Re: [asterisk-users] FYI about my Mona Vie business venture
wow, this stuff is awesome! it's the best thing EVER! it's like the much awaited asterisk for windows! it has fiber! fiber, people, fiber! And it Detoxifies the body of infectious toxins! not just any toxins, infectious toxins! somehow i get the feeling that asterisk is going to be paying less of the bills after this. You know, there are people who sell VOIP services this way, by charging ridiculous prices for ATA's (i've seen $150 and up) and having price plans that are not very attractive, but giving great incentives to sign up as independent agents or things like that - well over the like from multi-level marketing (which is not so kosher to begin with) and into a pyramid scheme. Those kind of things make the whole industry look bad. This, on the other hand, only makes the OP look bad, so i guess that's better. -yair On Mon, Mar 24, 2008 at 9:22 PM, Brent Davidson [EMAIL PROTECTED] wrote: LOL Reminds me of that old Ray Stevens Song - Jeremiah Peabody's Polyunsaturated Quick Dissolving Fast Acting Pleasant Tasting Green and Purple Pills Oh Yeah Binary System = Pyramid Scheme BJ Weschke wrote: I'll give you an A+ for originality after I get done laughing and then we'll still ask you to take this off list. :-) BerkHolz, Steven wrote: Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion dollar company in just 2 years. This company's compensation plan is the best and quickest that I have seen. My brother-in-law has only been in the business for a month and is already making a profit. The first thing that I noticed when researching the opportunity, was that I could find no negative statements about it. The product itself has many health benefits. So far: My knees no longer ache. We are both sleeping better. I literally do not stir once asleep. My restless leg syndrome has not been noticed. I seem to have more energy. The main ingredient is the acai berry. Here is a list of what it is supposed to do: Boosts energy levels Improves digestive function Improves mental clarity/focus Promotes sound sleep Provides all vital vitamins Contains several important minerals Is an extremely powerful free radical fighter Acai has very high levels of fibers Cleanses and Detoxifies the body of infectious toxins Strengthens your immune system Enhances sexual desire and performance Fights cancerous cells Slows down the aging process Promotes healthier and younger-looking skin Alleviates diabetes Normalizes and regulates cholesterol levels Helps maintain healthy heart function Minimizes inflammation Improves circulation Prevents artherosclerosis Enhances visual acuity The income can be made in two ways (actually more, but two primary ways) 1. Reselling the product at a marked up price. This is something that I have no interest in, and do not personally know anyone doing this. 2. Team Commissions. a. You make back 5 percent of the sales that occur below you in your tree. b. You only have to personally sign 2 people. Other people above you will be adding to your tree. c. They call it a binary system, where you only have 2 people directly under you, and any other people that you add go down to the bottom and benefit others as well as yourself. d. I already have two people underneath me and have not personally signed anyone yet, so it is a quick growing tree, even for people that may not be as motivated. e. After a month, My brother-in-law has NO more out of pocket expenses to stay in this system. The money he is earning is paying for his Minimum requirements. The rest is profit. To sign up to be a distributor , which is required to make money, is $54 A case of Mona Vie is $120. A case will last 2 people a month. (you only take 2 ounces a day) This may seem like a lot, but: 1. You will not need to buy any vitamins. 2. My brother-in-law is already making $200 a month, after being in the system for a month, So his cost for the Mona Vie is covered and he is making $80 a month. 3. As more people sign up, the amount he gets back will increase. Anyway, I am not intending this to be pushy or salesy, I just wanted to let my associates, that may be looking for additional income, know about this. Here is the Website, if you are interested in researching this: http://teamvie.blogspot.com/ http://www.monavie.com Also, feel free to Google it. I am very excited with this, both in the health benefits I am already seeing, and the income potential. Please feel free to let me know if this is something that you may be interested in, and I can get you more information. Thank You, Steven B [EMAIL PROTECTED] Please visit us on the web at www.hirotecamerica.com
Re: [asterisk-users] app_swift issues
Hi list,
just wanted to answer my own question for general knowledge - turns out
app_swift requires that the voice be set in swift.conf. If the default voice
in swift is not David-8KhZ then it will say no voice is found, even if a
swift voice is installed. Thanks to Earle for helping me with this.
-yair
On 10/22/07, Yair Hakak [EMAIL PROTECTED] wrote:
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift hello there -o test.wav and
then i play the wav and it includes the text. All good.
I've also installed app_swift according to the instructions here
http://www.mezzo.net/asterisk/app_swift.html, and show application swift
from the asterisk CLI brings up the application installed.
Now, when i put Swift('This is a test') in the extensions.conf file, i get
the following:
ERROR[3495]: app_cepstral.c:197 cepstral_speak: Failed to set voice.
I have not touched swift.conf (i'm using the defaults), and, i should add
that i have not yet purchased the cepstral voices so that when i run from
the command line it sticks this voice is unlicensed... or something like
that at the beginning of the file, if that makes some kind of difference.
I found the problem referenced here:
http://www.cepstral.com/forum/viewtopic.php?t=56sid=baa6669e9958920393c62510caa47123PHPSESSID=df1bcc629c4b8b37617d2d72c8b0232e
but no solution...
any help will be most appreciated,
thanks,
yair
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[asterisk-users] app_swift issues
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift hello there -o test.wav and
then i play the wav and it includes the text. All good.
I've also installed app_swift according to the instructions here
http://www.mezzo.net/asterisk/app_swift.html, and show application swift
from the asterisk CLI brings up the application installed.
Now, when i put Swift('This is a test') in the extensions.conf file, i get
the following:
ERROR[3495]: app_cepstral.c:197 cepstral_speak: Failed to set voice.
I have not touched swift.conf (i'm using the defaults), and, i should add
that i have not yet purchased the cepstral voices so that when i run from
the command line it sticks this voice is unlicensed... or something like
that at the beginning of the file, if that makes some kind of difference.
I found the problem referenced here:
http://www.cepstral.com/forum/viewtopic.php?t=56sid=baa6669e9958920393c62510caa47123PHPSESSID=df1bcc629c4b8b37617d2d72c8b0232e
but
no solution...
any help will be most appreciated,
thanks,
yair
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[asterisk-users] question about PSTN pickup
hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered channel an answered channel? thanks in advance for any help, yair ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re: putting 2 SIP channels together - hangup issues
Hello all, Hoping someone can help me with an issue...I have i .call file which calls out on a SIP channel and connects to an extension which dials another SIP channel. (both via voip providers) - both to PSTN. Problem is, hanging up the POTS phone doesn't release the channel (either one - hanging up the calling channel or the destination doesn't do it). Using IAX instead of SIP works better; it releases the voip channels but not the POTS channels (i.e. the POTS phones don't immediately go back to a dial tone or fast busy). I'm using asterisk 1.4 here are the relevant bits: the extension: exten= 157,1,Answer exten= 157,2,Dial(IAX2/[EMAIL PROTECTED]/PSTNnumberToCall2, 60) exten= 157,3,Hangup the .call file: # Create the call on group 2 dial lines and set up # some re-try timers # Channel: IAX2/[EMAIL PROTECTED]/PSTNnumberToCall1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Extension: 157 Priority: 1 if anyone can shed some light on this i'd be eternally grateful, first of all, why if i glue 2 SIP channels together hanging up the POTS phone doesn't release the SIP channels, and second why if i glue two IAX channels together it doesnt release the POTS lines. thanks, yair ___ --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] .call files - no hangup
hi all, i have the following .call file: Channel: IAX2/[EMAIL PROTECTED]/myPOTSline MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: default Extension: 156 Priority: 1 when i drop the .call file into the /var/spool/asterisk/outgoing/ it calls out on voipjet, connects to extension 156 (which runs the a2billing AGI) and everything is great - except that if i hang up the PSTN side, nothing happens. Only when the AGI decides to hang up does it hang up. Just for reference, extension 156 in default is: exten = 156,1,Answer exten = 156,2,Wait,1 exten = 156,3,DeadAGI(a2billing.php) exten = 156,4,Hangup anyone have any idea why a hang up on the PSTN side is not being accepted? thanks, yair ___ --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: L option in dial command
Hello all, I'm having a bit for a problem with the dial command limit option. I have the following dial command (executed from inside the a2billing agi) AGI Script Executing Application: (Dial) Options: ( IAX2/[EMAIL PROTECTED]/18005551212|30|HL(6:2:0)0) Now, from what i read in the wiki, this is supposed to limit me to one minute (6 ms), and warn me when there are 20 seconds left. Instead, it hangs up after 40 seconds. i understand there was an open bug about this...is it fixed, is there a patch, what can i do about this? for reference version is: Asterisk 1.2.13 built by root @ phone2 on a i686 running Linux on 2006-10-28 10:51:43 UTC any help is appreciated ___ --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
Fwd: [asterisk-users] How can i store PAP2 or any device config in Asterisk
aterisk does not do this, you need a provisioning server. google for pap2 and tftp. -yair On 10/8/06, ram [EMAIL PROTECTED] wrote: Hi all I have installed asterisk when any of the user device made on, it should contact Asterisk and download the config how can i asterisk does this job, does asterisk does or i should have any other server to meet my requirement Ram___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] -- Yair Hakak -Yair Hakak, CEOGo Telecom, Ltd., Israelisrael: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re: asterisk/SER integration - HELP
hi list,
i need some help here...
ihave the following setup
1. openser running on port 5060 - succesfully registering endpoints. all good.
