Re: [Asterisk-Users] Nufone Connection
sure it does, how is this any different from-- I'm having trouble with hardware component X and I can't seem to get help from the vendor. Does anyone have any suggestions? -reed brian wrote: Please take this off list and email [EMAIL PROTECTED] it has NO PLACE HERE! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Sent: Tuesday, May 25, 2004 7:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Nufone Connection I am having difficulty with ring back on my Nufone connection. As I have been unsuccessful in getting Nufone to respond and address this issue I would like to know if anyone else is having this problem. I have noticed in posts on this forum that others have had issues with the support from Nufone which is very disappointing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-providers mailing list?
It seems like it might be nice to have a mailing list to talk about (and to) voip providers for Asterisk users. It would be a good place to share info about config, pricing news, customer service, local numbers, transient outages, etc. Providers would be encouraged to contribute sales info. Users would be able to help each other out with technical and non-technical issues. Seems good for everyone and it would keep some of the noise and hurt feelings out of the other lists. The real goal of the list would be to improve the quality of the experience for customers and suppliers. This is something we need to improve in order for voip to be taken more seriously. ? -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-providers mailing list?
Eric Wieling wrote: I thought that is what the Asterisk-Biz mailing list was for. http://lists.digium.com/mailman/listinfo/asterisk-biz I'm thinking more along the lines of hey, VoicePulse is broken today type of email. Which, in fact, is what got me thinking of it when I had VoicePulse problems this morning. -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-providers mailing list?
John Todd wrote: Would providers actually contribute meaningful discussion and data on such a list? My experience shows that the majority of providers that I know (and have worked with or for) and who use Asterisk have not once, ever, posted anything to either the -dev list or the -users list. That number is more than ten and less than thirty, to be suitably vague. In fact, the only activity on any VoIP list or organizations from any of the providers I've worked for seems to be... me. Well, there's Jeremy, the NuFone guy. I can imagine he'd like a place he can send out notes regarding NuFone topics and responding to issues that may come up. Absent others, and in any case I would expect users with experience with other providers would provide the bulk of the info. Maybe -biz is already the right place and it's mandate could be clarified or expanded. -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse broken?
I had in and outbound problems with them for at least a couple of hours early this afternoon. Seems to be ok now tho. -reed Scott Weis wrote: Inbound is working here, no problems that I know of. Scott - Original Message - From: C. Sullivan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 12:52 PM Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival and gcc 3.3.2 (Fedora Core 1)
That did it.
thank you, thank you, thank you,
-reed
Marc Sutter wrote:
Hi,
had the same problem... and we wrote a patch.
This patch's are for speech_tools 1.2.3 and festival 1.4.3.
to use in the corresponding directory with:
#patch -p1 patch..
Hope this help. If so let it know.
Have fun !!!
On Sat, 2004-05-01 at 02:35, Reed Wade wrote:
Can someone tell me how to build festival on a machine with gcc 3.3.2?
I've searched all around and even found a reference or two that the
problem exists but I'm not seeing the fix.
thanks!
-reed
Symtoms are --
./configure, then
[EMAIL PROTECTED] speech_tools]# make
Check system type
Remake modincludes.inc
NATIVE_AUDIO
ok
EDITLINE
config/modules/editline.mak
SIOD
siod/siod.mak
WAGON
stats/wagon/wagon.mak
SCFG
grammar/scfg/scfg.mak
WFST
grammar/wfst/wfst.mak
OLS
stats/ols.mak
RXP
rxp/rxp.mak
LINUX16_AUDIO
config/modules/linux16_audio.mak
Making in directory ./siod ...
