Re: [Asterisk-Users] anyone seen this error when running asterisk!
On Sunday 08 June 2003 23:36, hallian hallian wrote: Hi all - I'm making gradual progress implementing asterisk on my box! Now, when I type asterisk it dies at this point. Does anyone have any idea why this is happening! It have checked everything but running out of options! [app_voicemail2.so] = (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain2' [app_transfer.so] = (Transfer) == Registered application 'Transfer' [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) Illegal instruction Best guess is that you're compiling for the wrong series of processors. What are the outputs of the following commands? 1) grep '^PROC' /usr/src/asterisk/Makefile 2) uname -a -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie question on soft phones with SIP and *
Thanks Andy, I'll send you feedback. /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Powell Sent: 06 June 2003 03:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie question on soft phones with SIP and * Tielman, You can take a look at the quick and dirty guide I'm slowly putting together if you like... http://www.automated.it/guidetoasterisk.htm I'd appreciate any feedback you have on it.. and if it helped Andy *** REPLY SEPARATOR *** On 06/06/2003 at 14:17 Tielman Koekemoer wrote: I am new to the Telephony world and am trying to get a basic idea of how things work. Can I use Asterisk to connect two soft phones on a LAN to communicate with one another without any additional hardware (besides the sound-card)? If this is possible, I seem to be doing something wrong. I have installed a version of X-Lite and can't seem to get it to ring on my Win PC. Any pointers to docs that can give me an idea (except for the handbook which I'm reading) would be greatly appreciated. TIA Tielman /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ _ Content and Virus scanned by Inflex and Mcafee ___ 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 _ Content and Virus scanned by Inflex and Mcafee _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 and extentions.conf
hi i am not using sio or iax but only oh323. i am trying to register my extensions like extensions.conf ;-- H.323 [alias = 665] exten = 665,1,Dial(OH323/172.18.1.133) oh323.conf context=voip-h323 ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 alias=665 alias=667 but when i try to dial 665 i get :23.412 H323 Cleaner H323Connection ip$172.18.1.133:1344/31268 terminated. ERROR[33816]: File chan_oh323.c, Line 1015 (oh323_answer): H323:31269: Failed to answer call with token ip$172.18.1.133:1368/31269 (timeout). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI
Has anyone seen this behavior? Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete messages. The 7 button no longer does anything... Thanks for any help... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] small office
Thanks Andrew John, What about the suggestion for the SOHO? 1) What is the best cost effective solution for a small office: I need 3 FXS 2 FXO. 2) Can I hookup a TDM400P (3FXS) and 2 X100P (FXO) on the same computer? TIA -Original Message- From: John Congdon [mailto:[EMAIL PROTECTED] Sent: Saturday, June 07, 2003 9:07 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] small office beware, I bought one of these phones and there is no documentation. I do not think their IP = the IP you want in a phone. I still have not found out what it means, but It surely doesn't look like what we need. John -Original Message- From: Andrew Gillham [mailto:[EMAIL PROTECTED] Sent: Friday, June 06, 2003 6:11 PM To: Dante Alzamora Subject: Re: [Asterisk-Users] small office On Fri, Jun 06, 2003 at 05:09:13PM -0400, Dante Alzamora wrote: What is the best cost effective solution for a small office: I need 3 FXS 2 FXO. Can I hookup a TDM400P and 2 X100P on the same computer? Also, I saw some IP phones for $25.99 http://www.wosmile.com/cgi-bin/view_store_item.cgi?pid=4424sid=5category=1 Can I use them with asterisk? will they be able to do the same as the TDM400P? Those are not IP Phones in the sense of TCP/IP. I believe the term IP in this case is something like 'initial prefix' or similar. I've seen these on eBay, and looked into them. From what (little) I could find, it appears that you can configure the phone to dial some pre-defined number when you pick it up, or before dialing the entered numbers. (like it can do '9' or '10-10-321' or whatever) It is simply not an H.323, MGCP, SCCP, SIP or anything IP phone. I would say look into the Grandstream phones. http://www.grandstream.com/ I read that to run the conference app Meetme you needed a Zaptel driver. You can use the ztdummy driver that uses your USB controller for timing. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small office