2 asterisk 1.2 running sip on port 5070 on the same machine.
3. asterisk 1.09 running sip on port 5070 a different machine.
i have 2 routes in my openser.cfg:
1:
if (uri =~ sip:[EMAIL PROTECTED]) { log(1, Forwarding to Asterisk\n); rewritehostport(asterisk1IP:5070);
route(1); return; }
2:
if (uri =~ sip:[EMAIL PROTECTED]) { log(1, Forwarding to Asterisk\n); rewritehostport(asterisk2IP:5070);
route(1); return; }
route 2 works fine (to asterisk 1.09).
sip.conf on asterisk 1.09 looks like this:
; SIP Configuration for Asterisk;[general]port=5070 ; Port to bind tobindport=5070disallow=all ; Disallow all codecsallow=ulawallow=alawallow=ilbc
allow=gsmdtmfmode=rfc2833relaxdtmf=yestos=lowdelay
context=myContextcanreinvite=nohost=dynamicinsecure=port,invitenat=yesqualify=1000
autocreatepeer=yes
for the life of me, i cannot get SER to talk to asterisk1 (with 1.2 release). the same sip.conf doesn't work at all - asterisk completely ignores the requests(even at most verbose)- I've tried everything. if anyone has any ideas i'd be very grateful. I need to have SER talk to asterisk without defining a
sip.conf entry for every entry.
really, i'm tearing my hair out - please help.
-yair
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Re: [asterisk-users] Voip providers and sip origination and termination?
the thing to remember is that these terms are from the point of view of the PSTN. So, SIP origination is Direct Inbound Dial (DID, or DDI in european parlance) which allows callers on the PSTN to originate calls that end up at your SIP server. SIP termination allows calls which originate on your SIP server to terminate on the PSTN, i.e. to reach a non-voip line. Voip providers who provide plans are bundling 2 distinct services. Broadvoice, for example, does not expect its users to understand the terms, and just offers them what used to be called a phone line - the ability to make calls (termination) and recieve them (origination). i hope this helps, yair On 9/10/06, Christopher Corn [EMAIL PROTECTED] wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones)and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for the minute, for their outgoing and incoming calls. is there a difference in the backend architecture here? if so, what? or is this is just a difference in marketing terms and setup? for example, http://www.broadvoice.com offers an unlimited plan in the US for calls, though they never use the term sip origination and termination. they say their systems also supports asterisk. yet http://www.bandwidth.com/content/enterprise?page=voice_services_origination_terminationcampaignId=7013JBJ calls it sip origination and termination any info is appreciated! thanks! ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel israel: (972) 54 5491266usa: (212) 202 2340[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] Voip providers and sip origination and termination?
actually Rich, not to be picky or anything, but your first paragraph is backwards. There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg, Origination only provider.) That's a termination only provider which allows you to terminate calls. otherwise, very informative.. -yair On 9/10/06, Christopher Corn [EMAIL PROTECTED] wrote: thanks for the verbose explanation! Rich Adamson [EMAIL PROTECTED] wrote: Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for the minute, for their outgoing and incoming calls. is there a difference in the backend architecture here? if so, what? or is this is just a difference in marketing terms and setup? for example, http://www.broadvoice.com offers an unlimited plan in the US for calls, though they never use the term sip origination and termination. they say their systems also supports asterisk. yet http://www.bandwidth.com/content/enterprise?page=voice_services_origination_terminationcampaignId=7013JBJ calls it sip origination and termination any info is appreciated! thanks!I'll take a stab at this...There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg, Origination only provider.)There are many providers that do the above, but also will assign you a normal pstn telephone number allowing the US/World pstn users to call you (via sip, iax, etc). (eg, Origination and Termination provider.)The back end differences for the providers essentially amounts to them having to purchase multiple T1's, obtain an allocation of pstn telephone numbers, and establish a dialplan to support calls from the pstn network. The architecture for origination-only verses origination plus termination is the same; the implementation is different for one verses the other.For the most part, there are no providers that truly provide unlimited service. The majority include words in fine print that impose some sort of limit on their so called unlimited service. For example, some will say things like their unlimited service provides 2500 minutes of use; call volumes that exceed 2500 minutes will be billed at $0.02/minute. Got to read the fine print.From an architectural perspective, those providers that suggest they have unlimited service plans also impose a limit on how many simultaneous calls are allowed. The majority of these have a limit of one, two, or some very small number of simultaneous calls. There way of limiting usage since they don't really want you to use up more then their stated fine-print usage.Those providers that sell their services based on a cost per minute (as opposed to unlimited plan) do not typically limit the number of simultaneous calls. They want you to use as many minutes as possible, so why would they try to limit the number of simultaneous calls?To get the best deal possible (from any provider) you need to come up with a reasonably accurate estimate of the number of minutes of incoming and outgoing calls that you are going to make. Then, compare providers to see which ones cost the least in terms of your requirements. Keep in mind the higher your call volumes, the more competitive the providers are. In other words, if your needs suggest 1,000,000 minutes of use per month (incoming and outgoing), you should be able to find providers that will charge you something like $0.012 per minute. (Stated a little differently, the majority of service providers have other unpublished plans that are discounted based on your expected level of usage.) Most providers are trying to pattern their plans based on how well the Cell providers have done in the past. You and I typically sign up for minutes of cell phone usage, but don't actually use all of those minutes. What's our real cost per minute in this case? And, how often do we make useless cell phone calls because we have free minutes left?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED
Re: [Asterisk-Users] asterisk sip listening port
in the [general] section of sip.conf bindport=5062 well documented here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf -yair On 6/23/06, Khaled Chehab [EMAIL PROTECTED] wrote: How I can let asterisk listen only at port 5062 since I have ser on the same machine listening to port 5060 , Please from where I can configure it *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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cant get voicemail
run asterisk in verbose mode (-vv) and see if the digits are being properly picked up. A common problem is a DTMF type mismatch, so the keypresses may not be getting to the server. -yair On 5/1/06, Jim Lynch [EMAIL PROTECTED] wrote: I've enabled voice mail for extension 200 in the extensions menu, and I've set the password to 1234. When I dial *97 which is listed as Your messages in the applications menu, it says Password I enter 1234 and it says, login incorrect, Password so what am I missing? Thanks, Jim.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk or ser
hi, SER is less about the number of callers than it is about the number of registered sip clients. Without NAT issues a pizza box server with SER can essentially register an unlimited number of SIP clients. With larger numbers of SIP clients i find SER handles them much better than asterisk. Now, i know this is unorthodox, but i route EVERY call through asterisk, even calls between SER users, for a few reasons: 1. billing - asterisk is much better at keeping CDRs 2. call control - asterisk can stay in the media path if neccesary, SER won't (by default, although you can use a b2bua), and for things like prepay calling cards this is a neccesity. 3. significantly easier to use dialing logic. on the down side, i still havent gotten my old setup with autocreatepeer=yes in my sip.conf and rewritehostport in ser.cfg working on 1.2 - anyone have any ideas about that? hope this helps, -yair On 4/14/06, Xaji Gaid [EMAIL PROTECTED] wrote: Hello:I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic. Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated. -Gaid___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: voipjet
anyone else having issues with voipjet? i am getting nothing but dead air and a hanging iax channel. -yair ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: questions about sip requests to asterisk 1.2
thanks for the answer. is this something new in 1.2? if so, where is it documented, and what is the point of autocreatepeer=yes if this is the case? -yair On 2/5/06, C. Zerbo [EMAIL PROTECTED] wrote: you need to setup a asterisk peer at port 5070 in sip.conf to get the callreplying correctly to ser. Cheick Zerbo Corbimas.com [EMAIL PROTECTED] From: Yair Hakak [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comTo: Asterisk Users List asterisk-users@lists.digium.comSubject: [Asterisk-Users] re: questions about sip requests to asterisk 1.2Date: Sun, 5 Feb 2006 14:55:32 +0200 hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport(myIP:5070); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. ifin ser i was sending to [EMAIL PROTECTED], it will make it [EMAIL PROTECTED]:5070, and asterisk will match it to 151 in the dialplan. In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep showsa not found returned toSER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want toupgrade but I don't want to lose thisfunctionality. thanks for any help, yair ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: questions about sip requests to asterisk 1.2
hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport(myIP:5070); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. ifin ser i was sending to[EMAIL PROTECTED], it will make it [EMAIL PROTECTED]:5070, and asterisk will match it to 151 in the dialplan. In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep showsa not found returned toSER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want toupgrade but I don't want to lose thisfunctionality. thanks for any help, yair ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: questions about sip requests to asterisk 1.2
Hi Jean-Michel, have you tried upgrading? can you confirm this behavior? It seems to me this is a major issue for those of us running SER + asterisk, and who dont want to configure each SIP client in SER and asterisk separately. -yair On 2/5/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote: In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a not found returned to SER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality.Since I use 1.0.9 and use exactly the same scheme, I am interested onhow to upgrade as well.Cheers,Jean-Michel.--Jean-Michel Hiver - http://ykoz.net/Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: help with redirect from SER
hello all,
i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to
1.23 requests from SER to asterisk die quietly, no matter how verbose my asterisk is. It's as if the requests dont exist at all.
