2 on Slackware 7
Hi, On 07/04/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Given the age of your distribution, I suspect your compiler is too old. What version of GCC are you using? I haven't compiled any Asterisk version on this machine since 2004, so you could well be right on that front. # gcc --version 2.95.3 Is __builtin_expect part of gcc, then, rather than an external library? (i.e. would I need to upgrade gcc in this instance) Thank you! Jonathan ___ --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] Cannot compile 1.4.2 on Slackware 7
Jonathan Hunter wrote: Is __builtin_expect part of gcc, then, rather than an external library? (i.e. would I need to upgrade gcc in this instance) Yes, it is a GCC extension added in GCC 3.x, I believe. ___ --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] make Zaptel 1.2.16 'struct inode' has no member named 'u'.
Zaptel has no direct code relationship with Asterisk. Your error is because zaptel is trying to use a member no longer exists in newer kernels. However you are using fedora, and fedora included that change in older kernel. I found this in xpp/xbus-core.c /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use such a kernel, Change the * following test from 2,6,19 to 2,6,18. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif The following resolved this issue: vi xpp/xbus-core.c Change code as follows: #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,18) make clean make Thanks Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/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] Verizon Vonage 101
Here's links and descriptions for the 8 you listed. All Bell Atlantic, GTE, or Verizon. This should make your research a bit easier. 6,137,869 Network session management http://www.google.com/patents?vid=USPAT6137869id=yl4GEBAJdq=6137869 Patent number: 6137869 Filing date: Sep 16, 1997 Issue date: Oct 24, 2000 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Network Services, Inc. Primary Examiner: Rexford N Barnie 6,430,275 Enhanced signaling for terminating resource http://www.google.com/patents?vid=USPAT6430275id=NmwLEBAJdq=6,430,275 Patent number: 6430275 Filing date: Jul 28, 1999 Issue date: Aug 6, 2002 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Services Network, Inc. Primary Examiner: Curtis Kuntz Secondary Examiner: Rexford M Barnie 6,104,711 (The famous: We think we invented ENUM patent) Enhanced internet domain name server http://www.google.com/patents?vid=USPAT6104711id=J18EEBAJdq=6,104,711 Patent number: 6104711 Filing date: Mar 6, 1997 Issue date: Aug 15, 2000 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,282,574 Method, server and telecommunications system for name translation on a conditional basis and/orto a telephone number http://www.google.com/patents?vid=USPAT6282574id=46sIEBAJdq=6,282,574 Patent number: 6282574 Filing date: Feb 24, 2000 Issue date: Aug 28, 2001 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,359,880 Public wireless/cordless internet gateway http://www.google.com/patents?vid=USPAT6359880id=tP4KEBAJdq=6,359,880 Patent number: 6359880 Filing date: Jul 30, 1999 Issue date: Mar 19, 2002 Inventors: James E. Curry, Robert D. Farris Primary Examiner: Wellington Chin Secondary Examiner: Steven Nguyen 6,128,304 (We think we own presence too...) Network presence for a communications system operating over a computer network http://www.google.com/patents?vid=USPAT6128304id=BnkGEBAJdq=6,128,304 Patent number: 6128304 Filing date: Oct 23, 1998 Issue date: Oct 3, 2000 Inventors: Steven E. Gardell, Barbara Mayne Kelly, Rajiv Bhatnagar, Thomas James Antell, Israel B. Zibman Assignee: GTE Laboratories Incorporated Primary Examiner: Frank Duong 6,298,062 (aka. Accepting H.323 phone calls/faxes from a computer network and terminating them on the PSTN) System providing integrated services over a computer network http://www.google.com/patents?vid=USPAT6298062id=jp4IEBAJdq=6,298,062 Patent number: 6298062 Filing date: Oct 23, 1998 Issue date: Oct 2, 2001 Inventors: Steven E. Gardell, Israel B. Zibman Assignee: Verizon Laboratories Inc. Primary Examiner: Shick Hom -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Buller Sent: Saturday, April 07, 2007 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon Vonage 101 J. Oquendo wrote: So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... I am looking at Verizon Services Corp. v. Vonage Holdings Corp., Slip Copy, 2007 WL 528749, E.D.Va.,2007, which is the result of the Markman hearing. That is the court interpreting the claim language, and here are the patents discussed: 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 I do not know which of these Vonage was found to have infringed. Patrick ___ --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] Cannot compile 1.4.2 on Slackware 7