You can, your only problem will be to have a really nice and clean pci bus on your motherboard so be very very carefull with your selection there. On Monday 09 Jun 2003 3:12 pm, Dante Alzamora wrote: Thanks Andrew John, What about the suggestion for the SOHO? 1) What is the best cost effective solution for a small office: I need 3 FXS 2 FXO. 2) Can I hookup a TDM400P (3FXS) and 2 X100P (FXO) on the same computer? TIA -Original Message- From: John Congdon [mailto:[EMAIL PROTECTED] Sent: Saturday, June 07, 2003 9:07 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] small office beware, I bought one of these phones and there is no documentation. I do not think their IP = the IP you want in a phone. I still have not found out what it means, but It surely doesn't look like what we need. John -Original Message- From: Andrew Gillham [mailto:[EMAIL PROTECTED] Sent: Friday, June 06, 2003 6:11 PM To: Dante Alzamora Subject: Re: [Asterisk-Users] small office On Fri, Jun 06, 2003 at 05:09:13PM -0400, Dante Alzamora wrote: What is the best cost effective solution for a small office: I need 3 FXS 2 FXO. Can I hookup a TDM400P and 2 X100P on the same computer? Also, I saw some IP phones for $25.99 http://www.wosmile.com/cgi-bin/view_store_item.cgi?pid=4424sid=5category= 1 Can I use them with asterisk? will they be able to do the same as the TDM400P? Those are not IP Phones in the sense of TCP/IP. I believe the term IP in this case is something like 'initial prefix' or similar. I've seen these on eBay, and looked into them. From what (little) I could find, it appears that you can configure the phone to dial some pre-defined number when you pick it up, or before dialing the entered numbers. (like it can do '9' or '10-10-321' or whatever) It is simply not an H.323, MGCP, SCCP, SIP or anything IP phone. I would say look into the Grandstream phones. http://www.grandstream.com/ I read that to run the conference app Meetme you needed a Zaptel driver. You can use the ztdummy driver that uses your USB controller for timing. -Andrew ___ 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] VoIP Provider
On Mon, 9 Jun 2003, Gary wrote: Just a quick look at their rates show they just might be into rip off's.. Australia0.06 (0-61-0) Australia-Cellular 0.31 (0-61-7) Australia-Cellular 0.31 (0-61-8) Australia-Cellular 0.31 (0-61-1) Australia-Cellular 0.31 (0-61-4) Australia-Cellular 0.31 (0-61-5) Australia-Cellular 0.31 (0-61-71) Australia-Cellular 0.31 (0-61-78) Australia-Cellular 0.31 (0-61-79) Australia-Melbourne 0.06 (0-61-3) Australia-Sydney 0.06 (0-61-2) What they label as Australia-Cellular is bullshit !! Only one being 61-4 is actually cellular. Also their rates are quite high for VoIP. 0.06 for minutes within the USA? That's twice as much as other VoIP providers. Their international rates to places I call are also similarly high. miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8 VOIP service now 1/2 the price
Unlimited US and Canada VOIP to PSTN calls for $20/month no equipment fees, no contract Does anybody have direct Asterisk to Packet8 fully working without the MTA? http://biz.yahoo.com/prnews/030609/sfm088_1.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie question on soft phones with SIP and *
Andy Very good, your doc was priceless in getting my basic setup working. On the feedback side I cannot say what else I wanted to see in your doc as my brain has not caught up to the many possibilities but something will come. Thanks again! Tielman /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Powell Sent: 06 June 2003 03:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie question on soft phones with SIP and * Tielman, You can take a look at the quick and dirty guide I'm slowly putting together if you like... http://www.automated.it/guidetoasterisk.htm I'd appreciate any feedback you have on it.. and if it helped Andy *** REPLY SEPARATOR *** On 06/06/2003 at 14:17 Tielman Koekemoer wrote: I am new to the Telephony world and am trying to get a basic idea of how things work. Can I use Asterisk to connect two soft phones on a LAN to communicate with one another without any additional hardware (besides the sound-card)? If this is possible, I seem to be doing something wrong. I have installed a version of X-Lite and can't seem to get it to ring on my Win PC. Any pointers to docs that can give me an idea (except for the handbook which I'm reading) would be greatly appreciated. TIA Tielman /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ _ Content and Virus scanned by Inflex and Mcafee ___ 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 _ Content and Virus scanned by Inflex and Mcafee _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI
I have the same problem. I use an Aastra 480 phone and as long as I don't touch any of the ADSI soft-buttons then my keypad stays active and the downloaded script works great. But as soon as I hit listen through the ADSI display, all of my normal 0-9*# keys get disabled and the script no longer maps any more options to my soft buttons. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Congdon Sent: Monday, June 09, 2003 6:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ADSI Has anyone seen this behavior? Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete messages. The 7 button no longer does anything... Thanks for any help... 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] Premisys channel bank