My setup is as follows: asterisk and SER on the same box, SER running on 5060 and asterisk on 5070.
All i want is a simple redirect from SER to asterisk, in ser.cfg thusly:
if (uri == sip:[EMAIL PROTECTED]) { log(1, Forwarding to Voicemail\n); rewritehostport(myIP:5070);
route(1); break;
}
and in SIP.conf (this is what i have after some hours of trying, but it doesnt seem to be helping):
bindaddr=myIP
bindport=5070disallow=all ; Disallow all codecsallow=ulawallow=alawallow=ilbcallow=gsmdtmfmode=rfc2833autocreatepeer=yesinsecure=port,invite
[SER]type=friendhost=myIPfromdomain=myDomaincontext=mycontextcanreinvite=noinsecure=very
if anyone can help i'd me most grateful. I originally thought it would be as simple as changing port to bindport in sip.conf. Oh, how wrong i was.
thanks,
yair
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Re: [Asterisk-Users] MOH begin behavior
hi, why cant you just playback what you want to play specifically before going to MOH, i.e. exten = 6000,1,Answer exten - 6000,2, Playback() exten = 6000,3,MusicOnHold() sorry if i'm missing something... -yair On 1/24/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello All, Does anyone know if you can start an MOH queue on an individual call? What I mean is, for example if you have a script that you want the moh to start with certain phrases, can it be done, or are you limited to the standard looping audio? It's almost like starting a stream for each call, and terminating it when the call comes off of hold. Regards, Greg ___ --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] Budge Tone-100 as a Ext in the LAN
lukeuse the wiki. (always wanted to do that) http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone hope this helps, yair On 1/6/06, luke devon [EMAIL PROTECTED] wrote: HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Manual
what is your question? you must set up extensions yourself in extensions.conf...you can set the extensions to whatever you want (say, if you are replacing an existing PBX and want the users to have the same extensions). -yair On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote: *411 Directory *43 Echo Test *60 Time *61 Weather *62 Schedule wakeup call *65 festival test (your extension is XXX) *70 Activate Call Waiting (deactivated by default) *71 Deactivate Call Waiting *72 Call Forwarding System *73 Disable Call Forwarding *77 IVR Recording *78 Enable Do-Not-Disturb *79 Disable Do-Not-Disturb *90 Call Forward on Busy *91 Disable Call Forward on Busy *97 Message Center (does no ask for extension) *98 Enter Message Center *99 Playback IVR Recording *666 Test Fax Simulate incoming call - Original Message - From: Vladimir Montealegre [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:34 PM Subject: [Asterisk-Users] Extension Manual in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? 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 __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: problem with asterisk and SIP on same box with 1.2
hello all, having a little problem.. asterisk and ser on the same box, SER on 5060 and asterisk on 5070. SER is set up to forward everything to asterisk. in 1.07 my sip.conf looked like this: [general] port = 5070 ; Port to bind to disallow=all; Disallow all codecs allow=ulaw allow=alaw allow=ilbc allow=gsm dtmfmode=rfc2833 context=myUsers canreinvite=no host=dynamic insecure=no nat=yes qualify=1000 autocreatepeer=yes and incoming SIP requests flowed to asterisk. now, it's failing, silently, with nothing in the CLI (at v). ngrep the sip packets show SER trying to forward the packets along and failing. anyone have something similar or have any tips? do i need to add insecure? thanks, yair ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ip phone
look, you get what you pay for. excellent value for the price, but i've found they need more handholding than others (sometimes they need to be rebooted, they freeze up, etc). i'm phasing out in favor of pap2 units and analog phones. i've never had a problem with audio quality, however, audio quality with other devices is noticably better. -yair On 11/18/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Fri, 2005-11-18 at 15:04 +0700, stevanus wrote: Hi, Maybe grandstream budgetone 100 series will fulfill your requirement. It's very good for such a cheap sub-50 phone. Once, I've tested and I've found myself that it's a good performer (even it has compatibility problem with old switch in my office :P) You can search the supplier through googling it. Don't ask me as I don't know any information about it. I have heard bad things about that phone. Specifically audio quality is questionable, the power connector that ships is the wrong size so it tends to fall out, there are firmware issues that locks the phone up, etc. Does anyone have any experience with that phone specifically? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQBDfZnX+1olxlzQw5cRAio1AKCYSsEAlVOLrGCFTyHWNwJyMPDZLwCfeVKk B8cwwSFLC6Acs1eH4qV4Axg= =ITZY -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: compile error
hi all, compiling 1.2 from CVS i get the following error in asterisk/apps make[1]: Entering directory `/usr/src/asterisk/apps' Makefile:14: *** missing separator. Stop. I looked at the makefile and i dont see anything glaring, but then again it's been a long time since i wrote any code. tried to compile 1-0 instead of 1-2 and got the same error. running Asterisk CVS-v1-0-07/23/05-21:30:43 on RHEL, compiled from CVS with no errors. anyone have anything similar? thanks, yair ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Wireless SIP Phones with Asterisk
i am very happy with my Zyxel P2000Wv2. the latest firmware solved all the problems (there were some NAT issues.) i'm running SER in front of asterisk. all good, except that it appends the port to sip requests and i had to put config in SER to handle that. sometimes there's a huge echo, but i'm relatively sure that's because of a bad Wifi connection and not anything wrong with the phone. -yair On 11/17/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, If you are using multiple wireless access points use the Zyxel P2000Wv2 with latest firmware. It has the shortest hand off time between access points. Info world did a test and the Hitachi did not do well. Thanks Juan Janczuk wrote: Hi, list. I'm looking for a (couple of) wireless (802.11b) IP phones with SIP support. At voipsupply, I can't find any with the works with asterisk logo. Any tried some Wireless IP phones with Asterisk? Comments, recomendations? Thanks in advance. Regards. Juan. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 16/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: a2billing /areski help
hello all, a slightly off topic question...i've successfully installed the AGI and admin and user interfaces of a2billing. everything seems to be working fine. thanks to areski for a very nice piece of software. however, since i'm not familiar with the calling card industry, all the talk of ratecards, tarrifs and trunks is greek to me. is there any documentation on the setup of the features (not the setup of the software, for that the guide on areski's site is great.) i.e. how to set up cards and successfully place calls with the cards? thanks, yair ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: changing protocols and transcoding
Hello all, forgive me if this is a simple question, but does bridging a SIP channel and an IAX channel that use the same codec (say, alaw) involve transcoding? i'm trying to figure out what kind of hardware i'll need, and i'm going to be using SIP endpoints and IAX trunking to move the audio along to another asterisk server(all with alaw), and i want to know if i'm going to need to figure transcoding into my hardware. I'm not familiar with the internals of IAX/SIP soagain forgive me if this is a dumb question. thanks, yair ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SER and Asterisk
hello, trace the SIP packets and see if they are actually addressed to 5062. if you post the ngrep or ethereal dump we'll see whats actually going on. I do this with SER on 5060 and asterisk on 5070 and there are no problems - my extensions point to 5060 and my DID's point to 5070 so asterisk servesas the gateway to the PSTN. -yair On 10/19/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: I have on one machine Openser and Asterisk. Since Asterisk was first, Ilet it have the port 5060 ;-) I have choosen for Openser the port 5062.I tried several hard and soft phones to connect to ser to the port 5062,however each of the phones tries to connect to asterisk.I am totally confused about that, what could redirect all requests to port 5060.(I could not get any answer from ser nor openser mailing list, maybe Iam lucky with a hint here)byeRonald Wiplinger___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SER and Asterisk