making dependencies -- siodeditline.c cc1: warning:
-Wno-non-template-friend is valid for C++ but not for C/ObjC
cc1: warning: -Wno-deprecated is valid for C++ but not for C/ObjC
el_complete.c cc1: warning: -Wno-non-template-friend is valid for C++
but not for C/ObjC
cc1: warning: -Wno-deprecated is valid for C++ but not for C/ObjC
editline.c cc1: warning: -Wno-non-template-friend is valid for C++ but
not for C/ObjC
cc1: warning: -Wno-deprecated is valid for C++ but not for C/ObjC
el_sys_unix.c cc1: warning: -Wno-non-template-friend is valid for C++
but not for C/ObjC
cc1: warning: -Wno-deprecated is valid for C++ but not for C/ObjC
slib.cc slib_core.cc slib_doc.cc slib_file.cc slib_format.cc
slib_list.cc slib_math.cc slib_sys.cc slib_server.cc slib_str.cc
slib_xtr.cc slib_repl.cc siod_fringe.cc siod_server.cc io.cc trace.cc
EST_SiodServer.cc siod.cc siod_est.cc
g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend
-Wno-deprecated -DSUPPORT_EDITLINE -I../include slib.cc
In file included from ../include/EST_String.h:50,
from ../include/siod.h:17,
from slib.cc:88:
../include/EST_iostream.h:54:26: strstream.h: No such file or directory
make[1]: *** [slib.o] Error 1
make: *** [siod] Error 2
[EMAIL PROTECTED] speech_tools]#
___
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diff -ur festival.bad/src/modules/base/phrasify.cc
festival/src/modules/base/phrasify.cc
--- festival.bad/src/modules/base/phrasify.cc 2001-04-04 13:55:20.0 +0200
+++ festival/src/modules/base/phrasify.cc 2004-04-25 15:57:52.0 +0200
@@ -218,7 +218,7 @@
EST_Val npbreak = wagon_predict(w,phrase_type_tree);
w-set(pbreak,npbreak.string()); // may reset to BB
}
- pbreak = w-f(pbreak);
+ pbreak = w-f(pbreak).string();
if (pbreak == B)
w-set(blevel,3);
else if (pbreak == mB)
diff -ur festival.bad/src/modules/base/word.cc festival/src/modules/base/word.cc
--- festival.bad/src/modules/base/word.cc 2001-04-04 13:55:20.0 +0200
+++ festival/src/modules/base/word.cc 2004-04-25 15:59:55.0 +0200
@@ -64,10 +64,10 @@
for (w=u-relation(Word)-first(); w != 0; w = next(w))
{
lpos = NIL;
- pos = ffeature(w,hg_pos);
+ pos = ffeature(w,hg_pos).string();
// explicit homograph pos disambiguation
if (pos == 0)
- pos = ffeature(w,pos);
+ pos = ffeature(w,pos).string();
if (pos != 0)
lpos = rintern(pos);
@@ -100,8 +100,8 @@
// from which a list can be read.
EST_String p;
-if (((p = ffeature(w,phonemes)) != 0) ||
- ((p = ffeature(w,R:Token.parent.phonemes)) != 0))
+if (((p = ffeature(w,phonemes).string()) != 0) ||
+ ((p = ffeature(w,R:Token.parent.phonemes).string()) != 0))
{
LISP phones = read_from_lstring(strintern(p));
diff -ur festival.bad/src/modules/Intonation/int_tree.cc
festival/src/modules/Intonation/int_tree.cc
--- festival.bad/src/modules/Intonation/int_tree.cc 2001-04-04 13:55:20.0
+0200
+++ festival/src/modules/Intonation/int_tree.cc 2004-04-25 15:58:42.0 +0200
@@ -87,11 +87,11 @@
for (s=u-relation(Syllable)-first(); s != 0; s=next(s))
{
if ((paccent = accent_specified(s)) == 0) // check if pre-specified
- paccent = wagon_predict(s,accent_tree);
+ paccent = wagon_predict(s,accent_tree).string();
if (paccent != NONE)
add_IntEvent(u,s,paccent);
if ((ptone
Re: [Asterisk-Users] Grandstream Budgettone 100 102
With shipping, I recall my 102 came to $97. I think it was $85 but I'd need to look it up and don't have the papers nearby. -reed At 06:39 PM 7/30/2003 -0500, you wrote: I was quoted $75 and $85 USD today. Ricardo Villa http://www.telesip.net - Original Message - From: Joe Cooke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:31 PM Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 102 I was quoted the $75 and $85 USD prices from Grandstream direct about 2 months ago. I'm not sure if it makes a difference, but I live in the US. - Joe - Original Message - From: marrandy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:17 PM Subject: [Asterisk-Users] Grandstream Budgettone 100 102 Checking the earlier mails, it stated that the phones were $75 (100) $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 $85 rate ??? Regards...Martin -- Too much is just enough. -- Mark Twain, on whiskey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider for Asterisk?