Is anyone using or have any information on the Premisys T1 FXS channel bank? I believe that the model number is 240224. I think it also goes under the nortel pn:nt-4k58-ab. I am interested i nknowing if this channel bank will work with asterisk and if anyone is using it how they like it? A.J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
The Nufone Network can set you up for $10/ month unlimited at 2.9 cents/minute through US and Canada. Buy a couple FXOs and you can share your savings with your friends. -Greg - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 09, 2003 10:55 AM Subject: [Asterisk-Users] Packet8 VOIP service now 1/2 the price Unlimited US and Canada VOIP to PSTN calls for $20/month no equipment fees, no contract Does anybody have direct Asterisk to Packet8 fully working without the MTA? http://biz.yahoo.com/prnews/030609/sfm088_1.html ___ 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] Adding an app
I am in the process of testing out the Cisco ATA 186 to provide analog phone service via VoIP for some of our remote users. I have that working fine and well, but am struggling with another aspect. We already have a large centralized voicemail system, which I would like to use for these users. I can get the call to roll to new the centralized voicemail no problem, but I'd like to provide message waiting for them as well. I've seen where Asterisk can provide MWI via stutter dial tone for its own internal voicemail. After looking at the source code, it appears that there is a flag set that determines whether or not * needs to play stutter dial tone when the user lifts the handset. I can get my voicemail system to pass the MWI to Asterisk in this form: 810NXX == turns off MWI 811NXX == turns on MWI I would like to set up extensions.conf like this: exten = 810NXX,1,MessageWaitOff(EXTEN:6) exten = 811NXX,1,MessageWaitOn(EXTEN:6) I would like the MessageWaitOff and MessageWaitOn apps to appropriately set the flag for the extension that is passed to it. This is where my problems start. I have spent some time looking through the source code, but I haven't determined what all I need to touch to add an application and have Asterisk recognize it. I have created the apps, compiled them and created the shared object file. What other steps do I need to take? Will what I'm proposing here even work? Any feedback is appreciated. Thanks, Jesse Knutsen Associate Project Engineer Union Pacific Railroad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question for someone running hylafax off *.
Hi,
I am setting up a hylafax server. From what I've read so far, hylafax
supports CID numbers and names but currently does not support DID. I assume
I can do something like this...
[40faxDIDs]
exten = _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN})
exten = _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP})
...and use the CIDName variable in hylafax to route the faxes to the
appropriate destination.
My question is this, is anyone doing something like this, and if so, what is
the best way to accomplish the routing.
Thanks,
John
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[Asterisk-Users] OH323 crashing
hi, does anyone have a problem with OH323 crashing with a segmentation fault whenever anything tries to connect to it ??? are the current CVS versions OK? Would like to speak to someone with a bit of OH323 experience, so if u're in a good mood to help, please do :) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
The Nufone Network does IAX termination at 2.9 cents a minute to US. -Greg [EMAIL PROTECTED] Please disregard my last post. $10 bucks will get you 344 minutes - no contracts. - Original Message - From: shido [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 09, 2003 1:59 PM Subject: Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price The Nufone Network can set you up for $10/ month unlimited at 2.9 cents/minute through US and Canada. Buy a couple FXOs and you can share your savings with your friends. -Greg - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 09, 2003 10:55 AM Subject: [Asterisk-Users] Packet8 VOIP service now 1/2 the price Unlimited US and Canada VOIP to PSTN calls for $20/month no equipment fees, no contract Does anybody have direct Asterisk to Packet8 fully working without the MTA? http://biz.yahoo.com/prnews/030609/sfm088_1.html ___ 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] iLBC, Speex and X-Lite
I've been trying out the newest X-Lite (Build 1012) with iLBC and speex codecs. If I enable only iLBC _or_ SPX on X-Lite and call the echo-test on my asterisk server, the call connects, but I get no sound. If I enable only iLBC _and_ SPX, X-Lite indicated that it has connected with iLBC, and I hear a weird squawking. My sip.conf contains: allow=iLBC allow=SPEEX allow=gsm I've heard that asterisk and kphone will work together using iLBC. Does anyone else have any interoperability experience? Any ideas on what I should do to debug this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