On 10/19/05, Yair Hakak [EMAIL PROTECTED] wrote: i do it this way because i want all the dialplan logic and CDR having to do with PSTN in asterisk, not SER. so, calls from the outside are adressed to [EMAIL PROTECTED]:5070 and hit asterisk. asterisk either sends them along to 5060, or handles them internally (IVR, voicemail, etc) based on the dialplan. clients on the inside are registered to the SER at 5060 and the SER automatically forwards them to asterisk. if they are PSTN asterisk serves as PSTN gateway, if they are internal, asterisk native bridges and drops out, but still keeps the CDR (i have full SIP addresses in my dial statements instead of asterisk SIP peers) the reason i do this is i found that if the endpoints are scattered on the internet, SER+rtpproxy is much more stable than asterisk as a SIP server (asterisk kept dropping endpoints). This way SER serves as a completely dumb SIP server, and just sends everything along. there is a minimal increase in overhead (i could handle internal calls just with SER) but it's worth it to have all the dialplan logic and CDR's in one place. also, obviously, if i use an IAX provider for outgoing, asterisk has to be in the middle. i agree though, it makes more sense to have SER on 5060 and asterisk somewhere else. hope i'm making some sense, please point out if i'm doing something really stupid. -yair On 10/19/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote: hello,trace the SIP packets and see if they are actually addressed to 5062. if you post the ngrep or ethereal dump we'll see whats actually going on. I do this with SER on 5060 and asterisk on 5070 and there are no problems -my extensions point to 5060 and my DID's point to 5070 so asterisk serves as the gateway to the PSTN. -yair also look for dns packets and see if htey are pulling the server info.Some sip clients look for specific server type dns records to see where they should go.5060 is the default, wouldnt it make more sense to have the default port be what you want the devices to goto and have that proxy to the deviceyou dont want direct connectivity to?Or am I missing something in that --Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDVg5f+1olxlzQw5cRAhl5AJ91lwjqMb2EPcDSXH69dOELBOq0IQCgvr8m4NqQAGLmWLokUXjl7Bi7SbI==thAz-END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Vonage-type service
I dont know about others, but i find that SER as a SIP proxy in front of asterisk works much better for endpoints on the public internet than just asterisk as a sip proxy. my 2 cents. -yair On 9/26/05, Federico Alves [EMAIL PROTECTED] wrote: I want to share some facts with the Asterisk community. I have been verysuccessful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage isVoyze.com. The key concept is that Asterisk works like a Cisco, for all theintelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The configuration files, like sip.conf, are downloaded from SQLServer, where they are generated and modified by triggers that execute inseveral tables. The Management GUI is simply an application that modifies SQL tables, and so does the Web application, for the end customer. Both arewritten with Microsoft Visual Studio 2003. It works perfectly, is scalableand very cheap to maintain. I use freetds and UnixODBC to link both worlds, Linux and Windows 2003 Enterprise.We don't sell the system. We provide a full independent system for customersincluding co-location, for a setup fee and 1/2 cent per call, regardless oflength. We also provide US termination via our own DS3 for 1.3 cents aminute, and it does support T.38 faxing.Federico Alves___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Register Today for Fall 2005 VON: The Destination for IP Communications
go to pulver's blog, there's a free code. -yair On 8/23/05, Dean Collins [EMAIL PROTECTED] wrote: Anyone able to get me a comp/highly discounted ticket to this? $150 just to visit the exhibition halls sounds crazy? Dean -Original Message- From: Jeff Pulver [mailto:[EMAIL PROTECTED] Sent: Tuesday, 23 August 2005 11:47 AM To: mailinglist1 Subject: Register Today for Fall 2005 VON: The Destination for IP Communications Hi There, While flying to London yesterday, I spent some time thinking about VON and how while some things change, other things about VON remain the same. Since our first VON event in the Spring of 1997, our VON events have over time become the worldwide Destination event for IP Communications. In fact, while we are actively marketing Fall 2005 VON using various channels around the United States, it is the continued strong word-of- mouth buzz that is bringing in delegates from around the world. So far, there are delegates registered from 40+ countries including: Argentina, Aruba, Australia, Austria, Belgium, Brazil, Canada, Chile, China, Costa Rica, Denmark, Dominican Republic, Finland, France, Germany, Ghana, Hong Kong, Hungary, India, Ireland, Israel, Italy, Japan, Korea, Mexico, Netherland Antilles, Netherlands, New Zealand, Norway, Russia, Singapore, Slovenia, South Africa, Spain, Sweden, Switzerland, Taiwan, Turkey, UK, UAE, USA and Uzbekistan. I expect the buzz to be pretty strong when the doors open in less than four weeks. The 330+ exhibitors in our Sold Out exhibit hall represent our largest exhibit hall...ever! (and has grown by more than 100 exhibitors since Spring 2005 VON.) The Fall 2005 VON conference sessions are returning to the size we experienced five and six years ago. The registered delegates in Boston are all part of the ecosystem that makes up our VON events. There will be people representing just about all aspects of the IP Communications food chain. Note: Vendors who are interested in exhibiting at Spring 2006 VON should consider signing up now. The pulvermedia Sales team is projecting that the exhibit hall at Spring 2006 VON will be close to sold-out before we arrive in Boston for the commencement of Fall 2005 VON. Experience the Journey and register today for Fall 2005 VON, The Destination for IP Communications. Please visit: https://secure.pulver.com/von/register.html to register. Best regards, Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: slightly OT
hello, please 'scuse the slightly offtopic question, but i see a lot of posts about the adit600, used as a channel bank, but from what i understand it can be used as a PRI interface as well. If anyone who is using the adit600 to interface to 4 T1/E1's has feedback, i would appreciate it, specifically involving the asterisk interface, echo, and DTMF. thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP exten to PSTN calls
post your dialplan, it's pretty safe to say that's where the problem is. without it, there's no way to help you. -yair On 8/16/05, Appan KH [EMAIL PROTECTED] wrote: Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP exten to PSTN calls
I am confused. what do you expect to happen when you call the PSTN? let's say you call 023459823 (assuming you are in a country where dialing codes begin with 0) first of all, why do you have 2 lines that match the same extension and tell asterisk to do different things? I am referring to these 2: exten=_0.,1,Dial(Zap/1/SIP/197,20,tT) exten=_0.,1,Dial(Zap/1/SIP/198,20,tT) Let's say the second one is operative. I dont understand how this is supposed to dial out on the zap channel. take a look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf this should help you sort things out. Personally, i would define an incoming context and an outgoing context separately, but that's just a preference. -yair On 8/16/05, Appan KH [EMAIL PROTECTED] wrote: The Dial plan is given below [incoming] exten = 197,1,Dial(SIP/197,20,tr) exten = 197,2,Hangup exten = 198,1,Dial(SIP/198,20,tr) exten = 198,2,Hangup exten=_0.,1,Dial(Zap/1/SIP/197,20,tT) exten=_0.,1,Dial(Zap/1/SIP/198,20,tT) appan kh - Original Message - From: Yair Hakak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 16, 2005 11:45 AM Subject: Re: [Asterisk-Users] SIP exten to PSTN calls post your dialplan, it's pretty safe to say that's where the problem is. without it, there's no way to help you. -yair On 8/16/05, Appan KH [EMAIL PROTECTED] wrote: Hi, I had configured Asterisk with the following 1). X100P - Card 2). Two -Greadstream100 SIP Phones. I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn. But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised. The SIP extn is not sending the correct number. I will be thank full if some solutions is suggested. appan kh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Need some statistics facts
According to the CIA world factbook there are 800 million landlines in use and about 6.4 billion people. This makes more sense than 800 billion. there are probably at least an equal number of cellular telephones in use as well, but i have no idea how one would go about getting those numbers (except maybe taking large countries and starting to add). http://www.odci.gov/cia/publications/factbook/geos/xx.html -yair On 8/10/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Dean Collins wrote: Ronald, Why? What do you need it for? For a power point slide. Would the statistic or the facts are different if I would need it for a report ? hehehehehe bye Ronald Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Tuesday, 9 August 2005 10:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need some statistics facts There is nothing more than figures, right? I am looking for: 1. How many phone lines are currently in this world (I estimate 800 billion analog lines) 2. How many data lines are currently in this world (I estimate 1 billion) 3. what is the forcast of 1 2 ? When will it swap? 4. How many PBX are in use? I guess that a part will be extended with FXO to the Interent. Some will be replaced at all, because of missing features. 5. Which countries favour VoIP? Which countries forbid VoIP? Where are the cheapest / best ADSL lines available? Hongkong, Korea, Japanis my guess bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: switch statement in dialplan
hi all, is there a switch statement in the dialplan? or do i have to daisy-chain GoToIf statements? i don't see a switch statement on the wiki, but you never know... thanks yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] slightly OT: firefly won't hang up!