Send email to [EMAIL PROTECTED] I did this late Friday afternoon and Jeremy had me set up in very short order. (I gave him wrong contact info Friday but he IM'd me the needed account info Saturday morning.) I've got problems with my own IP connection but aside from that the service just works. Once I resolve that I'll be able to truly vouch for the line quality. I've got an 800 number for incoming calls and I can route long distance calls out to NuFone. Both directions is $0.029/minute. They are still building their web based account management tools but I saw a preview and they look pretty nice. It's a prepay service and there doesn't seem to be an account set up fee right now so it's easy to get set up and try it out--that's what I'm doing. For what it's worth, I didn't do any investigation of alternatives. Good customer service, like NuFone appears to be in the business of, is usually worth a lot more than maybe getting the lowest rate. -reed At 09:20 AM 7/20/2003 +0300, you wrote: Hi, How can you subscribe to this service? There is no web page available to do it. Thanks, Dan - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 6:07 PM Subject: RE: [Asterisk-Users] Best VoIP provider for Asterisk? Agreed. Jeremy McNamara of Nufone.net is the top dog in Asterisk VOIP and long distance. His systems and network are the most stable I have ever seen. It is all ran out of the same facilities as the TOP long distance providers. All fiber, all stable, 3x and 4x redundancy. He has done some amazing things. Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of James H. Cloos Jr. Sent: Saturday, July 19, 2003 5:47 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best VoIP provider for Asterisk? Marcus == Marcus Adolfsson [EMAIL PROTECTED] writes: Marcus Nufone.net is the best VoIP provider for Asterisk Marcus integration. They offer IAX termination, 2.9 cents outgoing Marcus long-distance and incoming 800. We use them at our office for Marcus all phone calls. I second this. But note they are now at 2.0 cents for calls to US and Canada. They change the same per minute for incoming calls on the 800 numbers. They are responsive, competent; simply great to work with. Highly recommended. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream BudgeTone 102 initial experiences
Just to toss in my very limited experiences with the Grandstream phone-- I haven't tested it enough to really know nor is my Asterisk config set up enough to fully try all the features. Mostly, it just works. It was very easy to configure and get running. I've been toting it around to clients as a show and tell exhibit and it has helped get people excited about the possibilities. Voice quality is great using the handset. I have had some problems using the speaker phone. Folks say they can barely hear me. Also, it's prone to double digiting when the speaker phone is used--you type 307 and Asterisk hears 33077. I haven't upgraded the firmware or messed with any settings so these problems may be resolvable. I do intend to try because it's an otherwise reasonable phone and I want it to work. It seems to work well behind a NAT. I agree with others regarding the look and feel. I've seen far worse but it does feel a little cheap. If they just changed the name or labeling from BudgeTone to ExpensiTone but kept the good pricing that would help a lot. It lacks elegance. The soon to be released grey model may improve it's image. You cannot wall mount this phone (easily). There are mounting holes on the bottom and the handset has a little concavity in the right place but there's no nub/pin to keep the handset in place when on hook. I haven't asked the Grandstream folks about this--maybe I got a dud or maybe they already plan to address this. I just bought a new house and am seriously thinking about sprinkling these throughout and using Asterisk to control things like the lights and thermostat (oh, and for phone calls, too). -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone and NTP (redux)
Just to add another data point -- I have never had my BudgeTone 102 fail to get NTP service. I've used it behind two different NAT'd networks with relatively relaxed firewalls. -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
The best solution would be an enhancement to the X100P card. If the 2nd RJ jack was a pass through for the line except when the card had power and was initialized. Some kind of watchdog functionality would also be nice so that if, for example, Asterisk dies then pass through functionality would take effect after n seconds. This would probably mean adding a relay to the board which would raise to cast a little. But, as the original poster indicated this is critical for a serious system. An alternative would be an extra relay box, maybe powered by USB. One mode could be to switch based on presence of power, another mode could require periodic watchdog pings via the USB. I always wanted to build something using a USB flavored PIC... I can see this for small offices (like ours). We have 4 incoming lines in a hunt group. If Asterisk is not running I want one of those lines to ring the receptionist (maybe using a simple dedicated phone since they'd otherwise have an IP phone) and the others looped for busy. I can see a box with USB and 12 RJ jacks (4 x (1 in, 2 outs)) to make that work. Would anyone buy a product like that? -reed At 07:12 AM 7/14/2003 -0500, jltaylor wrote: This power failure thing does not have to be complicated. A few solutions come to mind: 1) A 3,5,12 (whatever is needed) power supply (wall wart)used with a relay (DPDT). When the wall wart has power, the computer takes the call. When power fails, the POTS line falls in to place. Now, this does not delay while the computer is booting up. 2) A basic stamp computer - about $25-30. It has 8 programmable i/o pins that will drive relays. One pin monitors either a wall wart or 5v from one of the plugs on your computer's power supply. When pin 1 goes low (no power) relay kicks in to bypass computer and connect POTS line direct. When power returns program jumps to a sleep or delay statement for xMINS until computer boots. And then releases relay for normal operation. www.parallaxinc.com and resellers. James Taylor [EMAIL PROTECTED] 903-793-1953 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