See this for one review. http://lists.digium.com/pipermail/asterisk-users/2003-March/009446.html I think some other people have it working but I am not sure if they got all the bugs out. - Original Message - From: Richard Alexander [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 09, 2003 2:00 PM Subject: RE: [Asterisk-Users] Packet8 VOIP service now 1/2 the price Do Packet8 provide the necessary info to use it with * for inbound and/or outbound ? Any idea what codecs they support ? ___ 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] ADSI
It doesn't happen on Nortel 350. Martin On Mon, 9 Jun 2003, David Carr wrote: I have the same problem. I use an Aastra 480 phone and as long as I don't touch any of the ADSI soft-buttons then my keypad stays active and the downloaded script works great. But as soon as I hit listen through the ADSI display, all of my normal 0-9*# keys get disabled and the script no longer maps any more options to my soft buttons. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Congdon Sent: Monday, June 09, 2003 6:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ADSI Has anyone seen this behavior? Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete messages. The 7 button no longer does anything... Thanks for any help... 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] working with SIP soft phones
Hi all, --- how can I use SIP softphones together with asterisk such as kphone, siset, ... for LINUX wich do not support number dialing. From my Snom 100 it was easy to dial the softphones, but it did not work in the opposite way. To Dial phone 2 I have just pressed 2 at the Som100 and I could reach the LinPhone (Linux Softphone). The I tried to ring the Snom100 from Linphone with sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] but I did not hear any response from the Snom 100. Any hints to solve this issue ?? regards Olaf - sip.conf [phone1] type=friend host=192.168.5.101 dtmfmode=rfc2833 mailbox= context=siphome callerid=Snom100 2123 [phone2] type=friend host=dynamic defaultip=192.168.5.103 dtmfmode=rfc2833 -- ? wich mode to use with a softphone ?? mailbox= context=siphome callerid=LinPhone 2124 --- extension.conf [siphome] include = fwd exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) ; VoiceMail exten = 1000,2,VoiceMail,u exten = 1000,102,VoiceMail,b exten = 1001,1,Ringing exten = 1001,2,Wait(2) exten = 1001,3,VoicemailMain,s - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
This is to state that a recent message posted to asterisk mailing list by [EMAIL PROTECTED] regarding the pricing of our sample phones is NOT accurate. Grandstream Networks has NOT changed the list price for its products and samples. Our BudgeTone 100 series IP phones lists at $75 for model 101, NOT $60. Grandstream is committed to supporting the asterisk community and this message is posted for the sole purpose of correcting a misinformation regarding our product. Thanks for your attention to this matter. Grandstream Customer Support ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modem driver question
I'm trying to start-up Asterisk with a local modem (on /dev/ttyS1) and I can't find a modem driver that loads. The attached modem is a USRobotics Sportster Voice 33600 Fax RS Rev. 2.0 (as reported by Asterisk). Anyone know which driver I should load? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much to use Dialogic?
Slightly Off topic: What are you using to convert the digital ports to analog with DS? We've tried the ATA and ATA 2, and neither have DS. The Norstar VMI was the only device we could find, but is a little overkill. Correct, with the Norstar or BCM the ATA's don't provide DS. I have attached a brouchure that will do it. It converts eight ports. I have talked to two people who have used it extensivly. It looks like it was built in someone's garage, but it works well I hear. The email was too large and got rejected by the bot. I put it on our web server, mail.comtelhi.com/nortel/DS8000A.pdf if you want to download it. I think dealer cost is about $1800. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
On Mon, 9 Jun 2003, Jim Flagg wrote: See this for one review. http://lists.digium.com/pipermail/asterisk-users/2003-March/009446.html I think some other people have it working but I am not sure if they got all the bugs out. Too bad noone makes a cheap ethernet FXO. Why is it always FXS... :-/ -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Back
We have Nextel Cell phones where incoming calls are free!
I would like to call a DID number on my Asterisk server,
have it grab my caller ID, not pick up, wait a few seconds, and call me back.
I have already set up qcall to do this and pass it to a
context that asks for a password via authenticate, and is limited to dialing in
my local area so I am not worried about fraud.
I am at the point where it all works except I do not know
the variables in extension.conf {$CALLERID} is the whole strings
including name!! I want just the number.
I could also use this to set up a ANI announcement where you
call the * box and it would use SayDigits to read the number you are calling
from. I searched the archives via google and found nothing.