hello all, i have a strange problemi am running SER in front of asterisk, and am testing softphones. x-lite works fine...i can dial, hang up, DTMF, all good. Firefly looks really cool and i'm very impressed with the IM-like interface and the skinning ability, but something strange is happening...when i call from the firefly and run something on the server and press hangup on the client, it hangs up immediately and i see it in the logs. When i try to call another channel (either a sip hardphone, i.e. 2 sip channels, or an IAX trunk to a voip provider, i.e. 1 sip and 1 IAX) firefly does not hangup even if i press hangup, and i have a channel, but no audio gets to the firefly client. anyone come across this? any hints? thanks for any assistance, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF with Asterisk as SIP client
Hello, I have the following setup: sip phones -SER - asterisk - voip provider1 - voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my bad DTMF (usually 2 keypresses for every 1)...transport is via SIP, i am registered in sip.conf with a register statement (i.e. asterisk is a SIP client) and ulaw and alaw are the first allowed codecs. When i set dtmf as info or RFC2833 i don't get any tones, and when i set inband i'm back to bad DTMF. if i call into the extension from one of my sip phones (i.e. not via voip provider) and interact with the menu (put in my authentication and dial the onward number) it works fine. anyone come across this? any tips on how to solve it? any help is appreciated, thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: DTMF woes, continued
hello all, I have a DID from nufone, transported via SIP to my * box, and even though i'm using rfc2833 DTMF i'm still getting double digits and all sorts of other stuff... sip.conf is as follows: [general] port = 5070 ; Port to bind to disallow=all; Disallow all codecs allow=ulaw allow=alaw allow=ilbc allow=gsm dtmfmode=rfc2833 register = username:[EMAIL PROTECTED]/myDID just to make sure everything is set properly, i even threw in exten = myDID,3,SIPDtmfMode(rfc2833) in extensions.conf to make sure that dtmf is coming over in rfc2833 and not inband. if it helps, the provider in question is nufone and jeremy from nufone told me to use only rfc2833. anyone else have this problem? any pointers or ways to solve this? it's making me insane...if anyone has working DTMF on incoming SIP from nufone, and can send me configs, that would be great. thanks for any help, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: help debugging dialplan
hello all,
another desperate request for help debugging my dialplan...
from a certain extension i do the following:
DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM})
a NoOp to the console says
DBput: family=CFIM, key=2122022001, value=2122022001
and database show says
/CFIM/2122022001 : 2122022001
so far, so good.
but in a macro, when i try to get the data,
exten = s,1,DBGet(${DB(CFIM/${ARG1})
(ARG1 is 2122022001)
first, i get the following:
Jul 6 18:50:14 NOTICE[587]: pbx.c:1114
pbx_substitute_variables_helper: Error in extension logic (missing
'}')
and the CFIM variable is empty.
so, the following questions:
1. where does the } go? i know i'm missing one, but i don't know what to enclose
2. why isn't CFIM getting the variable from the DB?
anyone who can help me, i very much appreciate it.
thanks,
yair
p.s. when are DBGet and DBSet being deprecated?
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Re: [Asterisk-Users] re: another database question
hi ferdy,
again, thanks for all your help. I will try this and report back.
as for your questions:
1. my version is from stable, Asterisk CVS-v1-0-01/22/05-12:27:04
2. the line used that gets this database result is:
exten = 154,2,DBPut(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
which is, of course, wrong. i'll fix that and i'll let you know how
everything works.
thanks again for all the help,
yair
On 7/4/05, Ferdy Riphagen [EMAIL PROTECTED] wrote:
Yair,
When you have an older version you can try to use DBput/DBget (if still
working, because set will replace it in CVS)
Set(DB(CFIM/${CALLERIDNUM})=u${CALLERIDNUM})
will be;
DBput(CFIM/${CALLERIDNUM}=u${CALLERIDNUM})
Set(CFIM=${DB(CFIM/${ARG1})})
will be;
DBGet(${DB(CFIM/${ARG1})
normaly a database entry looks like:
CFIM/999 : 999
What is the line you use to fill the database with: /DB(CFIM/999) : 999 ?
What is the version of asterisk your machine runs?
Regards,
/* Ferdy */
http://asterisk.nsec.nl
info(AT)nsec(DOT)nl
Yair Hakak wrote:
hi ferdy,
i did check your first post to the list, and i really appreciate your help.
however, when i run your code i get an error because the set
application is not recognized - perhaps it is a CVS-head thing?
thanks,
yair
On 7/3/05, Ferdy Riphagen [EMAIL PROTECTED] wrote:
Yair,
Check my first post to the list, about your other question (call
forwarding, most basic case)
SetVar will be removed (I heard)
Greetz,
/* Ferdy */
Yair Hakak wrote:
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following statement:
exten = s,1,SetVar(CFIM=${DBGet(CFIM/${ARG1})})
i get the following:
Executing SetVar(Local/[EMAIL PROTECTED],2, CFIM=) in new stack
any ideas why the CFIM variable is not getting the 999 value?
thanks for any help,
yair
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[Asterisk-Users] re: another database question
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following statement:
exten = s,1,SetVar(CFIM=${DBGet(CFIM/${ARG1})})
i get the following:
Executing SetVar(Local/[EMAIL PROTECTED],2, CFIM=) in new stack
any ideas why the CFIM variable is not getting the 999 value?
thanks for any help,
yair
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[Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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Re: [Asterisk-Users] call forwarding, most basic case
hello Mike,
we are talking about very different things here. please look at my
original mail again. I want the call recipient to be able to toggle on
and off do not disturb. I don't want the phone to ring at all.
thanks,
yair
On 7/3/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines (cell,
remote offices, etc..), although I would guess the same thing applies to SIP
phones. The dial plan variables are only necessary if you want to pass
caller ID from the originating caller through to the forwarded number. If
you don't use the variable then the caller ID you would see would be that
from the Asterisk configuration and not from the actual caller.
The 0 inserted into the number is helpful if you have calls forwarded
simultaneously to your cell phone (or other) so that you can see by the zero
that it is a forwarded call rather than a direct call to your PSTN number (I
guess you could also use this with internal calls to distinguish calls that
are forwarded from different extension numbers). If it is a forwarded call
then by not answering it, it would go to Asterisk VM. If a direct call, it
would go to whatever aswering funtion is set up on your cell phone (or other
PSTN phone). [Please reply through the mailing list]. Mike.