Jonathan Hunter wrote: The machine is quite old, so it is possible I need to upgrade/add something - but what? # uname -a Linux myserver 2.4.25 #5 Wed Jan 26 18:20:35 GMT 2005 i686 unknown # cat /etc/slackware-version 7.0.0 Slackware 7! Upgrade to something from this century! Seriously, 7.0 is so old that you should not be using it. It's not supported with security updates and uses older versions of glibc (which may be part of your problem). If anyone can point me in the right direction, I'd appreciate it! Slackware 11.0 was recently released. Move to that. There isn't anything special that would make it run less fast than 7.0 does on your hardware. You'll also have the option of installing a 2.6x kernel which will bring you to the 21 century. I used slackware back to version 3.0. It's a very clean distro. If this is a dedicated Asterisk box, consider looking at Astlinux. We will be releasing 0.4.5 in the next week or so with some major upgrades/improvements. For now, you can grab the release candidate images from here: http://www.djhsolutions.com/astlinux Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Re: asterisk-users Digest, Vol 33, Issue 29
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] remote SIP, no audio, or one way audio.
joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office|sip phone |-ditto Hope that is intelligible. joe a ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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] Balancing the Hybrid
Michael Boers wrote: Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I am in the US, so opermode should not be ok at default settings. I just recently got fxotune working on my system. The version that comes with zaptel 1.2.16 would simply hang. I am using the 1.4 fxotune now with the 1.2.16 driver. That has reduced the echo coefficient from 35% to 8%. We will see how that does. It should be lower still -- under 5%. Do you have other analog extensions hanging off the line between the demarcation and the Asterisk server? Do you have loose wires hanging off the line between the demarcation and the Asterisk server? How clean are the terminations? You can go a long way by cleaning up a substandard wiring job. -Stephen- ___ --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] Verizon-Vonage Lawsuit
Jay Milk wrote: That last point could be quite a big one against VZ -- Vonage is gaining customers not because they stole Verizon's doubtful IP, but because they offer a better deal. In my area, Vonage is cheaper than a Verizon dialtone alone -- and I'd still pay for each outgoing call if I had Verizon. That said, this is going to be interesting to watch for all us asterisk users. If Vonage loses this one, VZ is going to go after the next VOIP provider... and sooner or later, anti-trust regulation will kick in. You hope. The last twenty years in the United States has seen a steady erosion in anti-trust legislation. As for Vonage, the honeymoon is over in these parts. I know a few enthusiastic early adopters who are fed up with the poor call quality; one out of three times they call me I hear totally unintelligible buzzing or warbling. They're switching back to analog lines now. (There's a business I know that's on Vonage, but I haven't spoken to them for a few months, so I don't know how they're doing.) The lack of network-wide QoS will ultimately prevent VoIP from usurping the PSTN. -Stephen- ___ --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] Prompt for a PIN number to make long distance call?
I need to authenticate users to make long distance calls. Basically,when the user dials a long distance dialplan pattern, I want to prompt for his pin and look it up against a table of pins:usernames in a file. If it exists, I'll use the username in the cdr accountcode and permit the call. Authenticate() looked very promising nut I couldn't get the ma options to work. Any help is appreciated. Honestly, I'm not even sure how to read an external file and parse it from asterisk. ___ --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] Prompt for a PIN number to make long distance call?