Anybody got any ideas???
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
On Monday 09 June 2003 05:02 pm, [EMAIL PROTECTED] wrote: On Mon, 9 Jun 2003, Jim Flagg wrote: See this for one review. http://lists.digium.com/pipermail/asterisk-users/2003-March/00944 6.html I think some other people have it working but I am not sure if they got all the bugs out. Too bad noone makes a cheap ethernet FXO. Why is it always FXS... Probably because FXS is an endpoint. FXO ports interface between disparate networks, so they need a bit more intelligence (which is what Asterisk provides). And there's that small matter of FCC regulations for connecting a device to the PSTN. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcing IAXCLIENT v0.02 A cross-platformIAX client.
John, Are you on the iaxclient-devel list? That would be a good place for your feedback. I think we've talked on IRC, and there's definately lots of known issuess, but if you can document some of your problems, that would help! On Wed, 2003-06-04 at 20:15, John Todd wrote: Just wanted to say that this is very encouraging, and I'd like to see more! I've got MacOS-X, and I've been desparately waiting for a SIP or IAX client for this system for a while. I've got your test code running, which looks extremely promising. Alas, I am not a programmer, and couldn't do justice to putting a front-end onto it. But I anxiously await the progress of others, and would happily give you more feedback than you probably want. JT Asterisk-people, Some of you may have heard that we were working on a simple, cross-platform IAX client library called iaxclient. We've pretty much been on vacation with the project for a while, but recently have made some progress, and now have the library working across platforms, and a simple test client called testcall, up and running on 3 platforms: testcall runs on Linux, Windows and Mac OSX. I've been using it for a day now, and the quality is pretty good, of course it does depend on the sound hardware you're using, etc. testcall is a simple command-line app, which is just a test interface to the library, but it seems to work well, and I've used it to make some calls both across local networks, as well as over the internet. Some more information, including links to pre-compiled binaries, are available at http://iaxclient.sourceforge.net Source code and build directions are available from the sourceforge CVS servers. Iaxclient uses digium's libiax, the portaudio audio abstraction libraries, the gsm codec, and other components. Feedback is, of course, welcome. People interested in making a cross-platform GUI for a more full-featured client would also be welcomed. Features upcoming would include: 1) Audio processing code: Audio level detection, for callbacks to GUI level meters, silence detection, and Automatic Gain Control (for adjusting input levels automatically) [perhaps also better handling of clipped inputs?] 2) Upgrades to libiax2 3) Simple, and lightweight GUIs for Win32 and MacOSX 4) testing and tweaking of jitter buffers, recovery, and other reliability enhancements [after first moving to libiax2]. -SteveK -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 HorizonLive.com - collaborate . interact . learn The box said 'Requires Windows 95, NT, or better,' so I installed Linux.
Re: [Asterisk-Users] IAX on windows
Hey Ron, I had you on my list to notify when there was something available. See http://iaxclient.sf.net/ Still under development, but there's working cross-platform phones there now. -SteveK On Sat, 2003-03-08 at 20:32, Ron Gage wrote: I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 HorizonLive.com - collaborate . interact . learn The box said 'Requires Windows 95, NT, or better,' so I installed Linux.