-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case
hi,
thanks for your answer, but i'm not sure i understand. this dialplan says
1. call the extension
2. set a variable with the callerIDNum
3. dial out to the follow me number with a 0 prepended to the callerID
4. switch the callerID back to the original
5. go to voicemail
how does the user turn this on and off? that's what i'm trying to do
in my case. i want the user to be able to switch between asterisk
calling his extension and asterisk sending the call directly to
voicemail.
-yair
On 7/2/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
I used and it works well. Rather than segregate calls based on caller ID,
it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i
was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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Re: [Asterisk-Users] voice mail problem
I believe this may solve your problem, http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone works for me. -yair On 6/30/05, Betül Gözlükoğlu [EMAIL PROTECTED] wrote: Hi; Have a BUDGETONE-100 and using it with asterisk…Problem occurs when I dial message center…Message center does not accept tones (password for example) that I dial, Behaves as I do not dial any number and asks for the password again…Changed the DTMF Mode from in-audio to RTP(RFC2833) it works but this time, dialing internal numbers over telephony system is denied… Does anybody has any idea about correct configuration on Asterisk or Budgetone? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] LiveVoip is Bankrupt
well, i can't say i'm surprised. any company whose approach to customers is you are all scum trying to cheat us, don't ask questions, and we'll help you when we feel like it isn't going to be around for a long time. On 6/26/05, Andres [EMAIL PROTECTED] wrote: So it looks like Livevoip went Bankrupt --- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions you may contact our lawyer - Customers and Creditors are now under a U.S. Court Ordered Stay NOT to have any contact with LiveVoip LLC Management. LiveVoip LLC has Ceased Operations and Filed for Bankruptcy. This action was taken after the company was unable to resolve issues with carriers over billing, mass credit card fraud, suppliers not delivering on what they had been paid for among other things. A Stay Order is in effect at this time and all questions must be directed to our company lawyer. Creditors will be hearing from the Courts in due course. LiveVoip LLC is no Closed. United States Federal Bankruptcy Court District Montana Case: 05-62057 LiveVoip LLC Company Lawyer: Robert Kampfer Esq. 406.727.954 The LiveVoip network is offline. An Update will be issued on our main website. The trouble ticket server is also having its own problems. Please watch our main for site for complete details. LiveVoip LLC --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] SER and Asterisk question
yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 with ser to share the load
Hello Deepak, 1. don't post multiple times. it's annoying. enough said. 2. run asterisk in verbose mode (start it with asterisk -vgc), place a call from a SIP endpoint behind SER to the asterisk server, and see what happens in the asterisk CLI. 3. if you don't see anything there, get ngrep and place a call from the SIP endpoint while running ngrep SIP and post the output. 4. are asterisk and SER on the same machine? 5. if all else fails put autocreatepeer=yes in your sip.conf - this has bad security consequences, but it is useful for debugging. -yair On 12/2/04, Deepak Dhiman [EMAIL PROTECTED] wrote: Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] DID ~ Extension
Hello, this is not automatic, you need to set up the proper dialing rules. the fact that a DID dumps a call into the system and that there is an extension with the same numbers do not mean they will be automatically connected. post your config files... On 4/19/05, Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote: Hi, does anyone ever tried assigning a DID to an extension of similar value? Example: Extension 6945199 DID 6945199 It doesn't seem to work in my system. Please advice. Thanks. Cheers, Angelo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] query about cdr configuration
Hello Deepak, yes, you can use mysql. the packages are in asterisk-addons. there is a very good wiki page on the subject here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql hope this helps, yair On Apr 6, 2005 7:33 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: hi friends ! can anybody tell me something about cdr configuration. actually i want to confirm about the minimum requiremnts. is it possible to configure it with mysql server and myodbc anly or unixodbc is also required? in case unixodbc is also requied than help me to send some download links that already have worked well. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Instalation
I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Instalation
Jeez, people...learn to take a joke. I offered to help the man. I didn't make cracks about googling or anything like that. I said explicitly that if he posted details we'd try to help. I didn't insult him, call him a worthness noob, or otherwise offend him. IT WAS JUST A JOKE. Is this different from the post a few days ago about not using enough magic? And, as for emailing people direct and offering to help them i was under the impression that it doesn't work that way...that there is value in the archives, and if he posts his setup and problems and a solution found then that helps everyone. To you, Trevor, if i said something offensive inadvertantly i apologize and i hope we can help you. To Brandon and Randall, i suggest you both try to see more humor and less insult. -yair On Fri, 25 Mar 2005 01:44:54 -0700, Brandon Patterson [EMAIL PROTECTED] wrote: If we need a dose of Smart Ass, it's always good to know it's available here on this list. The person is new and he is asking a question. You could have emailed him direct, asked for more detail and helped him. Rather than be kind you posted dribble. Daily I speak with people like Trevor. If you do not have something positive to say just do not say it. Brandon Patterson LiveVoip LLC I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Instalation
Hi, a few pointers: 1. the wiki is your friend: http://www.voip-info.org/wiki-Asterisk lots of good stuff and good documents for getting started. If i were you i might reinstall asterisk from CVS just to make sure you have the latest version, and because this way you can learn about installing. In general asterisk usually runs on a machine without x-windows so doing everything from a command-line window might also help with the learning, rather than being dependent on prepackaging. 2. did you install asterisk with the example configs (make samples)? asterisk is dependent on a directory of config files (usually /etc/asterisk) and make samples populates the dir with some basic config files. 3.make sure that the x-lite that you want to register is defined in sip.conf and extensions.conf 4. configure x-lite using the following: http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf 5. run asterisk in verbose mode (asterisk -r vgc) so you can see what's happening You're in for quite a learning experience...hope this has been helpful. good luck and good hunting. -yair On Fri, 25 Mar 2005 20:21:57 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, Thanks for the wonderful advice, and comments, and anything I might of missed, and no offence taken. Yes I am new to this program, and Linux too, so this is a big learning curve. I installed the software Asterisk which I believed it did straight from the cd, rebooted the computers, and it installed more stuff. I am then lead to believe that I can use x-lite a phone interface, I guess, to interact with the new pabx-asterisk system I now have. I can see from the gui interface that I am trying to make calls, but that's about it, not much else is happening Well if you want to know much more, please ask, as I have no idea what I am doing :) You help and direction would be much appreciated. Cheers Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Need some help
Hello, what is the benefit of your scenario #2? I'm not understanding what it adds for you... -yair On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED] wrote: Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. Thanks for any advice Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Need some help
Duh, i'm an idiot. I meant scenario #1. -yair On Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak [EMAIL PROTECTED] wrote: Hello, what is the benefit of your scenario #2? I'm not understanding what it adds for you... -yair On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED] wrote: Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. Thanks for any advice Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] CDR database
http://www.voip-info.org/wiki-Asterisk+billing On Fri, 11 Mar 2005 19:58:37 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote: I am looking at AMP and read All the graphic reports are based over the CDR database. How do I get the CDRs into a database? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone + VoIPJet = busy busy busy