Hi J French, try with DISA ;p http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA -- Humberto Figuera - Using Linux 2.6.20 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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] Verizon-Vonage Lawsuit
Stephen Bosch wrote: Jay Milk wrote: That last point could be quite a big one against VZ -- Vonage is gaining customers not because they stole Verizon's doubtful IP, but because they offer a better deal. In my area, Vonage is cheaper than a Verizon dialtone alone -- and I'd still pay for each outgoing call if I had Verizon. That said, this is going to be interesting to watch for all us asterisk users. If Vonage loses this one, VZ is going to go after the next VOIP provider... and sooner or later, anti-trust regulation will kick in. You hope. The last twenty years in the United States has seen a steady erosion in anti-trust legislation. As for Vonage, the honeymoon is over in these parts. I know a few enthusiastic early adopters who are fed up with the poor call quality; one out of three times they call me I hear totally unintelligible buzzing or warbling. They're switching back to analog lines now. (There's a business I know that's on Vonage, but I haven't spoken to them for a few months, so I don't know how they're doing.) The lack of network-wide QoS will ultimately prevent VoIP from usurping the PSTN. -Stephen- From my experience with Vonage, the problem is the PSTN interface in certain locations Vonage to Vonage is quite good, not quite as good as Asterisk to Asterisk, but certainly quite acceptable. Where it falls down is in certain locations back to the PSTN This seems true also with others, such as Stanaphone and sipphone as well We also have a pretty good ISP with Commiecast John Novack ___ --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] Vonage fraud controls
Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in Asterisk and had your account terminated by Vonage? I'm curious as to whether they will stop your service if you push too many calls through their ATA in a specific period of time. Thanks in advance for the info, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --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] What is wink, prewink, start and preflash time
Hi, Someone knows and can explain what is wink, prewink, start and preflash time? Sds, Gustavo From: Gustavo Cordeiro [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX thru TDM400p Date: Fri, 06 Apr 2007 16:35:14 -0300 You can find zttest.c in the zaptel source package. Download it from the asterisk.org. Sds, Gustavo From: Joe Acquisto [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX thru TDM400p Date: Fri, 06 Apr 2007 15:19:16 -0400 Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM: Joe Acquisto wrote: AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but What it always fails. What does zttest say about your zap card configuration/installation? If it's not always 99.98% or better then it's due to hardware resource constriction and you need to escalate the zaptel card's priority on the hardware (like putting it at a lower IRQ). Lee. zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. joe a. ___ --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 _ Descubra como mandar Torpedos do Messenger para o celular! http://mobile.msn.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 _ Mande torpedos SMS do seu messenger para o celular dos seus amigos http://mobile.msn.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] Audio Gain Settings
On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote: snip You are right. zapata.conf is not used in IAX connections. My reading has led me to believe that manipulating gain on an IP PBX is neither necessary nor practical in VoIP channels, so Asterisk does not devise such settings. Thanks Yuan. I beg to differ with the developers if there really is no amplitude control on IP channels. I have an application where I am studying the spectrum of recorded voice. When I call into my Asterisk box I have to hold the phone away and speak softly to avoid clipping the recorded waveform - clipped waveforms play havoc with the spectrum. I guess it is time to study the source code (ugh!). Best regards, -- Bob Smither [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
RE: [asterisk-users] Vonage fraud controls
There's no way for them to tell if you have asterisk on the fxo port BUT they will terminate your account if you hook it up as the outbound for an office pumping call after call through it. What did you expect? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Salvatore Giudice Sent: Saturday, 7 April 2007 8:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Vonage fraud controls Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in Asterisk and had your account terminated by Vonage? I'm curious as to whether they will stop your service if you push too many calls through their ATA in a specific period of time. Thanks in advance for the info, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --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] Linux IAX client to zaptel voice quality issue