At 11:34 AM 7/14/2003 -0500, Steven Critchfield wrote: One wouldn't use a X100P in a serious system. How so? I assume you're talking about scale and not reliability. We get a relatively small number of calls but any one of them could be worth a large stack of cash for our business. A stinky phone system can make us look bad. The main reason I'm looking at Asterisk is to improve the reliability and control over our phone system. All the other great things it provides really are secondary for the folks who pay my salary. Only if you aren't pulling power from the USB bus. There isn't much there. There may be just enough depending on how many relays are needed, but it would be too close. I agree, better off not trying to get power from there. I do like the idea of some kind of watchdog functionality. Simply having power isn't sufficient to trust that a call is getting routed. -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
At 12:57 PM 7/14/2003 -0500, you wrote: This makes me think that you could take this a step further too and incorporate an external power supply and a relay that could interupt mains power so that you could power cycle the PC if the watchdog had power to operate and the PC wasn't responding or generating pings. i like that -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
Because you haven't written and contributed that functionality yet. (smiley face goes here) That sounds pretty sweet. I'm wondering if LDAP might be the more correct thing to use though. -reed At 10:50 AM 6/24/2003 -0600, you wrote: I've been scouring the archives for discussions on this: Why doesn't Asterisk use system user accounts for each extension/mailbox? That would add the benefit of encrypted passwords, logical grouping, unified mail/voice mail accounts (using /var/spool/mail instead of /var/spool/asterisk). I can already imagine Festival reading my emails to me, HylaFAX faxing documents to me while I'm on the road :). Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
thanks!, was Re: [Asterisk-Users] newbie needs SIP config examples -- especially soft phones
thanks to everyone for your gracious assistance; it stills wants plenty of minor adjustments but I now have the core of a nicely working system -reed At 11:56 PM 6/17/2003 -0500, John Laur wrote: So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used xten for every option everywhere (display name, username, password, and Domain/Realm) and the corresponding section in sip.conf. I've had no luck getting the SJ Labs soft phone to connect using a similar blunderbuss method. [youruser] ;username here and also below... type=friend;dial both to and from username=youruser ;same thing as in brackets above password=password ;password obviously context=default;or put whatever you want - this is the sip realm too mailbox=1234 ;for message waiting host=dynamic ;might be coming from different ip's callerid=Soft Phone 1234 nat=yes;might be behind a nat I'm wondering if someone could point me to SIP configuration examples or education so I can understand what I'm doing. I'm finding the client configuration more confusing that the * configs. Your client will want an auth name or two (use the username for these), a secret or password (the password), a port number (5060 is the default and you can change it in the [general] section of sip.conf), maybe a realm (the context though it is not important for authentication), a sip proxy address - your asterisk server's ip address, and that should be it. Most have an option you have to turn on to tell the client to actually register with the proxy. turn that on and check to see that your client is connected with 'show sip peers' on the asterisk console. It might also be helpful to turn on 'sip debug' to see if your client is trying to register. If you got the x-lite working the others should be easy too.. You'll see.. An example of password protected SIP phone access would also be very helpful. see above. I need to be able to support folks working from home connecting through the net as well inside the office. I expect NAT to be a pain. NAT is not so hard once you get it going. First: make sure your asterisk server has a public IP address and the ONLY default gateway on the machine is set to the router for the public ip. Make sure you have set nat=yes in the corresponding sip.conf entry for the device you're setting up, then start poking at your client for the settings that say I'm behind a NAT -- they are designed to make sure the packets source at the same UDP ports they need to come back to so that the NAT's will open up a pathway back to the internal device. Some clients do this by default anyway -- On the X-Lite phone you don't really have to do much of anything -- maybe uncheck the box that says Send Internal IP though I have found that it doesnt really matter if nat=yes on the asterisk box. On the cisco 7960 phones, the following settings work: nat_enable: 1 nat_address; voip_control_port: 5060 start_media_port: 16384 ; You can reduce this port range if you end_media_port: 32766 ; have a picky firewall nat_received_processing: 1 ; Makes phone re-register if your ip changes Hope this helps you some... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie needs SIP config examples -- especially soft phones
Hi, I'm experimenting with the dev kit lite and now past the USB unpleasantness it's working great with standard phones and lines. The priority right now is getting soft phones (under Windows XP) working well. So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used xten for every option everywhere (display name, username, password, and Domain/Realm) and the corresponding section in sip.conf. I've had no luck getting the SJ Labs soft phone to connect using a similar blunderbuss method. I'm wondering if someone could point me to SIP configuration examples or education so I can understand what I'm doing. I'm finding the client configuration more confusing that the * configs. An example of password protected SIP phone access would also be very helpful. I need to be able to support folks working from home connecting through the net as well inside the office. I expect NAT to be a pain. thanks, -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soft phones -- voice quality tuning
I was about to say 'yes' and they were worse but now I can't remember for certain. I'll try those again to make sure. Are they likely to be better? -reed At 02:27 PM 6/18/2003 +1000, Gary wrote: tried the ilbc and speex yet ?? On Wed, 18 Jun 2003 00:15:38 -0400, Reed Wade wrote: I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP? thanks, -reed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