the following is on voipjet's site: Please note we are having a temporary glitch with our New York location. Please send traffic to our West Coast Premium Server until the problem is fixed sometime today. New SERVER IP: 69.25.60.30 although i guess an email to this effect would have been nice. -yair On Wed, 09 Mar 2005 17:41:07 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] NAT Far End Traversal
Hello, i'm using ser+nathelper+rtpproxy in front of asterisk. It has been terrific. The only problem i have is with some DSL modems that grab port 5060 for themselves (why, i don't know, it's very annoying but easily solvable). Other than that, no issues at all, in the NAT, in the DMZ, between the modem and the router, all good. You can also look into ser+stun in front of asterisk. Or, you could just use IAX :-) -yair On Tue, 08 Mar 2005 17:49:36 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody gotten this working in conjunction with Asterisk? Another question... Are you aware of a SIP ATA or phone that has some kind of VPN (i.e. PPTP) client embedded in? This would make the NAT problem go away nicely and provide added security... Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Re : Calling card platform
See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Re : Calling card platform
I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] CDR
hi, you need to tell us how you're saving your cdr's - database, csv, whatever?- if you're saving to a database a stored procedure is probably best, unless you want to change the SQL statements in the proper module. yair On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A [EMAIL PROTECTED] wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Re : Calling card platform
It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Re : Calling card platform
I do not recall calling anyone a cheat or a fraud. We have a saying where i am from, something about a burglar, and a hat, and fire. I'll leave it at that. As for your last question, i can't answer that. -yair On Thu, 3 Mar 2005 10:26:05 -0500, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: Yair, I have been dealing with Amarfone as well as Ehsanul Karim for an year now and I never had any issue with them. Both were and have been customers. Ehsan is an honest individual. He might have omitted mentioning that he worked for Amarphone in the past. It does not make him a cheat or a fraud. I recommend IBM Servers and Citibank Checking Account as the best in their products and services, Servers and Banking. I worked for both of them in the past and I knew them. Am I a dishonest person? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Thursday, March 03, 2005 9:51 AM To: M. Ehsanul Karim; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re : Calling card platform It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/ 064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full
Re: [Asterisk-Users] CDR
Hello, Nir's suggestion seems to be best...is there a specific reason you don't want to save certain CDR's? Better to save everything and pull out what you need when you need it. -yair On Thu, 3 Mar 2005 07:33:03 -0800 (PST), R A [EMAIL PROTECTED] wrote: sorry i´m using MySQL database. there are something else that you need to now?? wert Yair Hakak [EMAIL PROTECTED] wrote: hi, you need to tell us how you're saving your cdr's - database, csv, whatever?- if you're saving to a database a stored procedure is probably best, unless you want to change the SQL statements in the proper module. yair On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSC RIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 + SER
it depends what you mean by billing and accounting. postpaid? prepaid? integrated into the dialplan or just for use later? you can use cdr_mysql or similar to dump everything into a DB and build billing apps on that, if you want as well. please read the stuff here: http://www.voip-info.org/wiki-Asterisk+billing most of the billing functions are very well documented. -yair p.s. the reason i said i would do the opposite of your suggestion is that SER is a better SIP proxy server than asterisk (it scales better, among other things). The downside is that the routing logic is more programmatic - i.e. extensions.conf is much simpler than ser.cfg, and there's also no handy nat=yes flag - you need rtp_proxy and nathelper, to get past NATs. I use asterisk as my PSTN gateway as well as handling all the dialing logic in asterisk, and SIP just takes care of registering endpoints. hope this helps. On Mon, 28 Feb 2005 10:13:20 -0800, Nitesh Divecha [EMAIL PROTECTED] wrote: Hey Thanks guys... But how can I use Asterisk for billing and accounting? Do you mean use the astcc module..? Please help... Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Saturday, February 26, 2005 11:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk + SER Yes, I use this method too. On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak [EMAIL PROTECTED] wrote: you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 + SER
you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] List tips for new subscribers
ok, not that i'm such an expert myself, but 1. there's a big difference between newbies asking specific question and the i want asterisk to run my life, make me coffee, and solve my problems, does asterisk do that? questions that are appearing lately. I'm not a member of the list police and they annoy the hell out of me. 2. many of the list police are active in the development process well, so your remarkably clever comments about the lack of help are uncalled for and untrue. People will help you, but they won't hold your hand. If you want your hand held, then hire a consultant. 3. get a gmail account and your search issues on the mailing list are over. In addition, the remarkable new gmail system doesn't mangle your email with HTML tags, rendering them readable to all. how revolutionary. the downside is, no smileys. (oh the horror.). 4.almost everyone here has been quite helpful. once or twice i didn't follow netiquette (posted once without a subject by mistake) and quite rightly got called for it. If your ego is so fragile a dressing-down on a email list from people you don't know bothers you, you have issues. And specifically in this case, recyclying a subject line that has nothing to do with your email is just lazy and screws up threads. seriously, get over it. -yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: difference between STUN servers and far-end solutions
Hi asterisk list, this is a bit off topic, but can anyone explain the point of the commercial far-end solutions floating around (jasomi, for example)? or are the far-end things just hyped up media proxies? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short, what advantage does such a setup have over, for example, an all-IAX setup, or STUN, or a setup with SER/mediaproxy as a SIP server and asterisk behind it? thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: cdr_mysql and system time
hi all, does anyone know what time variables are fed to to the calldate field in cdr_mysql? I have my system time set to israel time zone, have restarted mysql and a show variables shows timzone as IST which means now() should return israel time, but the calldate field keeps getting the system clock. I don't have the source for asterisk-addons handy so i can't check the SQL. anyone? thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 not recognizing key beeps
what endpoints are you using? You probably have a DTMF type mismatch between asterisk and your endpoint (IP phone or softphone) -yair On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote: Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Grandstream firmware to use?
i've actually had reboot issues since moving to 1.0.5.16, the phones seem to hang more often on soft reboot and require a hard reboot (unplugging). This is just a feeling and i can't quantify this but i don't remember having to physically reboot the phones this often before. I'm using one bt-101 and one bt-102. -yair On Tue, 18 Jan 2005 10:50:30 +0200, David Norton [EMAIL PROTECTED] wrote: I've been using 1.0.5.16 for more than a week now, haven't had a single problem, and have not had to reboot it once. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Tuesday, January 18, 2005 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best Grandstream firmware to use? I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11.It's relatively stable, and the last thing I want to do is update to a flaky firmware Paul -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: asterisk and libretel
hi list, is anyone succesfully using asterisk with libretel port-of-call (www.libretel.com)? If so, i would be grateful for configs..i set up libretel to forward to [EMAIL PROTECTED]:5070 (asterisk is running on 5070 and SER on 5060) and when i call the number i see SIP messages with ngrep but the asterisk CLI doesn't seem to catch them. I assume i need to register...is this even possible or do i need to send the libretel number to a FWD account and register asterisk as a FWD client? The reason i want to trap the call in asterisk rather than SER is to keep all the call processing logic in asterisk. any help is appreciated, thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
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Re: [Asterisk-Users] Voicemail for Current Extension?