Hi, I've had a hard time understanding what was going on in a new * setup. The deployment has a * box running on dual xeon RH9 stock 2.4.20-8 and some different versions of asterisk (1.2.10/1.2.16) + libpri + zaptel + wanpipe. Short version: audio from iaxclient clients is fine from windows but poor from linux when going to zaptel. E.g. Iaxcomm running on windows works fine, but the same version of it running on linux has poor quality when going IAX-Zap-PRI (sounds like samples are being dropped ~4 times/second). Return audio is clear. This happens no matter what version of linux we tried. Finally a way of dealing with it was setting jitterbuffers in zapata.conf up to 10 from default 4. All audio is fine if zaptel is not used (i.e. iax to iax or iax to sip calls sound great). I don't really understand what the issue is, but one possibility is that somehow iaxclient is not being able to set correct timestamps in linux, and that in turn triggers some kind of correction in zaptel. FTR, clients are using DSP-400 headsets (although problem can be reproduced with other sound devices) and the problem also happens with analog TDM zaptel interfaces. Keywords: linux client iax zaptel voice audio quality -- Carlos G Mendioroz [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
Re: [asterisk-users] Verizon-Vonage Lawsuit
If I were Vonage I'd have a delegation in HongKong now, moving all my Telco interconnects to somewhere where the US patent system is treated with the contempt it is starting to earn. ROTFL. The US patent system is treated with contempt in Hong Kong? You have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in Hong Kong. In any case, even if you can do things with penalty in Hong Kong against the stupid patents that are regularly being accepted by the USPTO, why would Hong Kong be a good choice? Bandwidth costs more than they do in the US and I doubt Vonage had their own trans pacific fibre to stick their telco connects here without having to pay reach or whoever loads of cash for their HK-US links. ___ --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 return dialstatus of second (sub) call
On Thu, Apr 05, 2007 at 02:06:53AM -0500, Jonathan Rivera wrote:
Hello all
I have this problem, i need a way to balance my trunks which are SIP
peers, when a SIP peer is busy then send the call for another peer and
so until i can send away the call, i think i can do it with queues.
Ok this is the scenario:
In extensions.conf
[balance]
exten = _,1,NoOp(Call to: ${EXTEN})
exten = _,2,Answer()
exten = _,3,SetVar(_ORGEXTEN=${EXTEN})
exten = _,4,SetVar(_ORGUNIQUEID=${UNIQUEID})
exten = _,5,Set(CDR(userfield)=${ORGUNIQUEID})
exten = _,6,Queue(qtest,r)
exten = _,7,Hangup()
I have a queue with 100 members which are local channels
In queues.conf
[qtest]
strategy=random
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
I had a similar problem of returning state to the queue manager to check the
call state.
You might want to try something like:
exten = check,1,ChanIsAvail(Local/[EMAIL PROTECTED],js);
exten = check,102,Goto(busy,1);
exten = busy,1,Busy();
Obviously you could replace this with a macro/DB lookup to avoid having lots
of repeated entries in the dial plan.
Busy() should return busy to the queue application if the Local
channel is in use, causing it to skip to the next entry in the queue.
After having a nightmare with chan_agent not working properly, I implemented
a modified (for 1.2.x) version of:
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
and stopped using AgentCallBackLogin(), which digium it appears have
deprecated anyway in 1.4.x
Agents without agent channel is a bit of a hack, but it works better than
chan_agent in my case.
This caused various other problems, notably that hints do not seem to work
with Local/ channels, it shows them as always available. I have not found
a workaround to this as yet. Any attempts I have made to dynamically update
hints in the dialplan from asterisk CLI (add extension .) seems to
cause it to core dump in my case. Other than that, it works quite well.
Rob
--
Robert Lister - London Internet Exchange- http://www.linx.net/
[EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE
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Re: [asterisk-users] Follow Me and Transferring Calls
Andy, Who is your VOIP provider ? Off the top of my head you can try Teliax (www.teliax.com), VoipJet (www.voipjetcom) and Nufone (http://nufone.net/). All of these providers let you set your own CID. Dovid - Original Message - From: Andy Gee To: asterisk-users@lists.digium.com Sent: Saturday, April 07, 2007 8:35 PM Subject: [asterisk-users] Follow Me and Transferring Calls When my follow me or transferred calls come out to me they appear as if they are coming from one of my lines rather than showing the caller id of the initial caller. I believe there is a way to make it forward the initial caller id information isn't there? Is it just that my voip provider is not allowing me to do this and if so does anyone have any suggestions on some voip providers that will let me provide the caller id info? Thanks, Andy ___ --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] Is it possible to Voicemail menus (not just audiofiles) ?