[Asterisk-Users] Dual T400P, SMP, performance issues
Title: Dual T400P, SMP, performance issues Hi, We are trying to validate Asterisk as a media gateway PRI - SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support). On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s out of 8. On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle. The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second, but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts (after the box reboot, so they are not that big as they were) - do they look OK? CPU0 CPU1 CPU2 CPU3 0: 122556 0 0 0 IO-APIC-edge timer 1: 4 0 0 0 IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0 IO-APIC-edge rtc 12: 20 0 0 0 IO-APIC-edge PS/2 Mouse 14: 23 0 2 0 IO-APIC-edge ide0 20: 516930 0 0 0 IO-APIC-level tor2 24: 516524 0 0 0 IO-APIC-level tor2 28: 10600 0 0 0 IO-APIC-level eth0 29: 4837 0 0 0 IO-APIC-level eth1 30: 24831 0 0 0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 122430 122429 122429 122428 ERR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin
[Asterisk-Users] Setting local IP address for the RTP port
Title: Setting local IP address for the RTP port If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin
Re: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
I think this sort of request is an appropriate question for the list, and I for one would not take offence in any way if a manufacturer answers to the list as a whole. Answering a direct question is fine, but unsolicited promotion is another matter. Even product announcements where they are relevant to the list members is fine by me, and I am sure most other people agree it is fine as well. This is why we have email subject headers - if you don't care about the price of a Budgetone 100, don't read the email - simple. At 06:44 PM 6/9/2003 -0700, you wrote: Hello Wade and Asterisk users, As we are committed to supporting Asterisk community, we will not be able to answer questions related to Grandstream product through Asterisk mailing list, this is to be fair and respectful to the Asterisk community as a whole. The previous email is to clear a pricing info regarding the product because a lot users start to use that price as reference price for the phone. Should you have any questions and issues regarding Grandstream product, please send your email to [EMAIL PROTECTED] or [EMAIL PROTECTED] Thank you for your attention and interest in Grandstream product. Best regards, Grandstram Customer Support Wow! A phone manufacturer is actually monitoring this list! Nice work Grandstream. Can you tell us which phones you currently have in stock, and pricing on all models? Can you also let us know if your 1-port FXS device is shipping? Pricing? Thanks in advance, -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Zhang Sent: Monday, June 09, 2003 4:14 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone This is to state that a recent message posted to asterisk mailing list by [EMAIL PROTECTED] regarding the pricing of our sample phones is NOT accurate. Grandstream Networks has NOT changed the list price for its products and samples. Our BudgeTone 100 series IP phones lists at $75 for model 101, NOT $60. Grandstream is committed to supporting the asterisk community and this message is posted for the sole purpose of correcting a misinformation regarding our product. Thanks for your attention to this matter. Grandstream Customer Support ___ 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] Adding an app
On Mon, 2003-06-09 at 13:01, [EMAIL PROTECTED] wrote: I am in the process of testing out the Cisco ATA 186 to provide analog phone service via VoIP for some of our remote users. I have that working fine and well, but am struggling with another aspect. We already have a large centralized voicemail system, which I would like to use for these users. I can get the call to roll to new the centralized voicemail no problem, but I'd like to provide message waiting for them as well. I've seen where Asterisk can provide MWI via stutter dial tone for its own internal voicemail. After looking at the source code, it appears that there is a flag set that determines whether or not * needs to play stutter dial tone when the user lifts the handset. I can get my voicemail system to pass the MWI to Asterisk in this form: 810NXX == turns off MWI 811NXX == turns on MWI I would like to set up extensions.conf like this: exten = 810NXX,1,MessageWaitOff(EXTEN:6) exten = 811NXX,1,MessageWaitOn(EXTEN:6) I would like the MessageWaitOff and MessageWaitOn apps to appropriately set the flag for the extension that is passed to it. This is where my problems start. I have spent some time looking through the source code, but I haven't determined what all I need to touch to add an application and have Asterisk recognize it. I have created the apps, compiled them and created the shared object file. What other steps do I need to take? Will what I'm proposing here even work? You need to get your shared object into the asterisk lib directory, and worst case, add a load command to the modules.conf file. Comment as to how you plan on implementing the above function. The exten does not get the MWI, it is a channel. Channels and extensions are different. Extensions can point to many channels, but a channel only points to one phone interface. You will need to make a lookup to go from extension to channel, then you can go toggle the MWI. Otherwise your mailbox is going to have to be your extension, possibly in that long format, and you will have to traverse every channel to find which ones to set MWI. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much to use Dialogic?
On Mon, 2003-06-09 at 16:08, Matthew John Darnell wrote: Slightly Off topic: What are you using to convert the digital ports to analog with DS? We've tried the ATA and ATA 2, and neither have DS. The Norstar VMI was the only device we could find, but is a little overkill. Correct, with the Norstar or BCM the ATA's don't provide DS. I have attached a brouchure that will do it. It converts eight ports. I have talked to two people who have used it extensivly. It looks like it was built in someone's garage, but it works well I hear. The email was too large and got rejected by the bot. I put it on our web server, mail.comtelhi.com/nortel/DS8000A.pdf if you want to download it. I think dealer cost is about $1800. plop sound of jaw hitting floor. Why not set up a T1 interconnect with an asterisk box? I'm sure the $500 T100P and the cost of a T1 port is smaller than the Dialogic card, the licenses for the dialogic driver, and this device you mention. Not to mention the wiring mess of that many analog ports. 1 wire is nicer. BTW, I think it has been covered here before that the D41 is a half duplex card and wouldn't be good for conferencing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