true enough, forgot the s...the s skips the password
my bad
-yair
On Sat, 4 Dec 2004 14:14:06 -0600, Brian West [EMAIL PROTECTED] wrote:
Forgot the s
VoiceMailMain(s${CALLERIDNUM})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Saturday, December 04, 2004 2:08 PM
To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
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[Asterisk-Users] re: DVG-1120
Hello, I know the d-link units (DVG-1120 ATA and their router as well) are supposed to work well with asterisk...does anyone know if the units that come with ATT callvantage are locked, or can they be used w/asterisk or SER? And if they are locked, is it linksys no way out locking or a simple password thing? thanks, yair ___ 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] xlite and asterisk
Hello, try this document (from the wiki): http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf setting the auth param and the canreinvite and reinvite might help. -yair On Thu, 11 Nov 2004 10:38:55 -, Ashling O'Driscoll [EMAIL PROTECTED] wrote: Hi, I havent received many replies so i was just wondering again if anyone has any thoughts of the 404 call not found issue.I have only a very basic configuration which can be seen below in the original email. I have since modified this so that each client (i.e. 2000 and 2001) have 'context=from-sip' included in their config and [from-sip] is in the extensions.conf file. I have now included the diagnostic log from the xlite client to see if that helps. Also when i do sip show peers I see: Username Host 2001 157.190.70.231 2000 84.203.148.14 The 84.203.148.14 is the address of asterisk, should the 2001 client be registering with that address too? Any help appreciated. Aisling. SEND TIME: 4679188 SEND 84.203.148.14:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE Proxy-Authorization: Digest username=2000,realm=asterisk,nonce=10dee878,response=fadc5f7ff0 2e3cbd5e8253d173f0b691,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 269 v=0 o=2000 4676744 4679018 IN IP4 84.203.148.14 s=X-Lite c=IN IP4 84.203.148.14 t=0 0 m=audio 8000 RTP/AVP 0 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 4679298 RECEIVE 84.203.148.14:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 84.203.148.14:5061;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A9;rece ived=84.203.148.14;rport=5061 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SEND TIME: 4679298 SEND 84.203.148.14:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 ACK Max-Forwards: 70 Content-Length: 0 Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] xlite and asterisk Date: Wed, 10 Nov 2004 12:39:22 + 404 not found can mean many things, are you using a supporting codec? On Wednesday 10 November 2004 05:25 pm, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - -- - ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] Reading extensions from MySQL database
use the wiki, luke. http://www.voip-info.org/wiki-Asterisk+Configuration+from+database On Tue, 02 Nov 2004 07:16:09 -0600, Director General: NEFACOMP [EMAIL PROTECTED] wrote: Hi list. Does anyone know of any configuration that will make asterisk read the extensions from a MySQL database instead of reading them from the configuration files? Thanks, __ NZEYIMANA Emery Fabrice NEFA Computing Services P.O. Box 5078 Kigali Office Phone: +250-51 11 06 Office Fax: +250-51 11 08 Web Fax: +1-530-326-4868 Mobile: +250-0851 7768 Email: [EMAIL PROTECTED] http://www.nefacomp.net/ ___ 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] re: asterisk SER and grandstream
hi list, anyone have any success getting asterisk to pass message waiting indicator to a grandstream with SER in the middle as a SIP proxy? I recently implemented SER between asterisk and my SIP clients and it's significantly more stable (no more dropped clients) but i haven't been able to figure out how to send message waiting indication so the grandstream's LCD flashes. If anyone has succesfully done this i'd be grateful for the info. thanks, yair ___ 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] Hardware Recommendations
if by digital phone you mean IP phone like a grandstream or a snom, then yes, you don't need any additional hardware to connect to * (except an rj45 cable, of course.) -yair On Fri, 22 Oct 2004 10:19:18 -0400, David Ishmael [EMAIL PROTECTED] wrote: I'm sure this has been asked more times than anyone cares to count, but I want to make sure I get the right stuff. I'm installing Asterisk at home solely to play with and learn VoIP (plus it sounds pretty cool so it has that going for it too). Eventually all my phones (~4 of them) will go through my Asterisk system. I've ordered my server hardware, here's what I've got: Trinity GC-SL Intel 2.66 533MHz 512MB PC2100 ECC WD 160GB IDE And some minor stuff (cdrom, floppy, etc.). Since this is strictly an educational platform for one person, I would think that's more than enough to handle my small number of calls. So I'm comfortable with that much...it's the Digium hardware I'm struggling with. What I'd like to do is connect the Asterisk PBX to my PSTN phone line so I think I need an X100P so I can make and take external PSTN calls, correct? I don't own a digital phone yet, so I think I also need to get at least one IAXy module for one of my analog phones (I'm still trying to figure out what kind of digital phone to get, recommendations are welcomed there too). I assume that any digital phone I get can talk to the PBX over Ethernet so I shouldn't need additional cards for that. Am I way off base on any of this or am I going in the right direction? Thanks, Dave ___ 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] re: ATA units: anyone have these working with * or SER?
Hello list, please take a look at these units: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596 are they locked? does anyone have one working with asterisk or SER? Are these rebadged units from a different manufacturer? anyone have any experience good or bad with these? thanks, yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: asterisk, SER and autocreatepeer
Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there something else to do? 2. according to the wiki the autocreatepeer creates peers based on the global variables. some variables, like dtmfmode, for example, are listed as belonging to individual peers. if i set dtmfmode, or qualify, or any of the others listed as individual variables, in [general] will the autocreatepeer use them? I suppose i could write a script to automatically generate peers for asterisk from SER's DB, (along the lines of the current retrieve_sip_conf_from_mysql.pl) but having duplicate SIP client entries seems kind of inelegant. And, of course, if i'm missing something basic conceptually, i'd be grateful if someone could point that out to me as well. any help is appreciated, thanks- yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: cdr and macros
i've been playing with cdr_mysql and the Master.csv file, and since i use a
macro to define extensions the csv and the db both save the destination of
the call as s, instead of the destination.
macro is as follows:
[macro-extensip]
exten =s,1,Dial(SIP/${ARG1},10)
exten =s,2,Voicemail(u${ARG1})
exten =s,3,Hangup()
exten =s,102,Voicemail(b${ARG1})
exten =s,103,Hangup()
and extension matching:
exten = _XXX,1,Macro(extensip,${EXTEN})
pretty standard stuff. how can i get the cdr to show the actual destination?
I guess i could parse the dstchannel field but i'd rather see what the user
actually dialed as well.
sorry if i'm missing something basic,
yair
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re: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem
hi, thanks for the help but transmit silence was set already. it appears to be an intermittent problem which makes me think it is local network related (i think i have packet loss.) thanks, yair From: Freddi Hansen [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: re: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem Date: Thu, 19 Feb 2004 11:53:17 +0100 From: yair hakak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 19 Feb 2004 07:54:00 + Subject: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem Reply-To: [EMAIL PROTECTED] Hello all, i have a one-way choppy sound problem that i can't fix... here are the relevant points 1. i am running 18.2.04 CVS on rhl 9.0 on a hosted server with a wide pipe up/down with no hardware, just SIP connections and voicepulse for outgoing IAX calls. 2. conecting to * with SJPhone (SIP) on a windows box that gets 1.5MB down and about 100K upload in speed tests (ADSL), so i'm pretty sure client bandwidth is not a problem either. the client can ping the server at 180-200ms as well. I've also tried x-lite and gotten the same issues. sip clients register fine, and i can hear incoming audio fine, but on the other end it is completely garbled. It is not an IAX problem; if i leave voicemail from the SIP client on * and try to pick it up it is garbled, but the voicemail prompts are crystal clear. there was a thread about this at the beginning of january - the only solution that came up was to sweep the windows box for worms - which i did, and i have no worms. if anyone who had the problem then has answers, or anyone else, i would be most grateful. thanks, yair Try to set the following in your x-lite config. I had one-way choppy sound and this was the cure. AdvancedSystemSetting-AudioSettings-SilenceSettings-TransmitSilence:yes Freddi ___ 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 _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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] help a poor newbie out with SIP choppy one-way problem
Hello all, i have a one-way choppy sound problem that i can't fix... here are the relevant points 1. i am running 18.2.04 CVS on rhl 9.0 on a hosted server with a wide pipe up/down with no hardware, just SIP connections and voicepulse for outgoing IAX calls. 2. conecting to * with SJPhone (SIP) on a windows box that gets 1.5MB down and about 100K upload in speed tests (ADSL), so i'm pretty sure client bandwidth is not a problem either. the client can ping the server at 180-200ms as well. I've also tried x-lite and gotten the same issues. sip clients register fine, and i can hear incoming audio fine, but on the other end it is completely garbled. It is not an IAX problem; if i leave voicemail from the SIP client on * and try to pick it up it is garbled, but the voicemail prompts are crystal clear. there was a thread about this at the beginning of january - the only solution that came up was to sweep the windows box for worms - which i did, and i have no worms. if anyone who had the problem then has answers, or anyone else, i would be most grateful. thanks, yair _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone
Does anyone have any ideas on how to stop these messages from the SJPhone? everything i've seen says they're harmless, but they're filling my console and if anyone has any ides on how to make them go away i would be appreciative. thanks, yair _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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] help with h.323 outgoing calls
is anyone using h.323 to send outgoing traffic to a voiIP termination provider? if so, could you send me a sample h323.conf file and the relevant line from extensions.conf thanks- yair _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: help with voicepulse connect IAX2
hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am pretty convinced the SIP setup is OK. This is the error message: Jan 29 12:21:54 NOTICE[262161]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' when i try to call the PSTN from the SIP device. i've tried the wiki, the handbook, the voicepulse site, and all sorts of other sites, and nothing helps. i also downloaded and compiled the code today (jan 29) and that didn't help either. if anyone has ideas i would be eternally grateful - this is driving me crazy. thanks- yair p.s. i am using the right login and password; not the ones from the website, and i know my account at voicepulse works because i can connect direct through a SIP client. it seems to be a specifically IAX2 problem. here are my files sip.conf ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm [yairphone] type=friend insecure=no username=yairphone secret=yairphone host=dynamic dtmfmode=inband callerID = Yair Hakak nat=true extensions.conf [general] ; static=yes writeprotect=no [default] exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20) exten = 8665,1,Dial(SIP/yairphone,20) iax.conf [general] port=5036 disallow=all allow=ulaw jitterbuffer=no [voicepulse] context = VPWS secret=mypassword auth=md5 type=friend host=gw5.voicepulse.com _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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