Olivier, You have two options. 1) Change the source code. 2) Pay a coder to give you the options. Also this mat be the lack of sleep talking but from what I remember there was talk about this before. Search the archives. Dovid - Original Message - From: Olivier To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, April 06, 2007 5:44 PM Subject: [asterisk-users] Is it possible to Voicemail menus (not just audiofiles) ? Hello, From dialplan perspective, it seems you can't tailor your voicemail behaviour to specific needs (dial 1 for old message listening, ...). Can anyone recommend a way to do it ? Does it make sense to write your own IVR and store audio files somewhere ? Best regards -- ___ --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] Verizon-Vonage Lawsuit
snip ROTFL. The US patent system is treated with contempt in Hong Kong? You have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in Hong Kong. /snip I find this hard to believe since most hack attempts to my box's originate from IP's in China. ___ --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] Prompt for a PIN number to make long distance call?
If you want a specific CID to show up it seems that your only options are to A) Write an AGI. B) If you don't have many users that can dial international you can use a series of GotoIf statements. C) You can use the Asterisk DB. Dovid - Original Message - From: J French To: asterisk-users@lists.digium.com Sent: Sunday, April 08, 2007 12:13 AM Subject: [asterisk-users] Prompt for a PIN number to make long distance call? I need to authenticate users to make long distance calls. Basically,when the user dials a long distance dialplan pattern, I want to prompt for his pin and look it up against a table of pins:usernames in a file. If it exists, I'll use the username in the cdr accountcode and permit the call. Authenticate() looked very promising nut I couldn't get the ma options to work. Any help is appreciated. Honestly, I'm not even sure how to read an external file and parse it from asterisk. -- ___ --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] Audio Gain Settings
Bob Smither wrote: On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote: snip You are right. zapata.conf is not used in IAX connections. My reading has led me to believe that manipulating gain on an IP PBX is neither necessary nor practical in VoIP channels, so Asterisk does not devise such settings. Thanks Yuan. I beg to differ with the developers if there really is no amplitude control on IP channels. I have an application where I am studying the spectrum of recorded voice. When I call into my Asterisk box I have to hold the phone away and speak softly to avoid clipping the recorded waveform - clipped waveforms play havoc with the spectrum. I guess it is time to study the source code (ugh!). The device doing the IP/TDM conversion should be the device that sets the gains correctly. The same applies to echo canceling. ___ --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] Prompt for a PIN number to make long distance call?
You can set up a simple mysql table with PIN-users this makes it more
extensible and you can create a simple web interface to change to pins/add
users.
after you have set up the table just use a simple IVR construct to prompt
for the PIN, fetch it from the table and authenticate it - something like
this (wrote it on my notepad so check the syntax):
exten = _,1,Noop
exten = _,2,MYSQL(Connect connid localhost changeme changeme changeme)
exten = _,3,MYSQL(Query resultid ${connid} SELECT\ pin\ from\
user_pin_table\ where\ pin=${EXTEN})
exten = _,4,MYSQL(Fetch fetchid ${resultid} pin)
exten = _,5,Authenticate(${pin})
if the auth is okay, you can fetch the username for that PIN using
Set(CDR(accountcode)=fetched_user)
Joss.
On 4/8/07, J French [EMAIL PROTECTED] wrote:
I need to authenticate users to make long distance calls. Basically,when
the user dials a long distance dialplan pattern, I want to prompt for his
pin and look it up against a table of pins:usernames in a file. If it
exists, I'll use the username in the cdr accountcode and permit the call.
Authenticate() looked very promising nut I couldn't get the ma options to
work. Any help is appreciated. Honestly, I'm not even sure how to read an
external file and parse it from asterisk.
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